LinuxMCE Forums
General => Users => Topic started by: jondecker76 on January 08, 2009, 07:17:26 am
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Looking for others that wish to connect LMCE to their analog phone line using a Sipura 3000. It would be nice to document this process so that in the future we can automate the setup.
Goal:
Connect an analog phone line to LMCE to take advantage of some of Asterisks and LMCE's features, such as:
- Forwarding calls depending on status (Asleep, away, at home, etc..)
- Priority caller features
- Integration with security system (call neighbors, your cell phone, police, fire dept. on different security events)
- Answering Machine features / mailbox features
- Dialing from orbiters (phone book, speed dial etc)
- Recording and archiving of conversations
- Hold features and on hold music
- pausing media upon receiving a phone call
As you can see, for those of us that don't want to use VOIP, there are still many benefits of hooking your analog phone system up to LMCE.
The idea is to have the internal phone line treated as 1 extension (all hard wire pstn phones ring together). Beyond that, normal VOIP phones can be used as additional internal lines, which will interface the Analog line through Asterisk.
Another nice feature with this approach and with the Sipura 3000 is that upon a power outtage or failure of the core, the FXO and FXS ports are jumpered automatically, linking your internal phone line to the analog line, allowing the phones to still be used.
Anyone want to put their heads together with me to get this set up? My Sipura 3000 is dieing to be configured and interacting with LMCE!
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Looking for others that wish to connect LMCE to their analog phone line using a Sipura 3000. It would be nice to document this process so that in the future we can automate the setup.
Goal:
Connect an analog phone line to LMCE to take advantage of some of Asterisks and LMCE's features, such as:
- Forwarding calls depending on status (Asleep, away, at home, etc..)
- Priority caller features
- Integration with security system (call neighbors, your cell phone, police, fire dept. on different security events)
- Answering Machine features / mailbox features
- Dialing from orbiters (phone book, speed dial etc)
- Recording and archiving of conversations
- Hold features and on hold music
- pausing media upon receiving a phone call
As you can see, for those of us that don't want to use VOIP, there are still many benefits of hooking your analog phone system up to LMCE.
The idea is to have the internal phone line treated as 1 extension (all hard wire pstn phones ring together). Beyond that, normal VOIP phones can be used as additional internal lines, which will interface the Analog line through Asterisk.
Another nice feature with this approach and with the Sipura 3000 is that upon a power outtage or failure of the core, the FXO and FXS ports are jumpered automatically, linking your internal phone line to the analog line, allowing the phones to still be used.
Anyone want to put their heads together with me to get this set up? My Sipura 3000 is dieing to be configured and interacting with LMCE!
Hi,
I'm using Sipura 3000 for connecting analog doorphone to my system... Configuration is I guess pretty similar, so I can probably help with some details...
Sipura is nice device, but has a lot of config options and I haven't configured it since ages when it started to work. I'm only changin sip credentials when I integrate it in new system...
Let's start discussion and wiki page on this one....
Update: I did my first and the only configuration with some web configuration utility, that let you only fill subset of needed information and it prepared configuration file for sipura. After that, I only did minor tweaks to the configuration. I can't find it anymore, but maybe this will be of some help :
http://www.aussievoip.com/wiki/index.php?page_id=119 (http://www.aussievoip.com/wiki/index.php?page_id=119). I can also make snapshots of my setup and send it to you over PM.
regards,
Bulek.
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Hey guys,
I have a spa-3102 that I use with my system. I wrote a wiki on it http://wiki.linuxmce.org/index.php/Linksys_spa-3102. Hopefully this helps.
Also, the articles I found relevant on setting it up are located at the bottom of that page.
Hope this helps,
Golgoj4
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thanks a lot - i'll be checking this out during the week!
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I started working on this a bit. Following your wiki instructions, I got to this step:
If you want the phone attached to your SPA-3102 (Line 1) to ring when people call, you should go to Wizard -> Devices -> Phone Lines, and click Settings for the appropriate phone line(s). There you can modify the ringing behaviour of your new phone.
If I go to Wizard->Devices->Phone Lines, there are no lines listed. There is a drop box to add a new line, but they are all VOIP specific. What shoudl I do here?
thanks,
jon
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Correct me if I'm wrong, but I think you need to finish the steps under "Configuring FreePBX" first for the lines to show up there.
By the way, nice wiki page golgoj4!
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I added the trunk in FreePBX. At the top of the page for my trunk, it has an error
WARNING: This trunk is not used by any routes!This trunk will not be able to be used for outbound calls until a route is setup that uses it. Click on Outbound Routes to setup routing.
Messing around with this is showing me how far behind this area of LMCE is (as far as configuration and ease of use goes) I'm really hoping to get this figured out so the process can be automated
I've started reading the book "Asterisk: The Future Telephony" - I have a feeling like i'm going to have to fully understand Asterisk before I can ever get it configured correctly.
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I started working on this a bit. Following your wiki instructions, I got to this step:
If you want the phone attached to your SPA-3102 (Line 1) to ring when people call, you should go to Wizard -> Devices -> Phone Lines, and click Settings for the appropriate phone line(s). There you can modify the ringing behaviour of your new phone.
If I go to Wizard->Devices->Phone Lines, there are no lines listed. There is a drop box to add a new line, but they are all VOIP specific. What shoudl I do here?
thanks,
jon
This is something I have not yet figured out how to do. Currently, the setup is only for voip. I had to do all my customization in freepbx / linksys admin page to get the call into lmce. I have no idea just yet on how to make the configuration options for pstn appear. My uninformed self tells me it involves figuring out a couple things
-base configuration similar to the voip setup only more stripped down as there isnt any authentication with pstn lines.
-pnp some how to auto configure a pstn line when supported ata is connected.
thats basically where I left off a while ago. But to be honest, im still trying to understand fully why it works and how free pbx works in relation to the webadmin.
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ok. well at least you have a start on this. I'm going to take some time to read through this book and get a better understanding of Asterisk, then disect some scripts and get an idea of how LMCE interacts with Asterisk. Maybe then the answer will pop out at us..
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Hi,
I did mess with Asterisk & LMCE integration some time ago and remember few things... The last state of this problem was that Aaron said, that they're seeking for Asterisk guru to fix that Asterisk & FreePBX integration under LMCE, but I guess nothing changed from there. We're using quite ancient versions of FreePBX and Asterisk, cause some problems were fixed in there and compilation/installation script is now the nightmare, only Asterisk guru can solve....
Writing out of memory:
I'm dealing with Asterisk installation under LMCE in this way :
- add everything that is possible to add via web-admin (at least phone devices), then some perl scripts will be run that will put those devices also in freepbx or Asterisk config files (look in /etc/asterisk/).
- then I do everything that is not possible in web-admin (Advanced-Configuration-Phones Setup) on "normal" Freepbx interface. But beware, changes in there will not be synced to LMCE database - so don't add phones there if you don't know why are you doing it...
I have setup with three VOIP providers, doorphone on Sipura interface and many routes etc...
My logic is usually setup in such manner, that I declare two dummy users (doorphone, housephone) and do all call routing features through them (I guess doing it for each family members is too much work, while we all use common phones). The incoming call logic goes in this way: I just reroute calls to doorphone or housephone user and then use all call routing features you can setup via web-admin under LMCE. There you can determine behaviour based on housemode, incoming caller etc...
I can help you with questions regarding that... Maybe we should start wiki page about using Asterisk under LMCE....
Regards,
Bulek.
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hi all,
I'd love to have this as a feature (tech abilities a bit limited as are funds), but would this,
http://www.pctradestore.com/code/ui/main/product_info.aspx?prdid=ATA&catid=7&heading=VOIP%20SIP%20/%20PSTN%20analogue%20ATA%20Adapter%20(%201*%20FXS,%20%201%20*%20FXO%20&%201%20RJ45%20WAN%20)
substitute for the sipura,( it's £25 as opposed to £150 ;) )
Any comments/advice regarding spec /shortcomings /potential pitfalls greatly appreciated,
regards,
Ian
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- then I do everything that is not possible in web-admin (Advanced-Configuration-Phones Setup) on "normal" Freepbx interface. But beware, changes in there will not be synced to LMCE database - so don't add phones there if you don't know why are you doing it...
Bulek,
what kind of things can't be done in the lmce interface? And would you mind to maybe add those things to the web admin interface of lmce? ;)
rgds
Oliver
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maybeoneday:
It has an FXO and an FXS port, so technically it shoudl work *if* you know how to configure it correctly (in its own admin interface - should'nt be much different on the asterisk end). However, with the cheap price, one would have to imagine that the quality will leave a lot to be desired.
Can anybody that is currently using a VOIP provider such as broadvoice post a screenshot of their Phone Lines admin page?
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- then I do everything that is not possible in web-admin (Advanced-Configuration-Phones Setup) on "normal" Freepbx interface. But beware, changes in there will not be synced to LMCE database - so don't add phones there if you don't know why are you doing it...
Bulek,
what kind of things can't be done in the LinuxMCE interface? And would you mind to maybe add those things to the web admin interface of LinuxMCE? ;)
rgds
Oliver
hmm, a lot of things... You cannot do inbound, outbound routes, trunks etc... Basically you can only add phones, some predetermined VOIP providers and of course quite nice routing settings via web-admin...
I think it would be really complex task to duplicate FreePBX under web-admin and probably pointless too, cause majority of users should have enough existing web-admin interface...
What we would really need right now in this area is to update Asterisk and FreePBX to more current versions for 8.10, but that's a complex task cause current setup has some hacks in it.... and only some experienced Asterisk guru can do this....
Regards,
Bulek.
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hi all,
I'd love to have this as a feature (tech abilities a bit limited as are funds), but would this,
http://www.pctradestore.com/code/ui/main/product_info.aspx?prdid=ATA&catid=7&heading=VOIP%20SIP%20/%20PSTN%20analogue%20ATA%20Adapter%20(%201*%20FXS,%20%201%20*%20FXO%20&%201%20RJ45%20WAN%20)
substitute for the sipura,( it's £25 as opposed to £150 ;) )
Any comments/advice regarding spec /shortcomings /potential pitfalls greatly appreciated,
regards,
Ian
I realize the linksys is more expensive, but I found that it has so many features that I dont even use them all. Key in those features was its ability to act as pstn trunk. Now, I dont know that much about phone systems, but one of the reasons I went with the linksys is that is that it was highly recommended on many forums I visited because of its flexibility. So what im sayin in essence is try to find out as much as possible about the device your looking at. google the product name and model number as its a good way to get hits on issues and more information in general from people who took the plunge already. My internet at home is a bit wonky but I will grab some screenies of the linksys admin page as well as what I have going on in free pbx so that you can see what I mean. The main benefit of my spa-3102 is there were guides to help me understand most of it to the extent I could get the system running.
So check out the support out there now for it as well as how configurable it is. From what I read at a minimum it should be able to connect to asterisk, but im not sure about it acting as the pstn bridge TO lmce without further information.
hth
golgo
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- then I do everything that is not possible in web-admin (Advanced-Configuration-Phones Setup) on "normal" Freepbx interface. But beware, changes in there will not be synced to LMCE database - so don't add phones there if you don't know why are you doing it...
Bulek,
what kind of things can't be done in the LinuxMCE interface? And would you mind to maybe add those things to the web admin interface of LinuxMCE? ;)
rgds
Oliver
hmm, a lot of things... You cannot do inbound, outbound routes, trunks etc... Basically you can only add phones, some predetermined VOIP providers and of course quite nice routing settings via web-admin...
I think it would be really complex task to duplicate FreePBX under web-admin and probably pointless too, cause majority of users should have enough existing web-admin interface...
What we would really need right now in this area is to update Asterisk and FreePBX to more current versions for 8.10, but that's a complex task cause current setup has some hacks in it.... and only some experienced Asterisk guru can do this....
Regards,
Bulek.
I agree that more can be done in freepbx, but I think that adding pstn setup option would be a start. I agree duplicating freepbx would be silly but thats a bit above my paygrade as im still working it out myself.
From what I understand, there are setup scripts that run when you configure a voip provider as well as extensions.
so wouldn't we do the same with the pstn? i suppose its a matter of hashing out a standard setup file for a pstn line based on location. And compatible devices that can accept the settings and act as the trunk.
just thinkin ...
golgoj4
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hi all,
@ golgo/jondecker ,
thanks for taking the time ,and the advice, considering that, and that I've found the lynksys at £50 ish,convinces me...spa3102 about to be ordered,,
many thanks,
Ian
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The first step is just education. The free downloadable O'Reiley book available on the Asterisk website ("Asterisk: The future of telephony") is a great start (600 pages of good information). I'm a couple hundred pages in so far and a lot more is making sense now. I highly suggest anyone that wants to help get this going read the book. There is detailed information on Dial plans and AIG (the protocol to interface to Asterisk from external programs). If we can understand thes relatively simple concepts, and do a little research on the current telecom_plugin, it actually shouldn't be har or take an expert to get things fixed and implemented. I'll try to spin off a post in the dev area to see who is interested in helping.
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hmm, a lot of things... You cannot do inbound, outbound routes, trunks etc... Basically you can only add phones, some predetermined VOIP providers and of course quite nice routing settings via web-admin...
I think it would be really complex task to duplicate FreePBX under web-admin and probably pointless too, cause majority of users should have enough existing web-admin interface...
Is there anything in the capabilities that is currently only provided thru FreePBX that you think you could add to Web Admin. So that maybe 10% more of the tasks needed could be accomplished within the lmce environment.
rgds
Oliver
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I'm making some really good progress with this so far. I now understand Asterisk and the codebase enough that once I figure out the best settings for the SPA3000 (both on the local device, and in FreePBX), I can get automated setup to work.
The "Phone Lines" section of the web admin is responsible for setting up a Trunk, Incoming Routes and Outgoing Routes in FreePBX. My plan at this point is to add a "SIP Device" for generic interfaces such as the SPA3000, and "Zap Device" option (if I can get an analog card) to the dropdowns to have automatically LMCE automatically configure. (At first sight, the SPA3000 appeared to be a Zap (analog) interface, but in reality it is a gateway and presents the phone line and the pstn as SIP devices to Asterisk). This is why we need to add configurations for both generic SIP and generic ZAP devices.
The voxilla website also has an automatic configurator for the SPA3000 which can automatically upload the settings to the device - which shows promise that setting up a device template and using a pnp configuration script, it may be possible to automate the entire process of configuring the SPA3000 settings. All very good news so far.
Also, I was very surprised at how well-coded and complex the custom-linuxmce dialplan is. Once again, it shows that the Pluto guys really knew what they were doing - even when things look jacked up at first glance.
What I could use right now to help get this moving:
-golgo already posted his method of getting the SPA3000 to work (though it would be nice to see how the Incoming Routes and Outgoing Routes are set up). Bulek or anyone else, can you post details on your SPA3000 configuration, as well as how things are set up in FreePBX for Trunk, Incoming Routes, and Outgoing Routes? The more examples we have, and the more examples I can test out, the better decision I can make on how to have LMCE set things up
-I do not have a true ZAP interface (analog PSTN card). If anyone has one they could put out on loan, I'd gladly try to add support for these devices as well (I'll buy one if I must, but thats not something I could do any time soon)
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Im going to start reading the o'reily book as well
my spa-3102 isnt setup fully. I forgot I nuked my lmce setup and only set it up as a voip extension on reinstall as nobody actually calls me on my pstn line. It will be done later tonight so I can post my results.
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The O'Reiley book was excellent - i finished it this morning. A lot to digest, but gives absolutely needed information on the workings of Asterisk (dialplans specifically) and gives you a really good breakdown of kinds of things you can do with Asterisk. I'm going to see if there is a printed version of the book, as it is a perfect desktop reference.
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Hi,
I'll post some details about my setup (I guess it's quite similar to yours - I only have GSM gateway on FXO interface - you will have your phone line). I have incoming calls from GSM gateway going to user housephone - this is dummy LMCE user (won't show on floorplan), but it's purpose is to have only one "user" receiving all incoming calls - so I can go in web-admin and set all call routing features only for that user (I came to this solution for me, cause we're 4 in my family, but we all use 2-3 common phones, so making routes for each user is too complex for our situation)...
Basically you have to imagine sipura in your case as two devices :
1. from FXO port (your line) to VOIP - Asterisk gateway -this is basically representing SIP extension to Asterisk (240) - "PSTN" line in Sipura web interface (I have also under Dial plan #8 line : (<S0:401>) - that means that on incoming call, extension 401 will be called)
2. from FXS (you can attach phone or doorphone in my case) to VOIP Asterisk gateway (another sip extension - 209 in my case) - "Line 1" in Sipura web interface
You must setup those two (I think that Voxilla configurator will setup this for you already)...
Now on Asterisk (LMCE) side :
1. I have SIP trunk defined for sipura device :
put setting data for 240 "PSTN line" in Peer details. This trunk will be used for your outgoing routes (so call will go to your phone line)
2. extension 401 is custom extension defined only (no real device) and I have this line in there :
This device uses custom technology.
dial Local/306@ext-local
That means that whenever I receive GSM call from Gateway (this is equal to your phone call), call will be forwarded to extension 306. Extensions 3XX are not visible anywhere in FreePBX, but are created in Asterisk config files for each LMCE user. So extension 306 is extension for my previously mentioned user housephone (where all incoming calls go to).
What happens with calls on that extension, is determined by call routing setup you can do on web-admin - there you can specify behaviour regarding house security mode, Caller id, etc....
Comment: maybe calling 306 in first place instead of 401 in dialplan of Sipura first will also work, but haven't tried it...
3. outbound routes setup in Freepbx:
this where you determine how and where your outgoing calls with go. You can easily use your defined trunk as part of outbound routes...
Beware: this setup is done more in Freepbx than via web-admin, phone lines ,etc... works for me, cause I used Asterisk before and I want to setup few extra things in Freepbx (like more trunks in certain order for outgoing calls etc...), so maybe similar result can be achieved via web-admin in more proper way...
HTH,
regards,
Bulek.
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SUCCESS! I have my spa3000 configured PROPERLY in LinuxMCE (properly meaning that it follows all the standards of other LMCE phone lines, and interacts with LMCE as it should)! I started with a blank slate, compared several of the automated setup phone lines, traced a bunch of code, applied what I learned from reading the Asterisk book, and what others have posted here. Thanks a lot for all of the useful information!
What works:
Just about everything! When a pstn call comes in, all of the orbiters in the house come on to the incoming call screen. If unanswered, LMCE voicemail takes over. CallerID comes in onto the orbiter screens just fine. Changing settings in the web admin works as it should. I can place outgoing calls just fine.
What doesn't:
-Dialing internal LMCE extensions... I haven't tried this much, when I get time I'll have to look through this a bit more.
-Dialing out works just like an ordinary phone ( you dont' have to dial a 9 first). This is incorrect and I need to fix this. It is also probably the reason dialing internal extensions didn't work when I tried.
Just wanted to give a quick report - its getting there! I would imagine being finished within a week or so.(to include full auto-config hopefully!)
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What works:
Just about everything! When a pstn call comes in, all of the orbiters in the house come on to the incoming call screen. If unanswered, LMCE voicemail takes over. CallerID comes in onto the orbiter screens just fine. Changing settings in the web admin works as it should. I can place outgoing calls just fine.
Wow! Jon you are one amazing dude!
Is this the toy you're using?
http://cgi.ebay.ca/NEW-Linksys-SPA3000-VoIP-adapter-1-FXO-SPA3102-HT-488_W0QQitemZ370124599022QQcmdZViewItemQQptZCOMP_Telecom_IP_Telephony?hash=item370124599022&_trksid=p3286.c0.m14&_trkparms=72%3A1215|66%3A2|65%3A12|39%3A1|240%3A1318
I may just go ahead and get one for playing with here!
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yes, thats the right one (you can also look for the older Sipura models from before Linksys bought them out - they may be even cheaper)
Also, I figured out why dialing internal extensions didn't work and why I didn't need to dial a 9 for outgoing calls.. I haven't set the FXS port up in the spa3000 yet - so essentially my outbound calls are using the internal routing of the unit, and bypassing Asterisk all together. No big deal, I will set up the FXS (internal phone line) port tonight, add a new Phone device template and assign it an extension - and all should work 100% exactly like the voip phones, with full LMCE integration.
I'lll be posting back here hopefully within the week with exact instructions on how to set this up so others can enjoy this while I work on the automated setup end of things.
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yes, thats the right one
Kewl. Just ordered one!
This unit can be put anywhere on the internet network I take it? closest network port and phone jack?
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Yes, the ehternet port of the spa3000 just connects to your internal network.
There is also 2 phone jacks - one for the phone line (pstn), and the other for a telephone. How I am using mine is brining my pstn line straight to the device on the FXO port, then hooking up the main house phone line (which all house phones are connected to) to the FXS port. This treats the entire house's "land line" into one extension. So then I can transfer calls to a voip phone (on its extension), to a media director (on its own extension), or to the normal house phones (which has its own extension)
Of course, if you wanted to add more FXS ports, each analog phone in the house could have its own extension, but I think that treating the entire existing analog house phone system as one extension makes more sense. If for some reason you want any rooms to have their own phone and extension, this can be done with simple cheap SIP phones, while keeping the landline as its own extension.
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jeebus dude im only barely into the book! But seriously, awesome that your up and running so far. It seems i misunderstood you in that the spa-3000 is just an older spa-3102. I didnt realize it had and fxs and fxo port on it. Anyways, im very interested in your results as I have had little luck with dial plans, allthough im attempting to split between voip and pstn but still, awesome work!
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AFAIK, the 3102 and 3000 are identical, with the exception that the 3102 adds router functionality (DHCP server). So anything I get working on the 3000 should work on the 3102!
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Just another update...
Everything is 100% working and set up. All LMCE standards have been used, and if you compared LMCE interaction and operation of using the pstn line/spa3000 and a VOIP account, you would not know the difference. Absolutely everything is 100% tested and working.. Conferencing, paging through the house, forwarding to other extensions, calling extensions, sending and receiving calls, routing to voicemail, priority callers, you name it! I will be posting instructions on how to set this up soon, but I want to do a few things first:
- standardize naming conventions on a few things. This way, once I automate the process, what you create manually through instructions will be the same as it will be when auto setup is implemented
- Test just a couple more things out (mainly interaction with Security - calling your cell phone on a security breach and the like.
- Watch a movie with my wife before she kills me
All I can say is that this is really really really fun, and really helps to complete a full automated house!
After this is all finished, i'm going to implement support for the spa2000 (2 FXS ports, for adding additional hard wired analog lines). I'm preferring this over the Digium cards because it doesn't hog up PCI slots, and only has to connect to the router on the internal network.
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jondecker,
you're amazing,
I've just (04:10 gmt) finished skimming the o'reilly book, and now I've got a problem,
1) go to bed
2)unpack and install the spa3102 that came earlier (actually yesterday, i didn't realise I'd been reading so long)
;D
just one befuddled thought, if a dect setup was connected rather than single phone could the dial plan include divert/conference and other capabilities of dect system?,
congrats,
regards,
Ian
Edit...bluetooth phones as handsets for md's?
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Ian,
I doubt you could ring separate extensions in a Dect setup (my phones in my house also have only one base station, which all "drone" phones connect to). For this reason, I decided to treat my entire internal landline as 1 phone, and I'm pretty happy with that.
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I have finished a wiki page detailing how to manually set this up. The more people we can get testing this, and the more we can verify everything works as it should, the sooner I can add this as a plug-and-play device for LinuxMCE. Please, lets test EVERYTHING out, I'd like to have this working perfectly and 100% supported for 0810.
Here is the wiki page:
http://wiki.linuxmce.com/index.php/Sipura/Linksys_spa3000_pstn_interface (http://wiki.linuxmce.com/index.php/Sipura/Linksys_spa3000_pstn_interface)
Problems I know of so far:
1) Upon power outtage, or network failure/core crash, the FXO and FXS lines are bridged so you can still use the phones. Dialing out works fine (just unplug the power and/or network cable from your spa3000 to see!). However, incoming calls in this failsafe mode only ring once. (Update: now fixed)
2) Incoming calls don't directly ring the house phone extension. All orbiters alert of the call, and the call can be directed to go to the house line, but I would have expected that the houseline would ring and could be picked up on a call with no orbiter interaction at all. (UPDATE: Found the problem - this will work once I add the web admin code to install the phone line - this is now fixed)
Please list any other problems you can find. If you know of a way to fix any of the problems we find, try it out and report back here if you are successful.
Thanks, and enjoy!
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GREAT WORK Jondecker,
& brilliant wikki,
will be installing / testing tommorrow, will feedback asap,
many thanks
Ian
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Hi jondecker,
been trying your setup with no joy :(
mainly , I think 'cos I'm in UK, & using spa3102
FOR UK USERS......www.aoakley.com/articles/2008-01-08.php has the relevant settings (untested by me as yet,imminent) and from there I'll try again with freepbx as per your wiki.
NB pay close attention to first part concerning cables and testing phones
regards,
Ian
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the only settings you are going to have to dial in should be on the spa3102 - everything else should be the same. Most of the settings should be under the "Regional" tab of the spa3000/3102 - these are the tones and signals the sipura looks for on the line to know whether to answer, hang up, etc.
Found out from someone who called today that music on hold works - though I didn't successfully navigate answering a call on call waiting and returning to the original caller. Will try again later
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Hi Jon.
I just installed the SPA3000 with mixed success.
I ran a network cable out of the basement office and through a bunch of floor joists to reach the demarc point of the phone line by the fuse panel. I then wired in a couple phone boxes so I could connect a phone cable from the demarc point to the box feeding the house phones (to keep things "normal") but that also lets me connect the SPA3000 in between so I can 'test' and put things back to normal when done.
First, the "House Line" doesn't seem to work at all. If I try to make a call I get a long pause and then a busy signal. Incoming calls appear on the orbiter, but if I route it to the "House Line" no inside phones ring. If I route it to voice mail, I can hear the prompt to leave a message, and when I do and hang up, the message never appears.
After creating the "dummy" phone line, when I went back into the FreePBX admin and looked at the Incoming Route for the boardvoice line, and Custom App option, it contained "custom-linuxmce,103,1". So I went back to the Incoming Route for the House Line, and changed its custom app line to match as instructed. (so it used 103,1 instead of 102,1)
I didn't see anything else in the instructions that varied. I reloaded the router before testing incoming/outgoing calls. When I went back into FreePBX I noticed the "Apply Configuration Changes" bar was there again so I clicked it and continued with the reload, tested again, same results.
I do see status boxes on the top left of the orbiter saying call lost or something in rather odd times (while I'm speaking to the voice mail for instance?)
I do have one major issue...
We use Bell HSE (ADSL) for our internet and when I connect the SPA3000 and then hang the house phones off it, we lose our internet. I guess I'll need to purchase a splitter and run a filtered line into the SPA and leave the jack on the SPA, which would feed the house, empty, feeding the house from the other side of the (unfiltered) splitter at the demarc point.
In this setup, as far as I can tell, the only things I'd gain are:
1) on the orbiters... I'd be able to see the incoming phone number displayed. Can it display the name of the caller along with the phone number?
2) the answering machine function should work
Would it be able to make outgoing calls (in the case of security alarm messages) set up this way?
The other way of doing the connection would mean relocating equipment and running even more network cable throughout the house, or perhaps running a dedicated phone wire to the upstairs dining-room to the ADSL modem and separating that from the rest of the house feed.
On a more general note, when this is set up in the manner you've described, does somebody have to route the call on an orbiter each time a call comes in? I'd like it to fall through to the house line and ring normally.
We have an existing answering machine and I already know the better half is not interesting in trying to get her messages from LMCE. I believe there is a setting somewhere to disable LMCE from taking messages?
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Freymann
I may have an error somewhere in the instructions I posted in the wiki. Can you get on IRC today, maybe we can find the problem by comparing.
As far as disabling the answering machine (IVR), in the web admin there is a field to adjust a timer for it to kick on. The instructions on the web admin say to set it to 0 to disable it, but this didn't work for me. I set it extremely high so it doesn't kick on at all (90 seconds) - our home answering machine picks it up well before then. (the horrible robotic voice IVR sucks and needs a complete overhaul in my opinion)
Yes, you can set incoming calls to ring on any combination of your media directors and your house line.
While on the subject - here are some things I have learned so far using pstn with Asterisk
- ECHO! Echo is a major problem. This is unavoidable and due to the facts that both signals transmit over a single pair of wires and the fact that Asterisk introduces some latency. Asterisk does have software echo cancellation - however, it is a part of the zapata driver and only works with PCI cards! You can get reasonable performance from the spa3000 if you match the impedance and gains well, and set the jitter buffers to low (to reduce latency). This takes some tweaking however.
- Digium sells a very good software echo canceler that can be compiled with zapata drivers ($10 license). Again, this only applies to pci cards that use the zapata driver.
- Digium sells cards with hardware echo canceling - this is probably the best option for call quality, but expect to spend at least $500 for one FXO and one FXS port (ouch!)
- I may try the cheap x100p cards that sell really cheap on ebay (about $30). They use the zapata driver and can make use of the software echo cancellation (both the freeware ones and the commercial one from digium).
- I haven't yet been able to change the on-hold music. I'll be playing with this more as I get free time
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I may have an error somewhere in the instructions I posted in the wiki. Can you get on IRC today, maybe we can find the problem by comparing.
I just had some free time but the chat link at the top of the forums does nothing for me?
As far as disabling the answering machine (IVR), in the web admin there is a field to adjust a timer for it to kick on. The instructions on the web admin say to set it to 0 to disable it, but this didn't work for me. I set it extremely high so it doesn't kick on at all (90 seconds) - our home answering machine picks it up well before then.
Ok, I think I found that.
Yes, you can set incoming calls to ring on any combination of your media directors and your house line.
Awh, this is in pluto-admin... Wizard > Devices > Phone Lines.
I see they have a Local Number Length. It was 7 but we require 10. Changed it.
From this same screen, look for the boardvoice line (my only one) and click on Settings under Actions on the far right and check off "House Phone" for all the scenarios required.
I put a splitter on the phone demarc and have one half feeding the house phones and the other half just feeding the spa3000, and this seems to be working OK.
I still don't end up with any messages if I say send to answering machine on the orbiter though.
EDIT: Actually, in pluto-admin, Telecom > General Voicemail, they appear here, but I can't listen to them on my ubuntu workstation (talks about installing something, then you say sure, then it says nothing installed). On XP in MSIE, I get a mini player but no sound. Can you not send a call to a specific user's voice mail?
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Hi Jon.
An update on my setup...
I can make outside calls from any orbiter, dialing 9 first...
I can answer a call from any orbiter. I can hear the caller but they don't hear me. I had a mic connected to my MD but most likely I gotta fiddle with a setting somewhere, someplace to activate it?
If I send a call to voicemail, it goes to the general voicemail account which I can only access via pluto-admin. If I click on any of the two user accounts nothing happens. This is weird.
I can see in the future, where having LMCE answer the phone and allow users to select 1 for me or 2 for the wife beneficial... especially if we can then have it forward those messages to our cell phones and/or email boxes.
I finally managed to figure out how to get into IRC and will hang out there a bit today.
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Hi,
you can also try to call *43 (loop to Asterisk and back)...
Regards,
Bulek.
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just updated the wiki with another diagram of how the connection should go if you use DSL/ADSL for your internet access
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Here's a few things I discovered today.
Logged into LMCE as user "Gerry"
Orbiter shows an incoming call. I send to User "Gerry"
Voice Message is recorded, and on the orbiter, it now shows 1 message waiting for me.
If, when the oribiter shows the incoming call, and I click on "Leslie" instead of "Gerry" (in which I assume it would sent to Leslie's voice mail) it still goes to "Gerry"'s voicemail.
Did another test.
Logged into LMCE as user "Leslie"
Orbiter shows an incoming call. I send to User "Leslie"
Voice Message is recorded, and on the orbiter, it now shows 1 message waiting for Leslie (and 1 for Gerry, as above).
To play back voice mail --
At a MD with sound... Make sure correct user is logged into MCE. Click on Telecom and the logged in user's name.
Now you get a display allowing you to change your user status, and below is a list of voice mails, called:
New message 1
New message 2
New Message 3
etc.
Click to the right of the "New message 1" in the unused green area and it should play back that message, over and over. To stop it from playing back, go back to the main screen, Green Button (or F7) and then OFF. The audio system is playing back your messages.
You have to keep going back to Telecom > User > click on the green space, main menu, off to cycle through voicemails.
If I'm logged in as "Gerry" but click on Telecom > Leslie it doesnt' show me her list of voicemails, it shows me mine.
You can log into the Pluto web admin to hear your messages through your web browser... click on Telecomm > My Voicemail then click the play button beside your message. On ubuntu, yesterday, when I tried this, it talked about installing a plugin, which appeared to have failed, and I never did hear anything. Today, now that the computer has been rebooted? It's working. Also seems to work fine under MSIE and WinXP.
To be able to click on Gerry and have it sent to voicemail, I configured Telecom > Call Routing (from the top of the pluto web admin) > Gerry User Mode At Home > Normal Caller to be set to 'Go To voicemail'.
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Jon mentioned one could press *98 to get to voicemail over a phone connected to the Phone jack on the sipura.
That doesn't work under my setup, I just get a busy signal.
If I dial *43, same thing, just a busy signal.
In fact, I can't make outgoing phone calls from a phone connected to the Phone jack on the sipura. Just get a busy signal. Dialing 9 then the number doesn't make any difference.
I can make a call from an orbiter just fine (dialing 9 first), and I can hear the person at the other end, but they can't hear me.
That's probably all I'll have time to experiment with today.
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Freymann
I should be on IRC tomorrow from around 1300 until 1500 EST. It sounds like something may be wrong with the Line1 setup tab on the spa3000. I'll try to catch you on there tomorrow and we should be able to get that last part figured out. Though if you get a chance, I think I just spotted a mistake in my wiki instructions. In the spa3000/Line1 setup page, for password, instead of l m c e as a password, put the extension number as the password (so that Display Name, UserID and Password all read the same number like 206 for example). I believe that this will fix your internal line problem.
Regarding some of the other abnormalities you mentioned, I'm sure that there are some bugs in the telecom system - after you are all set up, lets continue to track them down and put Trac tickets in on them so they get fixed. While I am realizing that the spa3000 may not be the pest way to interface the pstn (because of echo problems) - it will allow us to fully test the system, track down bugs and get things fixed. I will soon be getting some PCI pstn cards to try to add support for - in the end these may be the best choice because of echo cancelation. Thanks a lot for your effort in getting this sorted out!
thanks
Jon
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Freymann
I should be on IRC tomorrow from around 1300 until 1500 EST. It sounds like something may be wrong with the Line1 setup tab on the spa3000. I'll try to catch you on there tomorrow and we should be able to get that last part figured out.
Hi Jon. I reviewed the instructions again for Line 1 and all looks well, so maybe some of the defaults are different in sipura? I will drop by IRC tomorrow afternoon and hook up with you. I just connect a wireless phone there which lets me go back to the office and play (oops, I mean, work!)
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i just edited my post 2 posts up... I think I found your Line1 problem (as outlined in my last post above)
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i just edited my post 2 posts up... I think I found your Line1 problem (as outlined in my last post above)
Yep, changing the password to match the other two fields fixed the problem of not being able to use *98, etc. I now have my own voice on my voice mailbox. Very kewl!
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great - you should be able to dial an outside line now as well from your analog phones - your setup should be 100% now. I've updated the wiki page to correct the error on settig the password. Time to set up your routing and do some experimenting - if we can get some bug reports in in time maybe we can have the Telecom functioning better for 0810
Please post any bugs you have found so that we can discuss them and verify that it truely is a bug and not operating as intended. I already have a few bug reports i when it comes to changing on hold music.
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great - you should be able to dial an outside line now as well from your analog phones - your setup should be 100% now. Time to set up your routing and do some experimenting - if we can get some bug reports in in time maybe we can have the Telecom functioning better for 0810
Yes, may be able to experiment more tomorrow, not sure.
I do have one question which I'm still not clear on.
Connected the way you have illustrated.... when an incoming calls comes in, if nobody is at an orbiter to see/handle it, what happens? Does it just fall through to the built-in answering machine? Do the phones on the House Line ring? or is this dependant on the house mode and user modes? Basically, I'd like [specifically the wife demands] our house phones act just like they do now with no requirement to push buttons on the LMCE screen(s).
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make sure your phone line settings have your House Line checked for at least the Unarmed security mode (I would also do this for Entertaining and Armed at home). This will ensure that your analog phones always ring when you are home. It wouldn't hurt if House Line were checked for All security modes. This will ensure that it will ring regardless if you are home or not. This way the call can be answered as normal with no Orbiter screen interaction what so ever. The only difference should be between dialing 9 for an outside line(which I am going to experiment to get rid of having to dial a 9 - i think it is just tweaking the dialplan settings in FreePBX->Trunk)
On the other hand, if you uncheck House Line from Phone_Lines settings, then it will only ring on the orbiters (if they are checked).. The from the orbiters you can send the call to House Line, after which the analog phones will ring. This is a cool setup for the Entertaining security mode so you aren't disturbed while entertaining guests.
As you can see, there are a million ways to set this up and pretty much anything you can think of should be possible. The pain is setting it up the first time, but after that it should work pretty well
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Hi Jon.
Haven't had much free time lately to experiment with the sipura, but what I've noticed is:
-yes, the echo is there on my end, not sure if it's there at the caller's end too?
-it rings the house line and the orbiters just fine once I adjusted my call preferences and reloaded router. Very kewl.
-the "call display" on the orbiter shows me the phone number, but not the name? Does that pop up the # only?
-if I'm logged in as me, and I notice an incoming call that is for the wife, I can't direct the caller to her voicemail box, it only lets me send it to my voicemail box or the 'general' voicemail box
-the only access to the 'general' voicemail box is through pluto-admin web page or picking up a phone and *98 into mailbox 100.
-there's an option when viewing My Voicemail to activate the forward to my email address... this doesn't work
-Phone Book works fine and is quite handy
-outgoing calls from the orbiters work. How would one go about utilizing a microphone attached to the computer to be able to talk?
-playback of voicemail through orbiter on the tv works fine, although sometimes you have to click on the message line twice before it will start playing. Not a big thing
-after the orbiter disappears on an incoming call (and you did nothing with LMCE), the menu (as if you pressed F7) stays on the screen for a very long time. I have to grab the remote and clear the menu.
-I was able to customize my recorded message from a handset quite easily.
-I was able to retrieve my voicemail messages quite easily from a handset.
I found this page in the wiki too:
LinuxMCE telecom features
http://wiki.linuxmce.org/index.php/LinuxMCE_telecom_features
which may be useful.
We don't use the built-in answering machine just yet. The better half insists she doesn't want to have to flip on the tv to get her messages, she doesn't use the pluto-admin web page at all, and she doesn't want all the house phones hung off the sipura because she doesn't like the echo or having to dial 9 before a call which makes dialing *98 impossible.
In my testing, I've just grabbed a portable phone from the basement and connected that to the sipura and used that for the day. As I've mentioned to you, I split the phone line at the demarc point to feed the sipura and the rest of the house separately. This pleases the wife tremendously as nothing changed internally except call display on the orbiters and an alternate voice mail system.
Next time I'm able to play I will see if I can get freepbx to forward messages to my cell number. I don't think we have call forward on the house line so that would be one way to keep in touch with home calls while away.
This is really good stuff and your instructions regarding the setup were easy to follow and most helpful.
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Nice job... thanks for the update.
On a separate but equal front, I have an x100p card on it's way to my abode... I hope to get that going and draft a write-up. An issue that I foresee is with zaptel (now Dahdi). Dahdi has replaced zaptel with some decent updates. In order to upgrade to Dahdi, you need to re-compile asterisk (which as we know, will be a pain). Therefore, I plan to try the x100p card with an upgraded version of Zaptel (1.4.11). Now, you might be thinking: "1.4.11 is not the newest version...". You are correct. However, since the x100p does not come with hardware echo cancellation, I am going to use the Open Source Line Echo Canceller (OSLEC) to remedy the echo that will probably occur; the latest patch for zaptel as it pertains to OSLEC, is for zaptel version 1.4.11. I've installed the new zaptel and I've also installed OSLEC. Zaptel seemed to give weird errors, but I believe this is due to not having the card installed... Once I get the card (in about 10 business days), then I will report back my findings...
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Freymann,
I know you can do this with voip calls. Maybe some of the freepbx configs for voip forwarding would be useful?
just thoughts
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Freymann,
I know you can do this with voip calls. Maybe some of the freepbx configs for voip forwarding would be useful?
just thoughts
golgoj4, do you still get an echo with the spa-3102?
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Interesting conversation going on here, I also have an x100p and setup was relatively painless once I figured out what I needed. I had to create a blank /etc/zaptel.conf then do sudo apt-get install zaptel. I found that not creating that file ahead of time the install would fail out. From there it was just a matter of configuring my trunk, inbound, and outbound routes in the FreePBX config. I was going to do a wiki writeup on this, but at the time I did this nobody on the forums showed interest in pstn lines and lmce. I for one would prefer to keep my normal pstn line and use on of the providers like voipcheap to cover long distance since they are generally free for 300 minutes/month. I don't make that many long distance calls so for me this would be a nice alternative.
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Interesting conversation going on here, I also have an x100p and setup was relatively painless once I figured out what I needed. I had to create a blank /etc/zaptel.conf then do sudo apt-get install zaptel. I found that not creating that file ahead of time the install would fail out. From there it was just a matter of configuring my trunk, inbound, and outbound routes in the FreePBX config. I was going to do a wiki writeup on this, but at the time I did this nobody on the forums showed interest in pstn lines and LinuxMCE. I for one would prefer to keep my normal pstn line and use on of the providers like voipcheap to cover long distance since they are generally free for 300 minutes/month. I don't make that many long distance calls so for me this would be a nice alternative.
How was the echo with the x100p? Did you have to install oslec? If so, anything special beyond normal install procedures?
-Derrick
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Interesting conversation going on here, I also have an x100p and setup was relatively painless once I figured out what I needed.
I'm a tinkerer and wanted to see what asterisk was all about. I hope to start using the voicemail boxes a little more in the future.
To prepare on testing the forward voicemail to email setting, today I did this on the core:
sudo apt-get install postifx
Afterwards, I edited /etc/postfix/main.cf and put my ISP's outgoing mail server (SMTP) name beside "relayhost = "
and then reloaded postfix with:
sudo /etc/init.d/postfix reload
I still haven't had a chance to call in, direct my call to my voicemail and see what happens, but I don't see why it wouldn't now.
Jon: Perhaps this is something you may want to add to your sipura wiki page?
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One other thing, this is slightly off-topic, but I've been trying to connect a sip device to my asterisk box from another network and so far have been unsuccessful, I cannot even see any communication happen in my asterisk cli. Has anyone been able to get this type of setup working, the same device connects fine on my local lan. I tried setting up firewall rules in LMCE's webadmin, but they don't seem to be working. I know I need 5060 open and by default rtp uses 10000-20000 even ports only, and that all of these ports should be udp. If someone could shed some light on how to set these rules up in the webadmin, perhaps I'm something very silly the wrong way.
dlewis: echo is an issue on my sip devices, I need to investigate this more thoroughly, I can say Hi into the phone and I will hear it repeat almost a full second later. On the other end of the calls they do not hear any echo at all and report that voice quality is excellent. For now I've been ignoring the echo I hear, but it really is quite distracting when carrying on a lengthy conversation.
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One other thing, this is slightly off-topic, but I've been trying to connect a sip device to my asterisk box from another network and so far have been unsuccessful, I cannot even see any communication happen in my asterisk cli. Has anyone been able to get this type of setup working, the same device connects fine on my local lan. I tried setting up firewall rules in LMCE's webadmin, but they don't seem to be working. I know I need 5060 open and by default rtp uses 10000-20000 even ports only, and that all of these ports should be udp. If someone could shed some light on how to set these rules up in the webadmin, perhaps I'm something very silly the wrong way.
Do you have NAT set up correctly? What you might have to do is create an /etc/asterisk/sip_nat.conf file with the following lines:
1) nat=yes
2) externip=your.external.IPaddess (or externhost=your.external.hostname)
3) localnet=192.168.0.0/24 (assuming your network uses 192.168.0.x addresses).
Then "asterisk -rx sip reload" at the CLI.
dlewis: echo is an issue on my sip devices, I need to investigate this more thoroughly, I can say Hi into the phone and I will hear it repeat almost a full second later. On the other end of the calls they do not hear any echo at all and report that voice quality is excellent. For now I've been ignoring the echo I hear, but it really is quite distracting when carrying on a lengthy conversation.
Have you tried oslec? Look here: http://www.rowetel.com/ucasterisk/oslec.html . Let me know how it goes...
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I haven't tried OSLEC, but I did just turn on echotraining and ran ztmonitor and it seems to have almost completely cleared up the echo, I just hear a very very faint echo now.
I actually do have my sip_nat.conf file setup exactly as explained but I still cannot connect to my asterisk box. I really don't know what else to check or use as a debugging tool to nail down whether this is an asterisk configuration issue or a network issue.
While everyone seems to be interested in asterisk, I would love to see more LMCE wizard setup go along with phone lines, such as importing phone books or manually entering a phone book so that calls will route to where a user is logged in on the system or to their individual extensions. As I understand it this can be done through the webadmin, but it would be nice to have this in the initial setup. It would also be very cool if unknown numbers could be dumped to an IVR menu that the person setting up the system would be prompted to record a greeting for during setup.
Finally, I have a question, I'd like to know if this is currently possible, I'd like to create another IVR using an unannounced extension and pin number to check the status/control my LMCE system. This would be great for a few reasons: 1) You wouldn't need to expose your entire network to the outside to use the weborbiter when not on site. 2) You could control your home when away from a network or edge/3g connection from any telephone.
I'm not sure how to ease in the setup of this, but if scenarios could populate in the freepbx menu I'd imagine the setup could be fairly painless for the end user.
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I haven't tried OSLEC, but I did just turn on echotraining and ran ztmonitor and it seems to have almost completely cleared up the echo, I just hear a very very faint echo now.
I would try OSLEC... It's supposed to work wonders and completely remove the echo.
I actually do have my sip_nat.conf file setup exactly as explained but I still cannot connect to my asterisk box. I really don't know what else to check or use as a debugging tool to nail down whether this is an asterisk configuration issue or a network issue.
Join the IRC channel... Someone should be able to help. If you're trying to connect a hard-phone remotely, it might be very tough... IAX is best for doing that. If you're trying to connect a SIP soft-phone, this might be a soft-phone config issue...
While everyone seems to be interested in asterisk, I would love to see more LMCE wizard setup go along with phone lines, such as importing phone books or manually entering a phone book so that calls will route to where a user is logged in on the system or to their individual extensions. As I understand it this can be done through the webadmin, but it would be nice to have this in the initial setup. It would also be very cool if unknown numbers could be dumped to an IVR menu that the person setting up the system would be prompted to record a greeting for during setup.
I believe someone is already working on this...
Finally, I have a question, I'd like to know if this is currently possible, I'd like to create another IVR using an unannounced extension and pin number to check the status/control my LMCE system. This would be great for a few reasons: 1) You wouldn't need to expose your entire network to the outside to use the weborbiter when not on site. 2) You could control your home when away from a network or edge/3g connection from any telephone.
I'm not sure how to ease in the setup of this, but if scenarios could populate in the freepbx menu I'd imagine the setup could be fairly painless for the end user.
I don't know about this. This would require custom dial-plans with some custom code to connect to pluto for status...
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I'm considering ditching my X100P card for a Sipura device for two reasons. First, I actually need both a FXS and FXO card and I currently only have a FXO. Second, I have not been able to get my card on a dedicated IRQ so I'm struggling with serious latency issues. They have almost been ironed out, but I have had to do WAAAAY too much tweaking and the setup is not reliable enough for production use.
Can those of you using a Sipura verify for me that I could set it up to do what I want. I only have a single PSTN line and I'd like Asterisk to answer all calls and if the caller ID is recognized automatically route the call appropriately, if the caller ID is not recognized I'd like to dump the caller to an IVR where they can choose who they are calling for. I would have the internal PSTN phones configured as a single extension and understand that they would all ring together when selected, but ideally I do not want them to make a peep before asterisk has routed calls to them. A simple yes or no verification preferably from someone who is currently doing something similar will suffice, I have gotten a good handle on the workings of Asterisk and can handle the configuration of such a setup, I just need clarification before I spend the cash.
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los93sol:
The spa3000 works well, with the following caveats:
- Echo can be a problem. Since it doesn't use Zaptel drivers, echo cancellation will not work with this setup
- I haven't successfully "flashed" the hook yet (to answer call waiting for example), though I'm sure its a setting issue somewhere
Other than that, it works great with LMCE's call routing features etc.
Also - I just received a X100P and was going to add support for it under LMCE. Could you share your setup information with me to get me started on it? My goal is to have the X100P and the spa3000 as plug-and-play as possible under 0810.
thanks
Jon
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Jondecker:
Thanks for the info. I actually had full intentions of writing a wiki page for the installation of the X100P but have been tied up with other hardware at the moment. Please forgive any misinformation I'm writing this mostly from memory and a few notes I took when doing my install.
You should be able to get it working with the following instruction:
1) Create a blank zaptel.conf file in /etc/
NOTE: If this file does not exist the zaptel driver install will fail.
2) Open a shell and type: sudo apt-get install zaptel
3) In your shell type: genzaptelconf
This will populate the blank zaptel.conf file you created earlier
4) Now you are ready to write your zapata.conf file in /etc/asterisk I have attached both my zaptel.conf and my zapata.conf as working examples.
5) We should be able to run some diagnostic commands to be sure we're on the right track. You can run ztcfg -vvvv and you should get some output similar to the following:
Zaptel Version: 1.4.3
Echo Canceller: MG2
Configuration
======================
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
1 channels configured.
6) The manual part is done at this point, now we jump to the LMCE web-admin and navigate to Advanced>Phones Setup>Trunks>Add Zap Trunk.
7) My PSTN line is my only phone line so I route all traffic in and out of it so the only fields I filled out were :
Outbound Caller ID: Your Phone #
Maximum Channels: 1
Zap Identifier (trunk name): 1
8) Submit and apply your changes
9) This part I am fuzzy on since my current installation is simply ringing every phone in my house, whether it is a standard PSTN or a SIP phone. At any rate, I have a ring group I created to accomplish this, pm me if you want more details and I'll break it down but it is beyond the scope of a basic working setup. Anyways, we need to create and inbound route so click on Inbound Routes, I configured mine as follows:
Zaptel Channel: 1
Destination Ring Groups: RingAll <600>
You should configure your destination to your needs and based on how you want asterisk to work in your setup.
10) Submit and apply your changes
11) This is the last part, we need to create an outbound route so that asterisk knows when you use our zaptel trunk. Click on Outbound Routes. I configured mine as follows:
Route Name: OutPSTN
Dial patterns wizards: Local 7/10 and Emergency
Trunk Sequence: ZAP/1
12) Submit and apply your changes
13) Reload zaptel and verify our work from the CLI by loading up a shell and typing Asterisk -vvvvvvvvvvr, to reload zaptel we type zap restart, then check our card by typing zap show status. We should see some output like:
Description Alarms IRQ bpviol CRC4
Wildcard X100P Board 1 OK 0 0 0
ZTDUMMY/1 1 UNCONFIGUR 0 0 0
14) Finally we can verify the channels are created successfully by typing zap show channels and you should see something like:
Chan Extension Context Language MOH Interpret
pseudo from-zaptel default
1 from-zaptel default
That should be enough information to get you a basic working setup, if you have more questions please PM me. Also let me know if these instructions need to be tweaked to get it working and I will create a wiki page when the instructions work :)
PS. I could use some assistance with my alarm panel if you know anything about GSD
-
Jondecker:
Thanks for the info. I actually had full intentions of writing a wiki page for the installation of the X100P but have been tied up with other hardware at the moment. Please forgive any misinformation I'm writing this mostly from memory and a few notes I took when doing my install.
You should be able to get it working with the following instruction:
1) Create a blank zaptel.conf file in /etc/
NOTE: If this file does not exist the zaptel driver install will fail.
2) Open a shell and type: sudo apt-get install zaptel
3) In your shell type: genzaptelconf
This will populate the blank zaptel.conf file you created earlier
4) Now you are ready to write your zapata.conf file in /etc/asterisk I have attached both my zaptel.conf and my zapata.conf as working examples.
5) We should be able to run some diagnostic commands to be sure we're on the right track. You can run ztcfg -vvvv and you should get some output similar to the following:
Zaptel Version: 1.4.3
Echo Canceller: MG2
Configuration
======================
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
1 channels configured.
6) The manual part is done at this point, now we jump to the LMCE web-admin and navigate to Advanced>Phones Setup>Trunks>Add Zap Trunk.
7) My PSTN line is my only phone line so I route all traffic in and out of it so the only fields I filled out were :
Outbound Caller ID: Your Phone #
Maximum Channels: 1
Zap Identifier (trunk name): 1
8) Submit and apply your changes
9) This part I am fuzzy on since my current installation is simply ringing every phone in my house, whether it is a standard PSTN or a SIP phone. At any rate, I have a ring group I created to accomplish this, pm me if you want more details and I'll break it down but it is beyond the scope of a basic working setup. Anyways, we need to create and inbound route so click on Inbound Routes, I configured mine as follows:
Zaptel Channel: 1
Destination Ring Groups: RingAll <600>
You should configure your destination to your needs and based on how you want asterisk to work in your setup.
10) Submit and apply your changes
11) This is the last part, we need to create an outbound route so that asterisk knows when you use our zaptel trunk. Click on Outbound Routes. I configured mine as follows:
Route Name: OutPSTN
Dial patterns wizards: Local 7/10 and Emergency
Trunk Sequence: ZAP/1
12) Submit and apply your changes
13) Reload zaptel and verify our work from the CLI by loading up a shell and typing Asterisk -vvvvvvvvvvr, to reload zaptel we type zap restart, then check our card by typing zap show status. We should see some output like:
Description Alarms IRQ bpviol CRC4
Wildcard X100P Board 1 OK 0 0 0
ZTDUMMY/1 1 UNCONFIGUR 0 0 0
14) Finally we can verify the channels are created successfully by typing zap show channels and you should see something like:
Chan Extension Context Language MOH Interpret
pseudo from-zaptel default
1 from-zaptel default
That should be enough information to get you a basic working setup, if you have more questions please PM me. Also let me know if these instructions need to be tweaked to get it working and I will create a wiki page when the instructions work :)
PS. I could use some assistance with my alarm panel if you know anything about GSD
los,
could you create a wiki entry for this? I plan to install my x100p this upcoming weekend. Thanks!
-
Sure, I am planning on doing just that, I was hoping someone would run through the instructions I provided first so we can determine what changes if any need to go in. I wrote these instructions from memory of my install a few months ago.
-
Sure, I am planning on doing just that, I was hoping someone would run through the instructions I provided first so we can determine what changes if any need to go in. I wrote these instructions from memory of my install a few months ago.
I hope to do so this weekend.
-
Hi everybody,
EDIT SOLVED
custom-l(mce),102,1 (without brackets ) should have been.... custom-linuxmce,102,1..........(brain seeing something that wasn't there)...........sanity restored, dog safe !
many thanks to all contributors (especially jondecker)
{I've setup the spa 3102 as per jon's wiki, EXCEPT the dummy line,( my reasoning being that I allready have a working voip trunk) ,and things are almost working,- asterisk echo test ,voice mail ivr etc ,can ring out on both trunks from analog & voip trunks,from both orbiter phone /House Line (which rings analogue handset before ringing actual No.) and direct from analogue handset:-
BUT, incoming calls on pots line do not get any response from LinuxMCE, and worse do not ring the analogue handset, ie,I'm missing all incoming calls on House Line, ( voip incoming is 100%)
I've checked every setting,( both freePBX and spa) , numerous times,-....wizard/telecom/routing everything set to ring (both) extensions,..... both extensions and trunks show at cli "sip show peers"
.....my next action will be disembowelling the dog and examining the entrails....
.....any help would be greatly appreciated (especially by the dog ! )
................. & please state the obvious, they're the things I normally miss,(edit-told you!)}
thanks in advance,
Ian (and Bonny--the dog)
-
Jon,
Were you able to sort out not having to dial 9 to place an outside call? I was playing with this, but every change to the dial plans seemed to completely break the ability to dial out altogether. I'd like to get my phones setup to just dial like usual, any help is appreciated.
-
I haven't looked into it yet, though I plan to. In theory it shouldn't be all too hard.
The LMCE dialplan is created dynamically by a script, so editing it directly isn't the answer. I'll be sure to post back and update the wiki when I get it figured out
-
Next question I have is how do I get the system to answer unknown calls right away? Currently it rings all my phones a few times then they get the message, "To call everybody in the house press 0, to call XXX press 1, to call XXX press 2", then the call is routed accordingly. I've had a ton of complaints from people that they can't understand the Dr. Roboto voice after the lady says "To call everybody in the house press 0" so I'd like to record my own intro. with the options, is this possible? I know the setup seems to be sending callers to the lmce-custom script which apparently generates this, but since my callers can't even understand it I'd much rather just record my own. :)
-
dial:
*98 will get you to the Voicemail menu
*98XXX will get you to the Voicemail menu for local extension of XXX
From here, you can set your own custom greetings.
While on the subject, dial *999.. This is the IVR that calls you when there is a security breach.. Kind of nifty. These will actually be understandable once festival2 is rolled into LMCE
-
I haven't tried OSLEC, but I did just turn on echotraining and ran ztmonitor and it seems to have almost completely cleared up the echo, I just hear a very very faint echo now.
I actually do have my sip_nat.conf file setup exactly as explained but I still cannot connect to my asterisk box. I really don't know what else to check or use as a debugging tool to nail down whether this is an asterisk configuration issue or a network issue.
While everyone seems to be interested in asterisk, I would love to see more LMCE wizard setup go along with phone lines, such as importing phone books or manually entering a phone book so that calls will route to where a user is logged in on the system or to their individual extensions. As I understand it this can be done through the webadmin, but it would be nice to have this in the initial setup. It would also be very cool if unknown numbers could be dumped to an IVR menu that the person setting up the system would be prompted to record a greeting for during setup.
Finally, I have a question, I'd like to know if this is currently possible, I'd like to create another IVR using an unannounced extension and pin number to check the status/control my LMCE system. This would be great for a few reasons: 1) You wouldn't need to expose your entire network to the outside to use the weborbiter when not on site. 2) You could control your home when away from a network or edge/3g connection from any telephone.
I'm not sure how to ease in the setup of this, but if scenarios could populate in the freepbx menu I'd imagine the setup could be fairly painless for the end user.
weird...im looking into the same thing
so far i have discovered
1.the extensions are created by a script as jondecker indicated, so there a 2 approaches:
- modifiy the code to include this extension by default (the best option)
-Make use of the extras.conf (i believe) that isnt automagically modified by the existing scripts
2.look @ the iorbiter code to get an idea of how to get scenarios populated.
3. feed that into a custom menu linked to the asterisk AGI which would allow use to execute scenarios.*
4. have it properly integrated into lmce so its transparent. It should be updated any time an orbiter regen occurs
5.go 'wow thats cool'.
on point 4, a decision on the depth of how the menus would go needs to be made. I plan to test on 1 room, then expand it depending on what works.
where im at
1. been dissecting how everything is connected, so studying?
2. reading
3. testing out custom dial plans, still putting the asterisk book and other docs to use
4. no where yet. figure a working concept is in order 1st
5 the thought is cool...
lastly, you guys are really kicking ass on getting the pstn stuff integrated. How do we make stuff like these linksys ata's pnp?
-
You make device templates for them.
Look at the device templates for the existing phones. You'll find configuration and setup scripts for them, which basically do all the manual work of setting up the extension, loading any firmware required, setting settings on the phone automatically, basically anything you need to do to make the installation automatic.
Couple this with pnp records corresponding to the manufacturer MAC addresses for the phone (not an easy thing to find, you often have to guess, or contact the manufacturer), and you're set. Often you can infer possible device ranges if you have more than one of the unit, and eyeball the mac addresses.
-Thom
-
Thom - while on the subject... What detection methods are available for PCI cards?
-
Basically, we have one that we normally use. We talk to HAL.
HAL provides a device map of everything in the system, and we use this to detect PCI, and USB devices currently. There is a special case where we search for entire categories of devices, such as generic devices like optical disk drives. This is why when we had the HAL bug in 0810 a month ago, almost all pnp detection was affected.
But basically, the DHCPDevice table (which is what is filled when you fill a pnp section of a device template) acts as a filter. As devices are detected, they are sent a whole set of parameters as part of the event, some of them are filled, some are not..such as PCI or USB vendor ID/Model #, MAC address, etc... And the entries in the DHCPDevice table are matched against the incoming event as a conditional, anything matching is returned. If one device is returned, it merely asks to add the device. If multiple devices are returned, then a screen asking the user to pick a device is shown. The user selects a device, and the installation continues.
But technically, anything can fire a New Device Detected event, with a device template, to get the system to go into the routine of asking about and subsequently installing the device.
-Thom
-
Sure, I am planning on doing just that, I was hoping someone would run through the instructions I provided first so we can determine what changes if any need to go in. I wrote these instructions from memory of my install a few months ago.
los93sol, any updates on the x100p integration? How about the wiki site?
-
No updates, I have been waiting for confirmation that my directions posted in this thread have worked for others. Once I have that I will do a wiki, just don't want to put the information out there all willy nilly if it causes problems for people. I wrote from memory of what I had done several months earlier.
-
No updates, I have been waiting for confirmation that my directions posted in this thread have worked for others. Once I have that I will do a wiki, just don't want to put the information out there all willy nilly if it causes problems for people. I wrote from memory of what I had done several months earlier.
Are you currently using the x100p?
-
I switched to the Sipura but still have the x100p installed in my system just disabled for now.
-
how's the sipura? Any echo?
-
Hi everyone, last friday I've received a SPA3102 and during the weekend following jon's wiki I was able to make asterisk work with my PSTN for incoming calls but not for outgoing ones.
This is what I see in Asterisk's log.
05 05/31/09 21:33:53.594 NotifyRing from channel SIP/202-cc001fc0, callerid 202 to channel SIP/House Line-0080dec0, callerid <0x41802950>
05 05/31/09 21:33:53.618 NotifyHangup channel SIP/House Line-0080dec0, reason Unallocated (unassigned) number <0x41802950>
Any help please?
Thanks,
Gonzalo
-
Hi,
I install Sipura 3000 with an analog phone plug-in. It works in outgoing call with some problem. I made successfull call to my mobile from a MD. But the call not cutt off or drop even I have disconnect the call from my mobile. And, my analog phone don't ring when there is an incoming call to my phone line.
Please help.
-
First fire up asterisk cli with asterisk -vvvvvvvvvvvvvvvvr and make sure that incoming calls are even being routed to the analog extension. If they are then check the ring voltage settings on the Sipura and make sure they are set appropriately for your country.
-
First fire up asterisk cli with asterisk -vvvvvvvvvvvvvvvvr and make sure that incoming calls are even being routed to the analog extension. If they are then check the ring voltage settings on the Sipura and make sure they are set appropriately for your country.
The incoming call not working. Both MD and analog phone not ringing when I did a test call from my mobile. Form my mobile I hear 2 ringing tone then the call disconnected.
The outgoing call workd but with some problem. The call made to my mobile didn't hang-up / drop when I hang up the call after receiving it.
Please help.
-
Did you verify that the calls are not making it to your linuxmce core by using the asterisk cli as I suggested or are you just listening? You need to be more specific in exactly what is happening for me to help you troubleshoot.
-
Did you verify that the calls are not making it to your linuxmce core by using the asterisk cli as I suggested or are you just listening? You need to be more specific in exactly what is happening for me to help you troubleshoot.
I did as you told. And here is the result from terminal:
************************************************************************
Asterisk 1.4.10, Copyright (C) 1999 - 2007 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
== Parsing '/etc/asterisk/asterisk.conf': Found
== Parsing '/etc/asterisk/extconfig.conf': Found
Connected to Asterisk 1.4.10 currently running on dcerouter (pid = 7251)
-- Remote UNIX connection
Verbosity is at least 16
dcerouter*CLI>
***************************************************************************************************
On second though. Is the problems due to my regional setting in SPA3000? I am in Malaysia.
-
Did you attempt making a call to your house while the Asterisk CLI was open? It doesn't appear the call is ever even making it to your asterisk box which probably means your Sipura is not configured correctly. Verify that your settings are correct and that you Sipura is on your network and double check that your trunk is configured correctly in FreePBX.
If you need additional help beyond this point I can assist remotely, but will need access to your box and your Sipura.
-
Did you attempt making a call to your house while the Asterisk CLI was open? It doesn't appear the call is ever even making it to your asterisk box which probably means your Sipura is not configured correctly. Verify that your settings are correct and that you Sipura is on your network and double check that your trunk is configured correctly in FreePBX.
If you need additional help beyond this point I can assist remotely, but will need access to your box and your Sipura.
Hi los93sol ,
I will redo everything again. Will let you know the status.
Thanks for your advice.
Chin
-
If you're still having trouble after that let me know and I can provide some remote assistance to help you get this going.
-
If you're still having trouble after that let me know and I can provide some remote assistance to help you get this going.
Hi los93sol,
I manage to get the MD alert when incoming call. But there are 2 problems here:
1) The MD don't drop call when I disconnect call from my mobile
2) The analog phone don;t ring when incoming call.
Are there due to my regional ringging voltage setting? But, sadly. I am not able to find what is my country's ringging voltage. I did call up Malaysia telecom, but they failed to give me a correct answer.
And, how to enable remote assistance? From the forum i know the LMCE remote access server is currently ddown. Please advice.
-
For remote assistance it would be through giving telnet access and web admin access temporarily to your box rather than through the Remote Assistance feature of LMCE.
Now that your MD is ringing you should fire up the Asterisk CLI again and call your home. You should see in the CLI where your MD's extension is ringing and your analog extension. If you see the analog extension ringing now then we need to figure out your ring voltage. I had to adjust mine to get a normal ring on my phones since my phone almost sounded like it was trying to do a distinctive ring since the voltage was set too high by default.
I will do some digging and see if I can figure out anything for you as to how this should be set.
-
It looks like setting the regional settings to Trapezoid, 90Volts, and 20Hz is working for most people...wasn't able to find specifics on Malaysia though.
-
It looks like setting the regional settings to Trapezoid, 90Volts, and 20Hz is working for most people...wasn't able to find specifics on Malaysia though.
Hi los93sol,
Here is the status of the asterisk status:
*************************************************************************
Asterisk 1.4.10, Copyright (C) 1999 - 2007 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
== Parsing '/etc/asterisk/asterisk.conf': Found
== Parsing '/etc/asterisk/extconfig.conf': Found
Connected to Asterisk 1.4.10 currently running on dcerouter (pid = 6415)
Verbosity is at least 16
-- Executing [03-6280-9300@from-trunk:1] Set("SIP/House Line-08212700", "__FROM_DID=03-6280-9300") in new stack
-- Executing [03-6280-9300@from-trunk:2] GotoIf("SIP/House Line-08212700", "1 ?cidok") in new stack
-- Goto (from-trunk,03-6280-9300,4)
-- Executing [03-6280-9300@from-trunk:4] NoOp("SIP/House Line-08212700", "CallerID is "PSTN Call" <House Line>") in new stack
-- Executing [03-6280-9300@from-trunk:5] Goto("SIP/House Line-08212700", "custom-linuxmce|103|1") in new stack
-- Goto (custom-linuxmce,103,1)
-- Executing [103@custom-linuxmce:1] AGI("SIP/House Line-08212700", "pluto-gethousemode.agi") in new stack
-- Launched AGI Script /usr/share/asterisk/agi-bin/pluto-gethousemode.agi
-- AGI Script Executing Application: (Set) Options: (HOUSEMODE=1)
-- AGI Script pluto-gethousemode.agi completed, returning 0
-- Executing [103@custom-linuxmce:2] Goto("SIP/House Line-08212700", "103-hm1|1") in new stack
-- Goto (custom-linuxmce,103-hm1,1)
-- Executing [103-hm1@custom-linuxmce:1] Dial("SIP/House Line-08212700", "Local/200@trusted|15") in new stack
-- Called 200@trusted
-- Executing [200@trusted:1] Macro("Local/200@trusted-7519,2", "exten-vm|novm|200") in new stack
-- Executing [s@macro-exten-vm:1] Macro("Local/200@trusted-7519,2", "user-callerid") in new stack
-- Executing [s@macro-user-callerid:1] NoOp("Local/200@trusted-7519,2", "user-callerid: PSTN Call House Line") in new stack
-- Executing [s@macro-user-callerid:2] Set("Local/200@trusted-7519,2", "AMPUSER=House Line") in new stack
-- Executing [s@macro-user-callerid:3] GotoIf("Local/200@trusted-7519,2", "1?report") in new stack
-- Goto (macro-user-callerid,s,13)
-- Executing [s@macro-user-callerid:13] NoOp("Local/200@trusted-7519,2", "TTL: ARG1: novm") in new stack
-- Executing [s@macro-user-callerid:14] GotoIf("Local/200@trusted-7519,2", "0?continue") in new stack
-- Executing [s@macro-user-callerid:15] Set("Local/200@trusted-7519,2", "__TTL=64") in new stack
-- Executing [s@macro-user-callerid:16] GotoIf("Local/200@trusted-7519,2", "1?continue") in new stack
-- Goto (macro-user-callerid,s,23)
-- Executing [s@macro-user-callerid:23] NoOp("Local/200@trusted-7519,2", "Using CallerID "PSTN Call" <House Line>") in new stack
-- Executing [s@macro-exten-vm:2] Set("Local/200@trusted-7519,2", "FROMCONTEXT=exten-vm") in new stack
-- Executing [s@macro-exten-vm:3] Set("Local/200@trusted-7519,2", "VMBOX=novm") in new stack
-- Executing [s@macro-exten-vm:4] Set("Local/200@trusted-7519,2", "EXTTOCALL=200") in new stack
-- Executing [s@macro-exten-vm:5] Set("Local/200@trusted-7519,2", "CFUEXT=") in new stack
-- Executing [s@macro-exten-vm:6] Set("Local/200@trusted-7519,2", "CFBEXT=") in new stack
-- Executing [s@macro-exten-vm:7] Set("Local/200@trusted-7519,2", "RT=") in new stack
-- Executing [s@macro-exten-vm:8] Macro("Local/200@trusted-7519,2", "record-enable|200|IN") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("Local/200@trusted-7519,2", "0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] AGI("Local/200@trusted-7519,2", "recordingcheck|20090714-193153|1247625113.66") in new stack
-- Launched AGI Script /usr/share/asterisk/agi-bin/recordingcheck
recordingcheck|20090714-193153|1247625113.66: Inbound recording not enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing [s@macro-record-enable:5] NoOp("Local/200@trusted-7519,2", "No recording needed") in new stack
-- Executing [s@macro-exten-vm:9] Macro("Local/200@trusted-7519,2", "dial||tr|200") in new stack
-- Executing [s@macro-dial:1] GotoIf("Local/200@trusted-7519,2", "1?dial") in new stack
-- Goto (macro-dial,s,3)
-- Executing [s@macro-dial:3] AGI("Local/200@trusted-7519,2", "dialparties.agi") in new stack
-- Launched AGI Script /usr/share/asterisk/agi-bin/dialparties.agi
dialparties.agi: Starting New Dialparties.agi
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'admin' logged on from 127.0.0.1
dialparties.agi: Caller ID name is 'PSTN Call' number is 'House Line'
dialparties.agi: USE_CONFIRMATION: 'FALSE'
dialparties.agi: RINGGROUP_INDEX: ''
dialparties.agi: Methodology of ring is 'none'
-- dialparties.agi: Added extension 200 to extension map
-- dialparties.agi: Extension 200 cf is disabled
-- dialparties.agi: Extension 200 do not disturb is disabled
> dialparties.agi: extnum 200 has: cw: 0; hascfb: 0 [] hascfu: 0 []
> dialparties.agi: ExtensionState: 0
dialparties.agi: Extension 200 has ExtensionState: 0
-- dialparties.agi: Checking CW and CFB status for extension 200
-- dialparties.agi: DbDel CALLTRACE/200 - Caller ID is not defined
== Manager 'admin' logged off from 127.0.0.1
-- AGI Script dialparties.agi completed, returning 0
-- Executing [s@macro-dial:10] Dial("Local/200@trusted-7519,2", "SIP/200||tr") in new stack
-- Called 200
-- Local/200@trusted-7519,1 is ringing
-- SIP/200-0821b638 is ringing
-- Nobody picked up in 15000 ms
-- Executing [103-hm1@custom-linuxmce:2] Goto("SIP/House Line-08212700", "103-hm1-NOANSWER|1") in new stack
-- Goto (custom-linuxmce,103-hm1-NOANSWER,1)
-- Executing [103-hm1-NOANSWER@custom-linuxmce:1] Goto("SIP/House Line-08212700", "voice-menu-pluto-custom|s|1") in new stack
-- Goto (voice-menu-pluto-custom,s,1)
-- Executing [s@voice-menu-pluto-custom:1] Answer("SIP/House Line-08212700", "") in new stack
-- Executing [s@voice-menu-pluto-custom:2] Wait("SIP/House Line-08212700", "1") in new stack
== Spawn extension (macro-dial, s, 10) exited non-zero on 'Local/200@trusted-7519,2' in macro 'dial'
== Spawn extension (macro-dial, s, 10) exited non-zero on 'Local/200@trusted-7519,2' in macro 'exten-vm'
== Spawn extension (macro-dial, s, 10) exited non-zero on 'Local/200@trusted-7519,2'
-- Executing [h@macro-dial:1] Macro("Local/200@trusted-7519,2", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] ResetCDR("Local/200@trusted-7519,2", "w") in new stack
-- Executing [s@macro-hangupcall:2] NoCDR("Local/200@trusted-7519,2", "") in new stack
-- Executing [s@macro-hangupcall:3] GotoIf("Local/200@trusted-7519,2", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [s@macro-hangupcall:6] GotoIf("Local/200@trusted-7519,2", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] GotoIf("Local/200@trusted-7519,2", "1?theend") in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [s@macro-hangupcall:11] Hangup("Local/200@trusted-7519,2", "") in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'Local/200@trusted-7519,2' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'Local/200@trusted-7519,2'
-- Executing [s@voice-menu-pluto-custom:3] AGI("SIP/House Line-08212700", "pluto-callersforme.agi") in new stack
-- Launched AGI Script /usr/share/asterisk/agi-bin/pluto-callersforme.agi
-- AGI Script Executing Application: (NoOp) Options: (Finding if is a priority call for somebody)
-- AGI Script pluto-callersforme.agi completed, returning 0
-- Executing [s@voice-menu-pluto-custom:4] BackGround("SIP/House Line-08212700", "pluto/pluto-default-voicemenu") in new stack
-- <SIP/House Line-08212700> Playing 'pluto/pluto-default-voicemenu' (language 'en')
-- Executing [s@voice-menu-pluto-custom:5] Set("SIP/House Line-08212700", "TIMEOUT(digit)=10") in new stack
-- Digit timeout set to 10
-- Executing [s@voice-menu-pluto-custom:6] Set("SIP/House Line-08212700", "TIMEOUT(response)=20") in new stack
-- Response timeout set to 20
dcerouter*CLI>
-
It looks like setting the regional settings to Trapezoid, 90Volts, and 20Hz is working for most people...wasn't able to find specifics on Malaysia though.
Hi los93sol,
I managed to fixed the problem. Apparently it's due to the "disconnect tone" not properly setup in SAP3000. I manage to get Malaysia PSTN tone info and fill in the correct tone. It's work now.
Thanks for your helps in this.
Chin
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It looks like setting the regional settings to Trapezoid, 90Volts, and 20Hz is working for most people...wasn't able to find specifics on Malaysia though.
Hi los93sol,
I missed out something. The analog phone extension not ring. I already go to the wizard-device-phonelines and check "House Line" to ring in all scenario but the analog phone still not ring.
Please help.
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Hia,
i'm trying to setup the pstn analog line using the linksys spa-3102, using as reference the
http://wiki.linuxmce.org/index.php/Sipura/Linksys_spa3000_pstn_interface page.
But when following the following section, i got trouble:
Next, go to Wizard->Devices->Phone Lines (on the left pane in the LMCE web admin, not in FreePBX). We are going to add a dummy line (NOTE: this is a temporary hack for now! I won't go into too many details other than saying that it will allow you to use the "Settings" link next to the listing to do some call routing on your pstn line!) Use the dropdown to select broadvoice. Once you do this, you will see a form to fill in some data. Just put whatever you want in the fields, they won't be used with this hack! After you are done, you will see it listed as a phone line - use the "settings" link next to it to do call routing depending on security mode! (Note: After creating this "dummy" phone line, go back to the FreePBX admin, and look at the Incoming Route for the broadvoice line. Look at the Custom App option at the bottom of the page. It should contain that same custom-linuxmce,102,1. If it does not, go back to the Incoming Route for the House Line, and change its custom app line to be the same as this one! From my tests, it should be the same (though the 3 digit number can change, so check this to be sure).
Using the settings lik next to to it to do call routing... there is no Custom App option.
As this wiki was modified jan 26, 2009, i guess there were changes.
Does anyone knows how to use the spa-3102 with analog lines only and having the pap2t-na as extensions (using analog phone devices as softphones)?
TIA,
Paulo