Sipura 3000 setup for pstn analog lline (NOW WORKING! TESTERS NEEDED!)

[dlewis]

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...

[los93sol]

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.

[dlewis]

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...

[los93sol]

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.

[jondecker76]

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

[los93sol]

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

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

[dlewis]

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

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!

[los93sol]

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.

[dlewis]

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.

[maybeoneday]

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)

[los93sol]

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.

[jondecker76]

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

[los93sol]

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. 

[jondecker76]

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

[golgoj4]

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?