Author Topic: Usage of Telecom  (Read 4553 times)

posde

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Re: Usage of Telecom
« Reply #15 on: October 25, 2008, 03:46:15 pm »
dlewis,

thank you for your pointers. Especially the part with sip_nat.conf is clearly described in the wiki ;)

As this is ONLY relevant for calls leaving your network, and calls coming into your network, I don't think that SCCP is relevant here. At least I don't know an ITP providing SCCP services.

rgds
Oliver

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Re: Usage of Telecom
« Reply #16 on: October 26, 2008, 11:00:02 am »
Hi,

Does someone has experience with orange livebox, what is h.323?

Dave

jeangot

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Re: Usage of Telecom
« Reply #17 on: October 29, 2008, 07:01:05 am »
Hi,

I used asterisk before as the only phone service for my house (meaning i don't have a pots line anymore), and I now use Asterisk in linuxmce. It is very reliable and does not crash. I was able to setup emailing of voicemails via the Freepbx admin pannel. You can even tell it to email the voicemail and delete it from the mailbox. I was meaning to try and setup fax reception, which seems to be available in freepbx, but didn't get around to it yet.
There also seems to be a linuxmce event for receiving new voicemails and even one for when the mailbox is full, so it's possible to trigger certain events based on voicemail reception from within lmce (such as turning on a light somewhere for example).

The only problem  face is that when the phone rings while I am watching MythTV or a Video, the video or Tv will systematically stop and exit, instead of pausing, and simplephone will not display, which is very frustrating.
Other users do not seem to hav ethis problem and I did not find any difference with my system when comparing to other users in a thread a while ago. I was hoping that this may be fixed in 0810 when it comes out...

Dave: to answer your question, H323 is a different protocol for the transmission of voip calls, like sip or IAX. SIP and IAX are supported in Asterisk natively (hence also in Linuxmce) but H323 is not without special plugins. Also, less and less carriers are supporting H323, and while it is, in my opinion, a superior protocol from a telecom perspective, sip is better suited to deal with NAT situations of end users.

bulek

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Re: Usage of Telecom
« Reply #18 on: October 29, 2008, 04:31:05 pm »
Hi,

I used asterisk before as the only phone service for my house (meaning i don't have a pots line anymore), and I now use Asterisk in linuxmce. It is very reliable and does not crash. I was able to setup emailing of voicemails via the Freepbx admin pannel. You can even tell it to email the voicemail and delete it from the mailbox. I was meaning to try and setup fax reception, which seems to be available in freepbx, but didn't get around to it yet.
...

Hi,

can you please describe your procedure on Wiki, I'm sure it will ge helpful to other users. I was also considering to dig in to send voicemails on email, but didn't start yet. Your description would help a lot...

Thanks in advance,

regards,

Bulek.
Thanks in advance,

regards,

Bulek.

samuelmukoti

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Re: Usage of Telecom
« Reply #19 on: October 30, 2008, 05:54:38 am »
Hi,

I used asterisk before as the only phone service for my house (meaning i don't have a pots line anymore), and I now use Asterisk in linuxmce. It is very reliable and does not crash. I was able to setup emailing of voicemails via the Freepbx admin pannel. You can even tell it to email the voicemail and delete it from the mailbox. I was meaning to try and setup fax reception, which seems to be available in freepbx, but didn't get around to it yet.
...

Hi,

can you please describe your procedure on Wiki, I'm sure it will ge helpful to other users. I was also considering to dig in to send voicemails on email, but didn't start yet. Your description would help a lot...

Thanks in advance,

regards,

Bulek.


The nice thing is that LMCE includes nice & mature technologies like Asterisk & FreePBX, so their well documented else where too.. Because of this the voicemail -> email setup - its quite easy and is on the "Extensions module"  when ever you create/edit currenly existing extensions you can define this kind of rules - natively supported by FreePBX.  For extensions that have been automagicaly been provisioned by LMCE, look for extensions start with "Px_xxx" (i think, away from my LMCE box ATM)

here's a link: http://www.freepbx.org/support/documentation/module-documentation/extensions

hope that helps,

Sam


bulek

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Re: Usage of Telecom
« Reply #20 on: October 30, 2008, 10:58:47 am »

The nice thing is that LMCE includes nice & mature technologies like Asterisk & FreePBX, so their well documented else where too.. Because of this the voicemail -> email setup - its quite easy and is on the "Extensions module"  when ever you create/edit currenly existing extensions you can define this kind of rules - natively supported by FreePBX.  For extensions that have been automagicaly been provisioned by LMCE, look for extensions start with "Px_xxx" (i think, away from my LMCE box ATM)

here's a link: http://www.freepbx.org/support/documentation/module-documentation/extensions

hope that helps,

Sam


Hi,

thanks for info... I know about FreePBX, but there is currently a bit of a problem to setup voicemail via email properly under LMCE (correct me if I'm wrong).  Of course, there is easy PBX of setting voicemail & email for existing extensions, but under LMCE only physical extensions appear under FreePBX and not extensions for users...

Under LMCE, every user gets its internal extension, that starts from 300... And those extensions don't appear in FreePBX. So when you try to use Call routing from Web-Admin, you can only select to go to voicemail for existing USERS (not extensions). So that's why I'm interested exactly what you did...

If you take a look at web-admin interface, then you'll see that each user has a tick for Email-Voicemail, but nowhere to specify email address. And in /etc/asterisk/voicemail.conf there are entries for each user, but no email address is specified - which is expected cause I cannot see where to enter it...

But what I tried is to add email address manually to /etc/asterisk/voicemail.conf and it doesn't work. So that's why I'm interested in your setup details...

Thanks in advance,

regards,

Bulek.
Thanks in advance,

regards,

Bulek.

randomfeelings

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Re: Usage of Telecom
« Reply #21 on: October 30, 2008, 02:35:48 pm »
I'm trying to use it daily, although have always a bit of hazardous expectation whether Asterisk will die or not (since I'm using Cisco 7970 this happens quite often)...

Oh, that's a shame, just bought a couple of 7970s so I could use them as orbiters also! Is anyone else experiencing this with 7970s, is there a known problem?

tschak909

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Re: Usage of Telecom
« Reply #22 on: October 30, 2008, 02:37:32 pm »
it doesn't happen very often, maybe for me once every couple of months, typically restarting asterisk fixes it. It has to do with a small bug in chan_sccp.

-Thom

hari

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Re: Usage of Telecom
« Reply #23 on: October 30, 2008, 05:52:28 pm »
we should switch from sccp to sip...

best regards,
Hari
rock your home - http://www.agocontrol.com home automation

tschak909

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Re: Usage of Telecom
« Reply #24 on: October 30, 2008, 06:12:48 pm »
so long as we keep the feature set and orbiter push works, i'm game.

-Thom

hari

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Re: Usage of Telecom
« Reply #25 on: October 30, 2008, 06:27:55 pm »
orbiter works on sip. I've tested that on 0704. There is only one difference, the phone cannot be provisoned with chan_sccp any more, so we have to create proper SEPxxx.cnf.xml files with the lines and such (like for the 7960).

We also have to use 7.0.2 SIP firmware iirc, as the newer ones had problems with asterisk.

best regards,
Hari
rock your home - http://www.agocontrol.com home automation

posde

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Re: Usage of Telecom
« Reply #26 on: October 30, 2008, 08:00:29 pm »
I assume the user does not really care, if the phone runs SIP or SCCP, as long as it works.

I don't know, if it is the protocol, or the phone, but I have two phones on my desk. A Snom 360 running SIP and a Cisco 7970 running SCCP. The Cisco phone rings 1-2 seconds before the Snom. That's the only difference I have seen so far.

rgds
Oliver

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Re: Usage of Telecom
« Reply #27 on: October 30, 2008, 08:16:26 pm »
i had no "speed" problems when using the 7970 with sip and asterisk :-)
rock your home - http://www.agocontrol.com home automation

cesarscav

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Re: Usage of Telecom
« Reply #28 on: November 03, 2008, 03:41:15 am »
Does someone have it run it on 7.10 rc2 AMD64.?
no system recordings sounds. No voicemail answer, no music on hold, no IVR heard, no system sounds at all
any ideas on this?
every thing else works great phone calls in and out.

 

jeangot

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Re: Usage of Telecom
« Reply #29 on: November 03, 2008, 05:23:02 pm »
Cesarscav,

I had the same problem on an AMD64. It had something to do with the way Asterisk uses ztdummy as a timer. My "solution" and I wouldn't call it a real solution was to rename:
/lib/modules/2.6.22-14-generic/misc/ztdummy.ko
into
/lib/modules/2.6.22-14-generic/misc/ztdummy.ko.noload
on the core, that way this library doesn't get automatically loaded by the system/asterisk anymore. Then reboot, and for me it worked, I had prompts in Asterisk.

I hope it helps you as well.

Bulek:
I'm not sure exactly if I understand your problem correctly. For me to enable email forwarding on my user voicemail, I simply click on Telecom/My Voicemail in the LinuxMCE admin, this brings up a freepbx page where I click on Settings. There you can enter an email address after checking the box "Email notification enable".
Does that not work for you?

Jean