Internal codec don't change

pnaves

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#1
Hi Folks,

I already installed ilbc and g729 codec into my 1.5.2 elastix and when I make an external call through my voip provider the codec ilbc works fine, I can see the ilbc codec in use type (sip show channels) but when I do internal calls, between extensions, the elastix only be able to use codecs ulaw and alaw. I change extension`s configuration to disallow=ulaw&alaw and allow=ilbc but when I'm try to call sip phone brings me this message "Call Failed: Not Acceptable here" I can't figure out what is it happening because the ilbc codec works good between my pbx and my voip provider but between my extension only ulaw and alaw are being accepted.
 

Patrick_elx

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#2
are both your extensions equipped and authorizing the ilbc codec?
are both extensions setup with allow ilbc in the Elastix GUI?

When you call outside it's going through asterisk first. Are you sure that you are ilbc in both legs from the extension to asterisk, then from asterisk to the provider?

Have you allowed ilbc in sip_general_custom.conf?
 

pnaves

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#3
Yes, both extensions were setup? I configured in the Elastix GUI.
This extensions are using x-lite I suppose that x-lite support ilbc.
However, I didn't setup sip_general_custom.conf because I found "sip_general_custom.conf" in directory "/etc/asterisk" but is empty. I had the same problem with other files mentioned in "Elastix without tears" for example zaptel.conf, zapata-channels.conf. Should I include some command lines into this blanks files? What's happening with my distribution? Is it normal?

Thank you!
 

Patrick_elx

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#4
I think x-lite does not have ilbc allowed by default.

You need to go to configuration, advanced, audio codec and allow it.

Also if you go in the asterisk CLI (asterisk -rvvv)
enable the SIP debug by:
sip set debug

and you should see the SIP negotiation at the beginning of the call where you will see what codec the extension is accepting.


to stop the debug:
sip set debug off
 

pnaves

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#5
Thank you so much! You're right! I have never seen the advanced button in X-lite until You told me and ilbc codec isn't a default value. I fixed this problem but perhaps you could clarify why the configuration files in elatix 1.5.2 are empty. I tried to search but I couldn't find anything.
 

Bob

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#6
pnaves,

The sip_general_custom is empty as it is for custom entries that can be placed in the file.

On the whole, there are three main config files that are used for each configuration area e.g. IAX, SIP, Extensions etc..... so for instance SIP in the past has had the following:

sip.conf - mainly for global SIP settings - usually static and not changed
sip_additional.conf - mainly for freepbx to write its macros that keep the whole system running
sip_custom.conf - for your own custom entries

Each of this files are chained, so really they are viewed as one config file, but they are broken up to make it easier to manage....For instance, you can sip.conf and find that it INCLUDES these other two file names, and may include other filenames.

Whilst I have given you a simple example, you will find that it also contains INCLUDES for sip_general_custom, and possibly others....

The reason why there is a sip_custom, is that in general, you can place your own custom entries and know that when a Elastix/Freepbx upgrade comes along that it leaves this file intact, whereas the sip.conf and the sip_additional may be updated/reset back to default (losing your custom entries.

The reason Zaptel and similar files are empty is that ELastix as per the 1.5.2 release has moved to DAHDI in line with Asterisk requirements (Dahdi replaced Zaptel). The Dahdi config files and location have changed (e.g. the original /etc/Zaptel.conf is now called /etc/dahdi/system.conf). Zapata-channels is now /etc/asterisk/dahdi-channels.conf and some parts are split into other files, but generally start with Dahdi.... The old files are kept for legacy reasons, and I believe just need to exist empty. I am sure that they will disappear later down the track when everyone has moved to Dahdi....

Have you got the latest Elastix without Tears, as I believe Ben has put some updates in about DAHDI recently. I haven't had a chance to look, but it might be worth getting the latest version.

Now, one other point...generally there is no reason to edit the files directly, almost everything is accessible via the GUI, even setting the codecs...I understand that you have documents (e.g. ELastix without Tears, forums info etc), but many are now getting out of date. It is well worth learning the relationship between Freepbx and config files, as it makes it easier to translate what some of these older (or pure Asterisk) documents are trying to say.

Regards

Bob
 

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