external phone extension

[thomasAT]

hello forum,
i have a question about the external phone extension in askozia 2.0. for testing reasons i used it to get connected to the wbdemo@zipdx.com and i got the announcement that i am connected in narrowband. but my polycom is connected to the askozia box in wideband. is there any possibility to define the codecs that should be used for external phone extensions?

thanks a lot!

thomas

[Facundo Ameal]

In the "Providers" tab, there is a Codecs section. There you can set only the one you want.

Cheers.

[thomasAT]

thanks for the answer, but i am calling "directly" from the askozia box a sip-uri as external extention, the call is not going via the sip provider, so the provider settings are not changing this call.
some other ideas?

thank you very much!

thomas

[Facundo Ameal]

Try putting this

Code:

disallow=all
allow=g722

in Advanced -> SIP section in the Manual Attributes text box. Those are general options for SIP. In the allow parameter, you can put the codec you want to use. Take into account, that putting what I've said will ONLY let your phones use g722.

Or, create a Provider and define the codecs you want, you don't need to register. I think this is the best option.

Cheers.

[Michael]

This seems to be a problem with Asterisk 1.6.1 not being able to define which codecs should be used for direct URI dialing. If someone knows how to configure that, please post it here so I can add those codec options to the outward dial settings.

[Facundo Ameal]

This seems to be a problem with Asterisk 1.6.1 not being able to define which codecs should be used for direct URI dialing. If someone knows how to configure that, please post it here so I can add those codec options to the outward dial settings.

I think that Asterisk use the general SIP config when URI dialing. Or, you can do some hacking to set the codec in dialplan. I put an Asterisk dialplan example below. This should do the trick but I canoot figure out where should it be in the Askozia webGUI.

Code:

exten => _9X.,1,Set(__SIP_CODEC=g722)
exten => _9X.,2,Dial(SIP/${EXTEN:1}@sip.myprovider.com,300)

Hope it helps you make the improvement.

Cheers.

[giovanni.v]

This seems to be a problem with Asterisk 1.6.1 not being able to define which codecs should be used for direct URI dialing.

I believe that adding a default codecs selection, according to real translation capabilities, in sip.conf [general] section may help.

Never tried to play in but accordind to the quite nonexistent  documentation there are couple of channel variables to control the sip codecs offered:
- SIP_CODEC_OUTBOUND for the outbound call leg (http://svnview.digium.com/svn/asterisk?view=revision&revision=186635). According to the aforementioned documentation seems that the codec set using this variable must be one of the allowed codecs in the general settings.
- SIP_CODEC for the first call leg.

[Facundo Ameal]

- SIP_CODEC for the first call leg.

But if you set it this whay __SIP_CODEC=g722 it inherits to the second leg, I believe.

[giovanni.v]

But if you set it this whay __SIP_CODEC=g722 it inherits to the second leg, I believe.

I did not mean to contradict what you said, I had not seen yet.
However they strictly are different variables, leg does not refer to a channel and a bridged channel has (almost) two legs: one inbound an one outbound.  

[Facundo Ameal]

I did not mean to contradict what you said, I had not seen yet. 

Never thought that. I'm just trying to help the heroes

One more thing, perhaps, this two undocumented variables could be used for fax, somehow... Perhaps not... It's just a thought that came to my mind.