[asterisk-users] Routing premature media to the calling channel

Alex Balashov abalashov at evaristesys.com
Sun Mar 25 04:37:57 CDT 2012


I think I may have misunderstood your initial question, sorry.

You are looking for Asterisk to directly pass through the early media from upstream?  Why would it do that?

--
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Atlanta, GA 30030
Tel: +1-678-954-0671
Web: http://www.evaristesys.com/, http://www.alexbalashov.com

Leandro Dardini <ldardini at gmail.com> wrote:

>The asterisk box has only one interface. I am capturing all the traffic on
>the box and the only audio traffic is from the provider to the asterisk box.
>
>Obviously if I set progressinband=yes, then I get the ringing tone from the
>asterisk box, but no the audio from the provider I was looking for.
>
>Leandro
>
>2012/3/25 Alex Balashov <abalashov at evaristesys.com>
>
>> Are you absolutely sure that nothing is coming out, even on a different
>> interface than the one on which you are capturing? Are you capture on the
>> Asterisk server and not the receiving host?
>>
>> Secondly, are you absolutely positive that something is supposed to be
>> coming out? 183 does not logically imply or mandate backward early media,
>> though 183+SDP is generally used as a convention to indicate that it is
>> about to be sent.
>>
>> --
>> Alex Balashov - Principal
>> Evariste Systems LLC
>> 235 E Ponce de Leon Ave
>> Suite 106
>> Atlanta, GA 30030
>> Tel: +1-678-954-0671
>> Web: http://www.evaristesys.com/, http://www.alexbalashov.com
>>
>> Leandro Dardini <ldardini at gmail.com> wrote:
>>
>> All NAT and firewall problems are already been excluded. All peers are on
>> public IP address and no firewall is active between them. The missing
>> routing of the audio path to the peer has been checked with tcpdump ...
>> nothing is coming out from the asterisk box.
>>
>> Leandro
>>
>> 2012/3/25 Alex Balashov <abalashov at evaristesys.com>
>>
>>> I assume you have ruled out NAT and firewall issues?
>>>
>>> Between those two, 99% of the reasons why something may not be routed
>>> somewhere correctly are accounted for.
>>>
>>> If you don&apos;t know, your best bet is to take a packet capture or SIP
>>> debug on the Asterisk server and find out where that early media is going.
>>>
>>> --
>>> Alex Balashov - Principal
>>> Evariste Systems LLC
>>> 235 E Ponce de Leon Ave
>>> Suite 106
>>> Atlanta, GA 30030
>>> Tel: +1-678-954-0671
>>> Web: http://www.evaristesys.com/, http://www.alexbalashov.com
>>>
>>>
>>> Leandro Dardini <ldardini at gmail.com> wrote:
>>>
>>> Hello,
>>> I have a problem with premature media and inband progress audio. I am
>>> using the latest 1.8.10.1 and this is the setup:
>>>
>>> soft phone --- asterisk --- SIP provider
>>>
>>> The number I call is giving back some hints via inband audio I am not
>>> able to ear from the soft phone. They stop on the asterisk and are not
>>> routed down the path to the sip phone.
>>>
>>> The SIP part is simple:
>>>
>>> soft phone -> asterisk: INVITE
>>>
>>> asterisk -> soft phone: TRYING
>>>
>>> asterisk -> provider: INVITE
>>>
>>> asterisk -> soft phone: 180 RINGING
>>>
>>> provider -> asterisk: 183 SESSION PROGRESS
>>>
>>> provider -> asterisk: AUDIO
>>>
>>> Unfortunately the AUDIO received from the provider by the asterisk box is
>>> not sent to the soft phone.
>>>
>>> I think I have tried every combination of progressinband and
>>> prematuremedia, without success.
>>>
>>> How can I made the audio received from the provider to the asterisk be
>>> transmitted to the soft phone?
>>>
>>> Thank you
>>>
>>> Leandro
>>>
>>>
>>>
>>> --
>>> _____________________________________________________________________
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>>               http://www.asterisk.org/hello
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>>
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>               http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>--
>_____________________________________________________________________
>-- Bandwidth and Colocation Provided by http://www.api-digital.com --
>New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
>
>asterisk-users mailing list
>To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120325/2f0dde7a/attachment.htm>


More information about the asterisk-users mailing list