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

[Alex Balashov]

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
>>
>>
>>
>> --
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>
>--
>_____________________________________________________________________
>-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>               http://www.asterisk.org/hello
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