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

Leandro Dardini ldardini at gmail.com
Sun Mar 25 04:35:45 CDT 2012


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
>>
>>
>>
>> --
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>
>
> --
> _____________________________________________________________________
> -- 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
>
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
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