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

Sun Mar 25 04:51:34 CDT 2012

As far as I know, this is not the general tendency of any B2BUA that generates such media independently.  However, I could be mistaken.

--
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:

>I want to have the early media to pass from the provider down to the soft
>phone because it contains important information about the call, like "Your
>call cannot go through, please try your call again " ... The provider is
>giving this info via early media, just after the 183 SESSION PROGRESS.
>
>Leandro
>
>2012/3/25 Alex Balashov <abalashov at evaristesys.com>
>
>> 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
>>>>
>>>>
>>>>
>>>> --
>>>> _____________________________________________________________________
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>>>
>>>
>>> --
>>> _____________________________________________________________________
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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
>>
>
>--
>_____________________________________________________________________
>-- 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
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