21 Aug
2007
21 Aug
'07
9:18 a.m.
On 08/20/07 21:05, Kelvin Williams wrote:
THANK YOU!!!!!
The only problem here is the vendor, Arris Interactive believes their
products are the greatest. And "officially do not support SIP Deployment."
Could you point me in the direction of some documentation as to how I can
drop the 183 and generate a 182 instead?
generating other reply code is a bit hard, needs some coding. Dropping
183 is easy:
http://openser.org/dokuwiki/doku.php/core-cookbook:devel#drop
Cheers,
Daniel
Thanks,
kw
-----Original Message-----
From: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro]
Sent: Monday, August 20, 2007 12:30 PM
To: kelvin-lists(a)williamschadwell.com
Cc: users(a)openser.org
Subject: Re: [OpenSER-Users] Changing Audio Ports
Hello,
the sip device is broken, should update the destination port and IP from
the 200 ok reply. The best is to ask the vendor to fix it. Think about
para;lel forking, there could be many 183 coming from different
addresses and ports, what would do the device?
Temporary solution is to drop any 183 or to use perl substitutions for
such replies (all to be done in onreply_route).
Cheers,
Daniel
On 08/18/07 07:24, kelvin-lists(a)williamschadwell.com wrote:
In a previous query I asked if someone could shed
some light on as to why
my endpoints do not receive any audio when the call is redirected to an
announcement server.
After a lot of testing, I believe I have found the problem.
When I initiate a call from my end point the end point advises the callee
as to the port the RTP traffic will be present. When the call is handed
to my PSTN gateway the Gateway responds with its port for RTP in a 183
Session Progress.
When that call fails (due to timeout) we want to send it over to Asterisk
where an announcement will be played to the caller--however the caller
never hears it. The traces show that Asterisk advertises its RTP on a
different port that that of the PSTN Gateway. Some of my endpoints (Cisco
IP Phone 7940 and Sipura devices) see and listen for the audio on the new
advertised port, however my Arris EMTAs do not, it appears as though they
are still "tuned in" to the original audio port advised by the PSTN
gateway.
My question, is is possible to strip away the "m=audio 22040 RTP/AVP 0 8
18 101." from the SIP message? I would like to strip it away in the event
of a 183 from my gateway (that advertises the port), but pass it when the
call is actually answered.
If it is not possible to strip the RTP port information away from the
message, what would be the best way in handling a situation like this.
Many thanks in advance.
kw
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