[Serusers] Granstream AtA g729 codec --- help please

[Greger V. Teigre]

If ser is transaction stateful (t_reply) and you put setflag(accflag) so 
that it is reached once for all messages, both new dialog-creating and 
loose routed, you should be fine. Double for all calls should not happen.
g-)

ravi reddy wrote:
>
>
> ---------- Forwarded message ----------
> From: *ravi reddy* <mravikreddy at gmail.com <mailto:mravikreddy at gmail.com>>
> Date: Aug 29, 2006 9:56 AM
> Subject: Re: [Serusers] Granstream AtA g729 codec --- help please
> To: "Greger V. Teigre" <greger at teigre.com <mailto:greger at teigre.com>>
>
> Thanks Greger for asking me,
>
>       I want a small information regarding billing of sip calls (even 
> though it is not related SER)
> please make a note to me :-)
>
> here in radacct i am getting more than one record for every call ; 
> some folks told that i need some perl script to format all the 
> database entries and write a fresh copy so that i can get one record 
> for one call which is easy for billing.
>
>           So, what i need a suggestion from you is
>
> 1) Do i need to learn perl language and write the script ?.
>
> 2) or is there any other way to control the  overflow of start stop 
> messages  in to SER ...?.
>
> If you know please tell me .
>
>                                          Thank You.
>
>
> Regards,
> Ravi.
>
>
>
>
>
>
>
>
>
>
>
>
>
> On 8/28/06, *Greger V. Teigre* <greger at teigre.com 
> <mailto:greger at teigre.com>> wrote:
>
>     Good to hear!
>     g-)
>     PS! No need to "sir"-me. Keep it informal, I'm Greger ;-)
>
>
>     ravi reddy wrote:
>>     Dear Sir,
>>
>>          Thanks for your reply. i found the problem  :-)
>>
>>     the problem is itself in my pstn gateway that they have to
>>     configure to allow my SER sending g729 codecs ,
>>
>>     i came to know from the folling grep message.
>>
>>     v=0.
>>     o=32331001 8000 8001 IN IP4 192.168.0.74 <http://192.168.0.74>.
>>     s=SIP Call.
>>     c=IN IP4 81.21.33.35 <http://81.21.33.35>.
>>     t=0 0.
>>     m=audio 60040 RTP/AVP 18 4 99 2.
>>     a=sendrecv.
>>     a=rtpmap:18 G729/8000.
>>     a=rtpmap:4 G723/8000.
>>     a=rtpmap:99 iLBC/8000.
>>     a=fmtp:99 mode=20.
>>     a=rtpmap:2 G726-32/8000.
>>     a=ptime:20.
>>
>>
>>     #
>>     U 81.21.33.35:5060 <http://81.21.33.35:5060> -> 81.21.34.34:5068
>>     <http://81.21.34.34:5068>
>>     SIP/2.0 501 Not Implemented.
>>     Via: SIP/2.0/UDP 192.168.0.74:5068
>>     <http://192.168.0.74:5068>;rport=5068;received= 81.21.34.34
>>     <http://81.21.34.34>;branch=z9hG4bK8bcaffffe66dffff.
>>     From: "ravi" <sip:32331001 at 81.21.33.35
>>     <mailto:sip:32331001 at 81.21.33.35>>;tag=502effff5ddeffff.
>>     To: <sip:99106883 at 81.21.33.35
>>     <mailto:sip:99106883 at 81.21.33.35>>;tag=E067A27C-3ED.
>>     Date: Mon, 28 Aug 2006 08:24:22 GMT.
>>     Call-ID: 61d40000ddc8ffff at 192.168.0.74
>>     <mailto:61d40000ddc8ffff at 192.168.0.74>.
>>     Server: Cisco-SIPGateway/IOS-12.x.
>>     CSeq: 47822 INVITE.
>>     Allow-Events: telephone-event.
>>     Content-Length: 0.
>>
>>
>>     because this 501 is internal server or gateway error so iam
>>     working on that now.
>>        
>>                          Thankyou.
>>
>>     Regards,
>>     Ravi.
>>
>>
>>
>>
>>     On 8/28/06, *Greger V. Teigre* <greger at teigre.com
>>     <mailto:greger at teigre.com>> wrote:
>>
>>         My gut feeling would tell me that the codec has nothing to do
>>         with it unless you find error messages in your ser log or
>>         var/log/messages. Sure nothing else has changed?
>>         g-)
>>
>>         ravi reddy wrote:
>>         Hi SER users,
>>
>>                       Iam using SER-0.9.6 with mediaproxy-0.5 and
>>         every think works fine except with the codecs . G711 a & g711
>>         ulaw works fine but when i tuned grandstream settings to use
>>         g729 codec for pstn calls the call is not done by the
>>         Mediaproxy server .
>>          
>>                                                                but
>>         when i created a g729 fake rtp generator it looks fine in
>>         sessions.py  .  so in order to forward the g729 phone call in
>>         to the pstn world what i have to do ?.
>>
>>         Some folks told that "integrate SER with Asterisk Works!" is
>>         it really works for me?
>>
>>         my pstn provider supports all codecs . ofcourse iam working
>>         in that company itself. 
>>
>>         please suggest some thing , so that I can bye pass this problem.
>>
>>                                                                             
>>         Thank You.
>>
>>
>>         Regards,
>>         Ravi.
>>
>>         ------------------------------------------------------------------------
>>
>>         _______________________________________________
>>         Serusers mailing list
>>
>>
>>
>>         Serusers at lists.iptel.org <mailto:Serusers at lists.iptel.org>
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>>
>>
>>           
>>
>>
>
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