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

29 Aug 2006

---------- Forwarded message ----------
From: ravi reddy &lt;mravikreddy(a)gmail.com&gt;
Date: Aug 29, 2006 9:56 AM
Subject: Re: [Serusers] Granstream AtA g729 codec --- help please
To: "Greger V. Teigre" &lt;greger(a)teigre.com&gt;

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 &lt;greger(a)teigre.com&gt; 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.
 s=SIP Call.
 c=IN IP4 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 -> 81.21.34.34:5068
 SIP/2.0 501 Not Implemented.
 Via: SIP/2.0/UDP 192.168.0.74:5068;rport=5068;received= 81.21.34.34
 ;branch=z9hG4bK8bcaffffe66dffff.
 From: "ravi" <sip:32331001@81.21.33.35>;tag=502effff5ddeffff.
 To: <sip:99106883@81.21.33.35>;tag=E067A27C-3ED.
 Date: Mon, 28 Aug 2006 08:24:22 GMT.
 Call-ID: 61d40000ddc8ffff(a)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 &lt;greger(a)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.

  ------------------------------

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