<br><br>---------- Forwarded message ----------<br><span class="gmail_quote">From: <b class="gmail_sendername">ravi reddy</b> <<a href="mailto:mravikreddy@gmail.com">mravikreddy@gmail.com</a>><br>Date: Aug 29, 2006 9:56 AM
<br>Subject: Re: [Serusers] Granstream AtA g729 codec --- help please<br>To: "Greger V. Teigre" <<a href="mailto:greger@teigre.com">greger@teigre.com</a>><br><br></span><div>Thanks Greger for asking me,<br>
<br>
I want a small information regarding billing of sip calls (even though it is not related SER)<br>
please make a note to me :-)<br>
<br>
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.<br>
<br>
So, what i need a suggestion from you is <br>
<br>
1) Do i need to learn perl language and write the script ?.<br>
<br>
2) or is there any other way to control the overflow of start stop messages in to SER ...?.<br>
<br>
If you know please tell me .<br>
<br>
Thank You.</div><div><span class="e" id="q_10d58b7276a470a7_1"><br>
<br>
Regards,<br>
Ravi.<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br><br><div><span class="gmail_quote">On 8/28/06, <b class="gmail_sendername">Greger V. Teigre</b> <<a href="mailto:greger@teigre.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">greger@teigre.com
</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div>
<div bgcolor="#ffffff" text="#000000">
Good to hear! <br>
g-)<br>
PS! No need to "sir"-me. Keep it informal, I'm Greger ;-)</div><div><span><br>
<br>
ravi reddy wrote:
<blockquote cite="http://midcfbb389b0608280143r14cae712p56663a00abcfe79@mail.gmail.com" type="cite">Dear Sir,<br>
<br>
Thanks for your reply. i found the problem :-)<br>
<br>
the problem is itself in my pstn gateway that they have to configure to
allow my SER sending g729 codecs ,<br>
<br>
i came to know from the folling grep message.<br>
<br>
v=0.<br>
o=32331001 8000 8001 IN IP4 <a href="http://192.168.0.74" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">192.168.0.74</a>.<br>
s=SIP Call.<br>
c=IN IP4 <a href="http://81.21.33.35" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">81.21.33.35</a>.<br>
t=0 0.<br>
m=audio 60040 RTP/AVP 18 4 99 2.<br>
a=sendrecv.<br>
a=rtpmap:18 G729/8000.<br>
a=rtpmap:4 G723/8000.<br>
a=rtpmap:99 iLBC/8000.<br>
a=fmtp:99 mode=20.<br>
a=rtpmap:2 G726-32/8000.<br>
a=ptime:20.<br>
<br>
<br>
#<br>
U <a href="http://81.21.33.35:5060" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">81.21.33.35:5060</a> -> <a href="http://81.21.34.34:5068" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
81.21.34.34:5068</a><br>
SIP/2.0 501 Not Implemented.<br>
Via: SIP/2.0/UDP <a href="http://192.168.0.74:5068" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">192.168.0.74:5068</a>;rport=5068;received=<a href="http://81.21.34.34" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
81.21.34.34</a>;branch=z9hG4bK8bcaffffe66dffff.<br>
From: "ravi" <<a href="mailto:sip:32331001@81.21.33.35" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">sip:32331001@81.21.33.35</a>>;tag=502effff5ddeffff.<br>
To: <<a href="mailto:sip:99106883@81.21.33.35" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">sip:99106883@81.21.33.35</a>>;tag=E067A27C-3ED.<br>
Date: Mon, 28 Aug 2006 08:24:22 GMT.<br>
Call-ID: <a href="mailto:61d40000ddc8ffff@192.168.0.74" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">61d40000ddc8ffff@192.168.0.74</a>.<br>
Server: Cisco-SIPGateway/IOS-12.x.<br>
CSeq: 47822 INVITE.<br>
Allow-Events: telephone-event.<br>
Content-Length: 0.<br>
<br>
<br>
because this 501 is internal server or gateway error so iam working on
that now.<br>
<br>
Thankyou.<br>
<br>
Regards,<br>
Ravi.<br>
<br>
<br>
<br>
<br>
<div><span class="gmail_quote">On 8/28/06, <b class="gmail_sendername">Greger V. Teigre</b> <<a href="mailto:greger@teigre.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">greger@teigre.com
</a>> wrote:</span>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div>
<div bgcolor="#ffffff" text="#000000">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? <br>
g-)<br>
<br>
ravi reddy wrote:
</div>
<div><span>Hi SER users,<br>
<br>
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 .<br>
<br>
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 ?.<br>
<br>
Some folks told that "integrate SER with Asterisk Works!" is it really
works for me?<br>
<br>
my pstn provider supports all codecs . ofcourse iam working in that
company itself. <br>
<br>
please suggest some thing , so that I can bye pass this problem.<br>
<br>
Thank You.<br>
<br>
<br>
Regards,<br>
Ravi.<br>
<br>
</span></div>
<div>
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