31 Aug
2006
31 Aug
'06
9:01 a.m.
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(a)gmail.com <mailto:mravikreddy@gmail.com>>
Date: Aug 29, 2006 9:56 AM
Subject: Re: [Serusers] Granstream AtA g729 codec --- help please
To: "Greger V. Teigre" <greger(a)teigre.com
<mailto:greger@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(a)teigre.com
<mailto:greger@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@81.21.33.35
<mailto:sip:32331001@81.21.33.35>>;tag=502effff5ddeffff.
To: <sip:99106883@81.21.33.35
<mailto: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
<mailto:61d40000ddc8ffff@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(a)teigre.com
<mailto:greger@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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