Hello Team,
i am creating a ITSP SIP trunk between CME and Service provider..
the service provider has given me below detail and requested to change the below information..
I am confused.. and how to do the changes.. I have background on doing SIP profile changes.. but not able to understand..
below are the detail and recommendation from ITSP provider of India..
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Sample OG Format:-
P-Preferred-Identity:sip:68533800 @10.50.144.33----Use This header once making call from DID
INVITE sip:9225885xxx@10.0.90.2 SIP/2.0
Via: SIP/2.0/UDP 10.0.90.xxx:5060;branch=z9hG4bK5870c68d
Max-Forwards: 70
From: <sip:67105112@10.0.9xxx>;tag=as717caedd à Here please send the DID number
To: <sip:9225885887@10.0.9xxx>
Contact: <sip:67105112@10.0.90.xxx:5060> à Here please send the DID number
Call-ID: 3df9e4ed556d3de83dca17602b129788@10.0.90.182:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.20.0)
Date: Mon, 11 Feb 2019 07:59:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
P-Preferred-Identity:<sip:67105xxx@10.0.9xxx> à Here please send the SIP Pilot number
Content-Type: application/sdp
Content-Length: 305
v=0
o=root 1127501856 1127501856 IN IP4 10.0.90.1xxx
s=Asterisk PBX 11.20.0
c=IN IP4 10.0.90.xxx
t=0 0
m=audio 10490 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
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Kindly advise , how to create SIP profile based on above information..
Best Regards
Anil Singh
