Issue with SIP Trunk Provider

Hi All ,

One of my Asterisk (FreePBX 2.8.1) was connected to my ISP’s SIP Trunk. While making calls to the service number i am getting the error. I am pasting the following from CLI while debug is ON. Anybody over hear come across the same like problem.

[Sep 26 08:20:51] VERBOSE[14871] chan_sip.c: — (8 headers 0 lines) —
[Sep 26 08:20:51] VERBOSE[14871] chan_sip.c: Looking for s in default (domain sip-server-ip:5060)
[Sep 26 08:20:51] VERBOSE[14871] chan_sip.c:
<— Transmitting (no NAT) to sip-trunk-ip:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP sip-trunk-ip:5060;branch=z9hG4bKasaeb7kt7dps4oeaf7bkttfb4;received=sip-trunk-ip
From: sip:sip-server-ip:5060;tag=sbc0807be2ppd2p
To: sip:sip-server-ip;tag=as6a662eea
Call-ID: isbcbdpoaf7dpoft2oehedbba[email protected]
CSeq: 1 OPTIONS
Server: Asterisk PBX 1.8.5.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:10.66.200.166:5060
Accept: application/sdp
Content-Length: 0

MyPeerConf

host=ip-server-ip
dtmfmode=rfc2833
type=peer
dissallow=all
allow=alaw
qualify=yes
nat=no
context=custom-get-did-from-sip

Thanks Advance

What is the context “custom-get-did-from-sip”? Did you create it? It seems you made a mistake in your code and don’t have an S or starting point in the extension you wrote.