Hi,
I have a problem with the trunk registration on my asterisk.
I have Asterisk 11.4.0 version and I use freepbx 2.11.0
This trunk worked but I maybe changed some configuration because the sip show registry show me that he is not registered. This is my configuration on freepbx, I have the same configuration on an other server and that works, and If I try to register the sip account directly on a softphone it also works.
host=freephonie.net
username=mynumber
secret=mysecret
type=peer
qualify=yes
fromdomain=freephonie.net
dtmfmode=auto
context=from-trunk
disallow=all
allow=ulaw&alaw
canreinvite=no
Register String
mynumber:[email protected]
I only add on the sip configuration settings defaultexpiry = 1800, that is required for my provider.
this is the output of the sip debug when I try to start a call
SIP/2.0 403 not registered
Call-ID: [email protected]
CSeq: 102 INVITE
From: sip:[email protected];tag=as5a97f546
To: sip:[email protected];tag=00-32608-01763f49-7baaee754
Via: SIP/2.0/UDP myexternalip:5060;received=myexternalip;rport=5060;branch=z9hG4bK7a3a0bcc
Content-Length: 0
[2013-08-15 11:13:36] WARNING[30598][C-0000001a]: chan_sip.c:22834 handle_response_invite: Received response: “Forbidden” from 'sip:[email protected];tag=as63f7fcd2’
Scheduling destruction of SIP dialog ‘[email protected]’ in 6400 ms (Method: INVITE)
Audio is at 10032
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
and when try to register:
Really destroying SIP dialog ‘[email protected]:5060’ Method: OPTIONS
Reliably Transmitting (NAT) to 212.27.52.5:5060:
OPTIONS sip:freephonie.net SIP/2.0
Via: SIP/2.0/UDP myexternalip:5060;branch=z9hG4bK0cba0227;rport
Max-Forwards: 70
From: “Unknown” sip:Unknown@myexternalip;tag=as1c0dcbc2
To: sip:freephonie.net
Contact: sip:Unknown@myexternalip:5060
Call-ID: 6b6ba7a74450d017398b5fd50400a6c4@myexternalip:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(11.4.0)
Date: Thu, 15 Aug 2013 09:26:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<— SIP read from UDP:212.27.52.5:5060 —>
SIP/2.0 501 Not Implemented Yet
Call-ID: 6b6ba7a74450d017398b5fd50400a6c4@myexternalip:5060
CSeq: 102 OPTIONS
From: “Unknown” sip:Unknown@myexternalip;tag=as1c0dcbc2
To: sip:freephonie.net;tag=00-31667-01772c3e-5c12be2c1
Via: SIP/2.0/UDP myexternalip:5060;received=myexternalip;rport=5060;branch=z9hG4bK0cba0227
Content-Length: 0
Thank you