I am using 2.6 RC2/asterisk1.4.21/zap, and am trying to migrate from an old Asterisk 1.2 system (non freepbx), which worked completely on the same network with the same phones and same trunks. My build template is also used on several other systems without a fault.
I built my trunks and everything in Freepbx, but it appears that asterisk is not reading the includes properly because my sip registration was not showing up under ‘sip show registry’ - its empty.
Besides that I added some NAT settings to sip_general_custom.conf under [general], and when running sip debug on my trunk peer it showed 192.168 addresses, I added these settings directly to sip.conf and then it showed the correct NAT IPs in sip debug.
I then added the sip registration directly to sip.conf and ‘sip show registry’ showed trunks registered, however at this point my extensions would only ring once (at the extension), then go to voicemail.
At this point, I went in and hit submit on some extensions, no change, so I started from fresh asterisk/zap/freepbx source and compiled it again, now nothing works, no registrations show up, no phones work.
Original Ast 1.2 SIP.CONF:
register => 1166:[email protected]:5060
Trunks as entered in FreePBX:
[code]Trunk Name: 2800_in
registration: 1166:[email protected]:5060[/code]
Reliably Transmitting (NAT) to 22.214.171.124:5060: OPTIONS sip:126.96.36.199 SIP/2.0 Via: SIP/2.0/UDP 188.8.131.52:5060;branch=z9hG4bK33595b2b;rport From: "asterisk" <sip:[email protected]>;tag=as298629cd To: <sip:184.108.40.206> Contact: <sip:[email protected]> Call-ID: [email protected] CSeq: 102 OPTIONS User-Agent: PBXware Max-Forwards: 70 Date: Thu, 15 Oct 2009 19:42:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <--- SIP read from 220.127.116.11:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 18.104.22.168:5060;branch=z9hG4bK33595b2b;rport;received=22.214.171.124 From: "asterisk" <sip:[email protected]>;tag=as298629cd To: <sip:126.96.36.199>;tag=as5c1e1069 Call-ID: [email protected] CSeq: 102 OPTIONS User-Agent: Telcoware Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Accept: application/sdp Content-Length: 0 [2009-10-14 23:28:06] DEBUG chan_sip.c: Adding subscription for extension 2801 context from-internal for peer 2801 [2009-10-14 23:28:06] VERBOSE logger.c: -- Incoming call: Got SIP response 400 "Bad Request" back from 192.168.102.250