FreePBX questions should be asked at FreePBX forums. As for the asterisk part, where are you defining the context to receive incoming calls? Probably in the trunk settings (Peer Details) you need to add "context=from-trunk" if FreePBX still uses it as the default context for incoming calls.
Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-06-10 11:24 AM, "bruce bruce" <bruceb...@gmail.com> wrote: Hi Guys, I have Spikko setup as provider of DID and outbound routes and I can make calls out but no inbound calls via DID can be made. I did a sip debug which is reported below. I never receive the call though, I have a catch all in my inbound routes and it doesn't hit my context at all or not sip invite comes in: FreePBX: Trunk Name: *Spikko* Peer Detail *username=MyUsername* *type=friend* *secret=MyPassword* *host=sip.spikko.com* *nat=no* *port=5090* *fromuser=MyUsername* *disallow=all* *allow=g729&gsm&ulaw&alaw* Register String: *MyUsername:mypassw...@sip.spikko.com:5090/MyUsername* Inbound Router: *Send Any DID and ANY CID to Music on Hold* Sip debug: *Really destroying SIP dialog ' 417b3c8f3a97a82d4629343a53b2f...@177.177.177.177' Method: REGISTER* *tel*CLI>* *<--- SIP read from UDP:82.80.252.29:5090 --->* *INVITE sip:myusern...@177.177.177.177 <sip%3amyusern...@177.177.177.177>SIP/2.0 * *Via: SIP/2.0/UDP 82.80.252.234:5090;branch=z9hG4bK07b38a0c;rport* *From: "Unknown" <sip:unkn...@82.80.252.234:5090>;tag=as24089849* *To: <sip:myusern...@177.177.177.177 <sip%3amyusern...@177.177.177.177>>* *Contact: <sip:unkn...@82.80.252.234:5090>* *Call-ID: 55a4cf1f4e5575e97f8b3b23495f0...@82.80.252.234* *CSeq: 102 INVITE* *User-Agent: AG1* *Max-Forwards: 70* *Date: Thu, 10 Jun 2010 14:58:09 GMT* *Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY* *Supported: replaces* *Content-Type: application/sdp* *Content-Length: 331* * * *v=0* *o=root 6129 6129 IN IP4 82.80.252.234* *s=session* *c=IN IP4 82.80.252.234* *t=0 0* *m=audio 10172 RTP/AVP 18 3 97 101* *a=rtpmap:18 G729/8000* *a=fmtp:18 annexb=no* *a=rtpmap:3 GSM/8000* *a=rtpmap:97 iLBC/8000* *a=fmtp:97 mode=30* *a=rtpmap:101 telephone-event/8000* *a=fmtp:101 0-16* *a=silenceSupp:off - - - -* *a=ptime:20* *a=sendrecv* * * *<------------->* *--- (14 headers 16 lines) ---* *Using INVITE request as basis request - 55a4cf1f4e5575e97f8b3b23495f0...@82.80.252.234* *Found peer 'Spikko' for 'Unknown' from 82.80.252.29:5090* I also sometimes get this even though trunk shows registered and can make calls out: *<--- Transmitting (no NAT) to 82.80.252.29:5090 --->* *SIP/2.0 489 Bad event* *Via: SIP/2.0/UDP 82.80.252.234:5090 ;branch=z9hG4bK463b703d;received=82.80.252.29;rport=5090* *From: "asterisk" <sip:aster...@82.80.252.234:5090>;tag=as4af8cf81* *To: <sip:saarsha...@173.203.29.102 <sip%3asaarsha...@173.203.29.102> >;tag=as64c0ba34* *Call-ID: 497197a679122f5d448d324f571f3...@82.80.252.234* *CSeq: 102 NOTIFY* *Server: Asterisk PBX 1.6.2.7* *Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO* *Supported: replaces, timer* *Content-Length: 0* Thanks, Bruce -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
