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:[email protected]:5090/MyUsername* Inbound Router: *Send Any DID and ANY CID to Music on Hold* Sip debug: *Really destroying SIP dialog ' [email protected]' Method: REGISTER* *tel*CLI>* *<--- SIP read from UDP:82.80.252.29:5090 --->* *INVITE sip:[email protected] <sip%[email protected]>SIP/2.0 * *Via: SIP/2.0/UDP 82.80.252.234:5090;branch=z9hG4bK07b38a0c;rport* *From: "Unknown" <sip:[email protected]:5090>;tag=as24089849* *To: <sip:[email protected] <sip%[email protected]>>* *Contact: <sip:[email protected]:5090>* *Call-ID: [email protected]* *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 - [email protected]* *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:[email protected]:5090>;tag=as4af8cf81* *To: <sip:[email protected] <sip%[email protected]> >;tag=as64c0ba34* *Call-ID: [email protected]* *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
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