[asterisk-users] Am I having problems with codecs? or am I not receiving an invite at all from my DID provider?

[bruce bruce]

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:MyPassword at sip.spikko.com:5090/MyUsername*


Inbound Router:
*Send Any DID and ANY CID to Music on Hold*


Sip debug:

*Really destroying SIP dialog '
417b3c8f3a97a82d4629343a53b2feb6 at 177.177.177.177' Method: REGISTER*
*tel*CLI>*
*<--- SIP read from UDP:82.80.252.29:5090 --->*
*INVITE sip:MyUsername at 177.177.177.177 <sip%3AMyUsername at 177.177.177.177>SIP/2.0
*
*Via: SIP/2.0/UDP 82.80.252.234:5090;branch=z9hG4bK07b38a0c;rport*
*From: "Unknown" <sip:Unknown at 82.80.252.234:5090>;tag=as24089849*
*To: <sip:MyUsername at 177.177.177.177 <sip%3AMyUsername at 177.177.177.177>>*
*Contact: <sip:Unknown at 82.80.252.234:5090>*
*Call-ID: 55a4cf1f4e5575e97f8b3b23495f0a43 at 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 -
55a4cf1f4e5575e97f8b3b23495f0a43 at 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:asterisk at 82.80.252.234:5090>;tag=as4af8cf81*
*To: <sip:saarshalom at 173.203.29.102 <sip%3Asaarshalom at 173.203.29.102>
>;tag=as64c0ba34*
*Call-ID: 497197a679122f5d448d324f571f3c3e at 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
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