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=g729gsmulawalaw*
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...@177.177.177.177SIP/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
