Thanks for replying. Here is the output of sip set debug peer voipprovider:
-- Called 1829257x...@voipprovider Retransmitting #1 (NAT) to myextip:5060: INVITE sip:18292574...@myextip SIP/2.0 Via: SIP/2.0/UDP myextip:5060;branch=z9hG4bK61c970ad Max-Forwards: 70 From: "102" <sip:usern...@myextip>;tag=as78863882 To: <sip:[email protected]> Contact: <sip:[email protected]> Call-ID: [email protected] CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.5 Date: Tue, 17 Nov 2009 12:28:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 473 v=0 o=root 1332315330 1332315330 IN IP4 190.80.152.7 s=Asterisk PBX 1.6.1.5 c=IN IP4 190.80.152.7 t=0 0 m=audio 13752 RTP/AVP 0 3 8 112 5 10 7 111 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Retransmitting #2 (NAT) to myextip:5060: INVITE sip:1829257x...@myextip SIP/2.0 Via: SIP/2.0/UDP myextip:5060;branch=z9hG4bK61c970ad Max-Forwards: 70 From: "102" <sip:usern...@myextip>;tag=as78863882 To: <sip:1829257x...@myextip> Contact: <sip:usern...@myextip> Call-ID: 2908dd00500059761cc66bd81553e...@myextip CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.5 Date: Tue, 17 Nov 2009 12:28:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 473 v=0 o=root 1332315330 1332315330 IN IP4 myextip s=Asterisk PBX 1.6.1.5 c=IN IP4 190.80.152.7 t=0 0 m=audio 13752 RTP/AVP 0 3 8 112 5 10 7 111 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Retransmitting #3 (NAT) to myextip:5060: INVITE sip:1829257x...@myextip SIP/2.0 Via: SIP/2.0/UDP myextip:5060;branch=z9hG4bK61c970ad Max-Forwards: 70 From: "102" <sip:usern...@myextip>;tag=as78863882 To: <sip:1829257x...@myextip> Contact: <sip:usern...@myextip> Call-ID: 2908dd00500059761cc66bd81553e...@myextip CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.5 Date: Tue, 17 Nov 2009 12:28:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 473 v=0 o=root 1332315330 1332315330 IN IP4 myextip s=Asterisk PBX 1.6.1.5 c=IN IP4 myextip t=0 0 m=audio 13752 RTP/AVP 0 3 8 112 5 10 7 111 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv Scheduling destruction of SIP dialog '2908dd00500059761cc66bd81553e...@myextip' in 32000 ms (Method: INVITE) //////////// By looking at this trace I dont see my provider's ip address anywhere. I guess I'm doing something wrong in my conf. --- On Mon, 11/16/09, Warren Selby <[email protected]> wrote: > From: Warren Selby <[email protected]> > Subject: Re: [asterisk-users] can't call through voip provider > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <[email protected]> > Date: Monday, November 16, 2009, 9:51 PM > On Mon, Nov 16, > 2009 at 2:40 PM, Landy Landy <[email protected]> > wrote: > <snip> > > > I don't know what else to try. When I try to call I get > this at the cli: > > > > == Using SIP RTP CoS mark 5 > > -- Executing [91xxx763x...@default:1] > Dial("SIP/102-b6a06a40", > "SIP/1xxx763x...@voipprovider") in new stack > > == Using SIP RTP CoS mark 5 > > -- Called 1xxx763x...@voipprovider > > <snip> > > We could really use a little more of the CLI output of a > failed call. Maybe increase your verbosity to at least > 10. Also, what does the SIP debug of a call to the VOIP > provider look like (from the cli, type "sip set debug > peer voipprovider")? > > > -- > Thanks, > --Warren Selby > http://www.selbytech.com > > > -----Inline Attachment Follows----- > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
