Hello. Please help me with this, I can find any solution on this pls help. Your help will be very appreciated. Thanks.
--- On Tue, 11/17/09, Landy Landy <[email protected]> wrote: > From: Landy Landy <[email protected]> > Subject: Re: [asterisk-users] can't call through voip provider > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <[email protected]> > Date: Tuesday, November 17, 2009, 7:33 AM > 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 > _______________________________________________ -- 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
