Ok then, Let's start again from the beginning. GUI update to latest version 475
Now in sip providers I got 2 trunk from the same provider. I call from the number 0707777777 to the number 0708888888. They are just examples numbers. When I call to one of the two trunks I got this debug: <--- SIP read from 83.211.227.21:5060 ---> INVITE sip:[EMAIL PROTECTED] SIP/2.0 Record-Route: <sip:83.211.227.21;ftag=as46f4cb8b;lr=on> Record-Route: <sip:83.211.227.14;ftag=as46f4cb8b;lr=on> Record-Route: <sip:83.211.227.21;ftag=as46f4cb8b;lr=on> Via: SIP/2.0/UDP 83.211.227.21;branch=0 Via: SIP/2.0/UDP 83.211.227.14;branch=z9hG4bK34fa.a64a2f01.0 Via: SIP/2.0/UDP 83.211.227.21;branch=0 Via: SIP/2.0/UDP 91.121.136.13:27390;branch=z9hG4bK6e81d628;rport=27390 From: "Alek Corona" <sip:[EMAIL PROTECTED]>;tag=as46f4cb8b To: <sip:[EMAIL PROTECTED]> Contact: <sip:[EMAIL PROTECTED]:27390> Call-ID: [EMAIL PROTECTED] CSeq: 103 INVITE Max-Forwards: 14 Date: Sat, 12 Jul 2008 11:03:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY P-src-ip: 62.10.180.233 Content-Type: application/sdp Content-Length: 294 Remote-Party-ID: <sip:[EMAIL PROTECTED]>;party=calling;id-type=subscriber;screen=yes ;privacy=off v=0 o=root 4211 4212 IN IP4 91.121.136.13 s=session c=IN IP4 83.211.223.196 t=0 0 m=audio 63752 RTP/AVP 0 8 111 97 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> --- (20 headers 13 lines) --- Sending to 83.211.227.21 : 5060 (NAT) Using INVITE request as basis request - [EMAIL PROTECTED] Found peer 'trunk_2' <--- Reliably Transmitting (NAT) to 83.211.227.21:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 83.211.227.21;branch=0;received=83.211.227.21 Via: SIP/2.0/UDP 83.211.227.14;branch=z9hG4bK34fa.a64a2f01.0 Via: SIP/2.0/UDP 83.211.227.21;branch=0 Via: SIP/2.0/UDP 91.121.136.13:27390;branch=z9hG4bK6e81d628;rport=27390 From: "Alek Corona" <sip:[EMAIL PROTECTED]>;tag=as46f4cb8b To: <sip:[EMAIL PROTECTED]>;tag=as7560d724 Call-ID: [EMAIL PROTECTED] CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="55d372a2" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in 32000 ms (Method: INVITE) s301086*CLI> <--- SIP read from 83.211.227.21:5060 ---> ACK sip:[EMAIL PROTECTED] SIP/2.0 Max-Forwards: 15 Record-Route: <sip:83.211.227.21;ftag=as46f4cb8b;lr=on> Via: SIP/2.0/UDP 83.211.227.21;branch=0 Via: SIP/2.0/UDP 83.211.227.14;branch=z9hG4bK34fa.a64a2f01.0 From: "Alek Corona" <sip:[EMAIL PROTECTED]>;tag=as46f4cb8b Call-ID: [EMAIL PROTECTED] To: <sip:[EMAIL PROTECTED]>;tag=as7560d724 CSeq: 103 ACK Content-Length: 0 <-------------> --- (10 headers 0 lines) --- s301086*CLI> <--- SIP read from 83.211.227.21:5060 ---> INVITE sip:[EMAIL PROTECTED] SIP/2.0 Record-Route: <sip:83.211.227.21;ftag=as46f4cb8b;lr=on> Record-Route: <sip:83.211.227.14;ftag=as46f4cb8b;lr=on> Record-Route: <sip:83.211.227.21;ftag=as46f4cb8b;lr=on> Via: SIP/2.0/UDP 83.211.227.21;branch=0 Via: SIP/2.0/UDP 83.211.227.14;branch=z9hG4bK14fa.0796de33.0 Via: SIP/2.0/UDP 83.211.227.21;branch=0 Via: SIP/2.0/UDP 91.121.136.13:27390;branch=z9hG4bK1b2ec1dc;rport=27390 From: "Alek Corona" <sip:[EMAIL PROTECTED]>;tag=as46f4cb8b To: <sip:[EMAIL PROTECTED]> Contact: <sip:[EMAIL PROTECTED]:27390> Call-ID: [EMAIL PROTECTED] CSeq: 105 INVITE Max-Forwards: 14 Date: Sat, 12 Jul 2008 11:03:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY P-src-ip: 62.10.180.233 Content-Type: application/sdp Content-Length: 294 Remote-Party-ID: <sip:[EMAIL PROTECTED]>;party=calling;id-type=subscriber;screen=yes ;privacy=off v=0 o=root 4211 4214 IN IP4 91.121.136.13 s=session c=IN IP4 83.211.223.197 t=0 0 m=audio 63528 RTP/AVP 0 8 111 97 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 <-------------> --- (20 headers 13 lines) --- Sending to 83.211.227.21 : 5060 (NAT) Using INVITE request as basis request - [EMAIL PROTECTED] Found peer 'trunk_2' s301086*CLI> <--- Reliably Transmitting (NAT) to 83.211.227.21:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 83.211.227.21;branch=0;received=83.211.227.21 Via: SIP/2.0/UDP 83.211.227.14;branch=z9hG4bK14fa.0796de33.0 Via: SIP/2.0/UDP 83.211.227.21;branch=0 Via: SIP/2.0/UDP 91.121.136.13:27390;branch=z9hG4bK1b2ec1dc;rport=27390 From: "Alek Corona" <sip:[EMAIL PROTECTED]>;tag=as46f4cb8b To: <sip:[EMAIL PROTECTED]>;tag=as7560d724 Call-ID: [EMAIL PROTECTED] CSeq: 105 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7ef90aa5" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in 32000 ms (Method: INVITE) s301086*CLI> <--- SIP read from 83.211.227.21:5060 ---> ACK sip:[EMAIL PROTECTED] SIP/2.0 Max-Forwards: 15 Record-Route: <sip:83.211.227.21;ftag=as46f4cb8b;lr=on> Via: SIP/2.0/UDP 83.211.227.21;branch=0 Via: SIP/2.0/UDP 83.211.227.14;branch=z9hG4bK14fa.0796de33.0 From: "Alek Corona" <sip:[EMAIL PROTECTED]>;tag=as46f4cb8b Call-ID: [EMAIL PROTECTED] To: <sip:[EMAIL PROTECTED]>;tag=as7560d724 CSeq: 105 ACK Content-Length: 0 Now what? Alek -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brandon Kruse Sent: venerdì 11 luglio 2008 23.13 To: Asterisk GUI project discussion Cc: Asterisk GUI project discussion Subject: Re: [asterisk-gui] FW: Paid support for GUI and Asterisk configuration? >----- Original Message ----- >From: "Alek Katamail" <[EMAIL PROTECTED]> >To: "Asterisk GUI project discussion" <[email protected]> >Sent: Friday, July 11, 2008 2:38:32 AM GMT -06:00 US/Canada Central >Subject: Re: [asterisk-gui] FW: Paid support for GUI and Asterisk configuration? > >Dear Pari, >I've spent a month understanding how to solve the income problem for different trunks from the same provider and now you ask me to >roll back? :-) >Sorry that way doesn't work for incoming calls. > >So BKruse no help? >So I can't buy support for my configuration? > >[snip] Why don't you upgrade the GUI, and also, USE THE GUI. What you are doing is currently breaking, and that is your fault because you manually edited the config files. If you did it the right way, through the GUI, you could have pointed the second context to the other "Dialplan" (DID_trunk_1) Why don't you paste some debug information? You can buy a configuration package from digium: http://www.digium.com/en/services/consulting.php -bk _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-gui mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-gui _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-gui mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-gui
