Brandon, As I told you before, I'VE UPDATE IT! I've followed this guide as you told me: http://asteriskNOW.org/install-related
And my last info is: Asterisk Build: Asterisk 1.4.20 Asterisk GUI-version 475 I've done it again this morning and now it says: Asterisk Build: Asterisk 1.4.20 Asterisk GUI-version 3516 Now what? Where I can get a configuration option? bye -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of bkruse Sent: mercoledì 16 luglio 2008 20.05 To: Asterisk GUI project discussion Subject: Re: [asterisk-gui] FW: Paid support for GUI and Asterisk configuration? Like I said before, You can get a configuration option, or UPDATE THE GUI! The latest revision is in the 3,000s, not 475! -brandon Alek Katamail wrote: > 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" <asterisk-gui@lists.digium.com> >> 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 > _______________________________________________ --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
