Alek, Like I have said many, many, many times before, you can buy a configuration package here:
http://www.digium.com/en/services/consulting.php -bk Alek Katamail wrote: > I've done it already. But still I got same issues. > > Alek > > -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of James > Middendorff > Sent: giovedì 17 luglio 2008 13.59 > To: Asterisk GUI project discussion > Subject: Re: [asterisk-gui] FW: Paid support for GUI and Asterisk > configuration? > > Now your revision is correct. You must run .configure make and make install > again and your gui will be updated > James aka riddlebox > > -----Original Message----- > From: "Alek Katamail" <[EMAIL PROTECTED]> > To: "'Asterisk GUI project discussion'" <asterisk-gui@lists.digium.com> > Sent: 7/17/2008 3:24 AM > Subject: Re: [asterisk-gui] FW: Paid support for GUI and Asterisk > configuration? > > 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 > > > _______________________________________________ > --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
