[asterisk-gui] FW: Paid support for GUI and Asterisk configuration?
Alek Katamail
diablo2875 at katamail.com
Thu Jul 17 03:24:32 CDT 2008
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: asterisk-gui-bounces at lists.digium.com
[mailto:asterisk-gui-bounces at lists.digium.com] 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:s at 91.121.68.82 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:0707777777 at voip.eutelia.it>;tag=as46f4cb8b
> To: <sip:0708888888 at voip.eutelia.it>
> Contact: <sip:0707777777 at 91.121.136.13:27390>
> Call-ID: 241941db71c5f15351a9f9a9644a2d74 at voip.eutelia.it
> 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:0707777777 at voip.eutelia.it>;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 -
> 241941db71c5f15351a9f9a9644a2d74 at voip.eutelia.it
> 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:0707777777 at voip.eutelia.it>;tag=as46f4cb8b
> To: <sip:0708888888 at voip.eutelia.it>;tag=as7560d724
> Call-ID: 241941db71c5f15351a9f9a9644a2d74 at voip.eutelia.it
> 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
> '241941db71c5f15351a9f9a9644a2d74 at voip.eutelia.it' in 32000 ms (Method:
> INVITE)
> s301086*CLI>
> <--- SIP read from 83.211.227.21:5060 --->
> ACK sip:s at 91.121.68.82 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:0707777777 at voip.eutelia.it>;tag=as46f4cb8b
> Call-ID: 241941db71c5f15351a9f9a9644a2d74 at voip.eutelia.it
> To: <sip:0708888888 at voip.eutelia.it>;tag=as7560d724
> CSeq: 103 ACK
> Content-Length: 0
>
>
> <------------->
> --- (10 headers 0 lines) ---
> s301086*CLI>
> <--- SIP read from 83.211.227.21:5060 --->
> INVITE sip:s at 91.121.68.82 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:0707777777 at voip.eutelia.it>;tag=as46f4cb8b
> To: <sip:0708888888 at voip.eutelia.it>
> Contact: <sip:0707777777 at 91.121.136.13:27390>
> Call-ID: 241941db71c5f15351a9f9a9644a2d74 at voip.eutelia.it
> 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:0707777777 at voip.eutelia.it>;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 -
> 241941db71c5f15351a9f9a9644a2d74 at voip.eutelia.it
> 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:0707777777 at voip.eutelia.it>;tag=as46f4cb8b
> To: <sip:0708888888 at voip.eutelia.it>;tag=as7560d724
> Call-ID: 241941db71c5f15351a9f9a9644a2d74 at voip.eutelia.it
> 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
> '241941db71c5f15351a9f9a9644a2d74 at voip.eutelia.it' in 32000 ms (Method:
> INVITE)
> s301086*CLI>
> <--- SIP read from 83.211.227.21:5060 --->
> ACK sip:s at 91.121.68.82 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:0707777777 at voip.eutelia.it>;tag=as46f4cb8b
> Call-ID: 241941db71c5f15351a9f9a9644a2d74 at voip.eutelia.it
> To: <sip:0708888888 at voip.eutelia.it>;tag=as7560d724
> CSeq: 105 ACK
> Content-Length: 0
>
>
> Now what?
>
> Alek
>
> -----Original Message-----
> From: asterisk-gui-bounces at lists.digium.com
> [mailto:asterisk-gui-bounces at lists.digium.com] 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" <diablo2875 at katamail.com>
>> To: "Asterisk GUI project discussion" <asterisk-gui at 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
>
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