[asterisk-gui] FW: Paid support for GUI and Asterisk configuration?
Alek Katamail
diablo2875 at katamail.com
Fri Jul 18 03:13:19 CDT 2008
Dear Brandon,
You told me also to update the GUI so I thought that maybe in this way I
could find a solution to the problem. I thought it was something that
someone could simply tell me. But maybe my company is the only one that uses
more trunks from the same provider. Anyway I wrote to Digium let's see.
Bye
Alek
-----Original Message-----
From: asterisk-gui-bounces at lists.digium.com
[mailto:asterisk-gui-bounces at lists.digium.com] On Behalf Of bkruse
Sent: giovedì 17 luglio 2008 21.33
To: Asterisk GUI project discussion
Subject: Re: [asterisk-gui] FW: Paid support for GUI and Asterisk
configuration?
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: asterisk-gui-bounces at lists.digium.com
> [mailto:asterisk-gui-bounces at lists.digium.com] 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" <diablo2875 at katamail.com>
> To: "'Asterisk GUI project discussion'" <asterisk-gui at 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: 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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>>
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>>
>>
>> _______________________________________________
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>>
>>
>
>
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