[asterisk-gui] FW: Paid support for GUI and Asterisk configuration?

bkruse bkruse at digium.com
Fri Jul 18 14:45:38 CDT 2008 

I see the problem you are having, and I have said that
it requires more configuration OUTSIDE the gui, since
the GUI cannot handle this.

Digium will do this external configuration AND falls under
the category of "paid support".

Glad we are on the same page :)

-brandon

Alek Katamail wrote:
> 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
>>>
>>> _______________________________________________
>>> --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--
>>
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>>
>>
>> _______________________________________________
>> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>>
>> asterisk-gui mailing list
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>>    http://lists.digium.com/mailman/listinfo/asterisk-gui
>>
>>
>> _______________________________________________
>> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>>
>> asterisk-gui mailing list
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>>    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
>
>
> _______________________________________________
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>
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