[asterisk-gui] FW: Paid support for GUI and Asterisk configuration?
Alek Katamail
diablo2875 at katamail.com
Sat Jul 12 06:06:38 CDT 2008
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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