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uk.telecom.voip (UK VOIP) (uk.telecom.voip) Discussion of topics relevant to packet based voice technologies including Voice over IP (VoIP), Fax over IP (FoIP), Voice over Frame Relay (VoFR), Voice over Broadband (VoB) and Voice on the Net (VoN) as well as service providers, hardware and software for use with these technologies. Advertising is not allowed.

Linking Asterisk Servers - partial success



 
 
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  #1  
Old April 13th 07, 10:39 AM posted to uk.telecom.voip
Matt
external usenet poster
 
Posts: 141
Default Linking Asterisk Servers - partial success

Hi,

We've linked the servers and can call both sets of extensions etc.

If I try to call out over one of the trunks on the remote server the
call fails and I see

NOTICE[23648]: chan_iax2.c:6793 socket_process: Rejected connect
attempt from xx.xx.xx.xx, request '[email protected]' does not exist.

I was trying to call out on the trunk to the trunks voicemail services
(1571 - voipfone).

Any ideas what I've got wrong, or missed out?

Cheers


Matthew

  #2  
Old April 13th 07, 11:51 AM posted to uk.telecom.voip
Desk Rabbit
external usenet poster
 
Posts: 169
Default Linking Asterisk Servers - partial success

Matt wrote:
Hi,

We've linked the servers and can call both sets of extensions etc.

If I try to call out over one of the trunks on the remote server the
call fails and I see

NOTICE[23648]: chan_iax2.c:6793 socket_process: Rejected connect
attempt from xx.xx.xx.xx, request '[email protected]' does not exist.

I was trying to call out on the trunk to the trunks voicemail services
(1571 - voipfone).

Any ideas what I've got wrong, or missed out?

Cheers


Matthew

What version of FreePBX?
What version of Asterisk?

  #3  
Old April 13th 07, 12:14 PM posted to uk.telecom.voip
Matt
external usenet poster
 
Posts: 141
Default Linking Asterisk Servers - partial success

On Apr 13, 11:51 am, Desk Rabbit wrote:
Matt wrote:
Hi,


We've linked the servers and can call both sets of extensions etc.


If I try to call out over one of the trunks on the remote server the
call fails and I see


NOTICE[23648]: chan_iax2.c:6793 socket_process: Rejected connect
attempt from xx.xx.xx.xx, request '[email protected]' does not exist.


I was trying to call out on the trunk to the trunks voicemail services
(1571 - voipfone).


Any ideas what I've got wrong, or missed out?


Cheers


Matthew


What version of FreePBX?
What version of Asterisk?


Cheers for the reply Desk Rabbit.

The local end is freepbx 2.2.0 with asterisk 1.2.7.1

And the remote end is the Digium Asterisk GUI (beta) on Asterisk
1.4.2.

The problem occurs when I try to dial out over the remote voipfone
trunk. The trunk all works ok for the users at their end.

Thanks again

Matthew

  #4  
Old April 13th 07, 12:33 PM posted to uk.telecom.voip
Matt
external usenet poster
 
Posts: 141
Default Linking Asterisk Servers - partial success

On Apr 13, 12:14 pm, "Matt" wrote:
On Apr 13, 11:51 am, Desk Rabbit wrote:



Matt wrote:
Hi,


We've linked the servers and can call both sets of extensions etc.


If I try to call out over one of the trunks on the remote server the
call fails and I see


NOTICE[23648]: chan_iax2.c:6793 socket_process: Rejected connect
attempt from xx.xx.xx.xx, request '[email protected]' does not exist.


I was trying to call out on the trunk to the trunks voicemail services
(1571 - voipfone).


Any ideas what I've got wrong, or missed out?


Cheers


Matthew


What version of FreePBX?
What version of Asterisk?


Cheers for the reply Desk Rabbit.

The local end is freepbx 2.2.0 with asterisk 1.2.7.1

And the remote end is the Digium Asterisk GUI (beta) on Asterisk
1.4.2.

The problem occurs when I try to dial out over the remote voipfone
trunk. The trunk all works ok for the users at their end.

Thanks again

Matthew


I should also add that the remote server users can successfully make
outgoing calls over the local trunks here

  #5  
Old April 13th 07, 12:41 PM posted to uk.telecom.voip
Desk Rabbit
external usenet poster
 
Posts: 169
Default Linking Asterisk Servers - partial success

Matt wrote:
On Apr 13, 11:51 am, Desk Rabbit wrote:
Matt wrote:
Hi,
We've linked the servers and can call both sets of extensions etc.
If I try to call out over one of the trunks on the remote server the
call fails and I see
NOTICE[23648]: chan_iax2.c:6793 socket_process: Rejected connect
attempt from xx.xx.xx.xx, request '[email protected]' does not exist.
I was trying to call out on the trunk to the trunks voicemail services
(1571 - voipfone).
Any ideas what I've got wrong, or missed out?
Cheers
Matthew

What version of FreePBX?
What version of Asterisk?


Cheers for the reply Desk Rabbit.

The local end is freepbx 2.2.0 with asterisk 1.2.7.1

Thats fine and thats what I have here at both ends.


And the remote end is the Digium Asterisk GUI (beta) on Asterisk
1.4.2.

Ah! At this point you are on your own. Sorry.


The problem occurs when I try to dial out over the remote voipfone
trunk. The trunk all works ok for the users at their end.


The only thing I can think of is the conext. FreePBX uses
"from-internal". It looks like (And I am guessing here) your remote site
is using "default".
  #6  
Old April 13th 07, 12:49 PM posted to uk.telecom.voip
Matt
external usenet poster
 
Posts: 141
Default Linking Asterisk Servers - partial success

On Apr 13, 12:41 pm, Desk Rabbit wrote:
Matt wrote:
On Apr 13, 11:51 am, Desk Rabbit wrote:
Matt wrote:
Hi,
We've linked the servers and can call both sets of extensions etc.
If I try to call out over one of the trunks on the remote server the
call fails and I see
NOTICE[23648]: chan_iax2.c:6793 socket_process: Rejected connect
attempt from xx.xx.xx.xx, request '[email protected]' does not exist.
I was trying to call out on the trunk to the trunks voicemail services
(1571 - voipfone).
Any ideas what I've got wrong, or missed out?
Cheers
Matthew
What version of FreePBX?
What version of Asterisk?


Cheers for the reply Desk Rabbit.


The local end is freepbx 2.2.0 with asterisk 1.2.7.1


Thats fine and thats what I have here at both ends.

And the remote end is the Digium Asterisk GUI (beta) on Asterisk
1.4.2.


Ah! At this point you are on your own. Sorry.

The problem occurs when I try to dial out over the remote voipfone
trunk. The trunk all works ok for the users at their end.


The only thing I can think of is the conext. FreePBX uses
"from-internal". It looks like (And I am guessing here) your remote site
is using "default".


I'm trying to persuade them to move away from the asterisk gui!

I'll check the contexts - looks like a good point to start.

Thanks!

  #7  
Old April 14th 07, 12:38 AM posted to uk.telecom.voip
alexd
external usenet poster
 
Posts: 1,765
Default Linking Asterisk Servers - partial success

Matt wrote:

NOTICE[23648]: chan_iax2.c:6793 socket_process: Rejected connect
attempt from xx.xx.xx.xx, request '[email protected]' does not exist.
I was trying to call out on the trunk to the trunks voicemail
services (1571 - voipfone).


I'm trying to persuade them to move away from the asterisk gui!


I've not used it myself - is it any good?

I'm bound to think that any GUI that tries to capture the functionality of
Asterisk's text config files - which is practically a programming language
in itself - is going to be a disappointment on some level.

I'll check the contexts - looks like a good point to start.


Also, are the local users who can dial 1571 dialling 9 first, perchance? How
are you routing the call? @default doesn't look right. For example [vanilla
Asterisk], in my sip.conf, my Sipgate context is called 'sipgate' and my
voip.co.uk context is called 'voipcouk'. And to use the Sipgate trunk, I
dial 89 then the number. So if I 8910000, the leading 89 gets stripped off,
and Asterisk says:

Executing Dial("SIP/6012-00637c30", "SIP/[email protected]|60|tr") in new stack

Dial 9 for voip.co.uk trunk, 9 gets stripped off:

Executing Dial("SIP/6012-00637c30", "SIP/[email protected]|60|o") in new
stack

So your dial requests should look like [email protected], or whatever your
Voipfone trunk context is. You should be able to see what it's doing by
opening an Asterisk and making sure the verbosity is at least 3 ['set
verbose 3'].

--
http://ale.cx/ (AIM:troffasky) )
00:22:52 up 4:41, 2 users, load average: 0.03, 0.06, 0.07
Yes. I'm just guessing.

  #8  
Old April 14th 07, 08:23 AM posted to uk.telecom.voip
Matt
external usenet poster
 
Posts: 141
Default Linking Asterisk Servers - partial success

On Apr 14, 12:38 am, alexd wrote:
Matt wrote:
NOTICE[23648]: chan_iax2.c:6793 socket_process: Rejected connect
attempt from xx.xx.xx.xx, request '[email protected]' does not exist.
I was trying to call out on the trunk to the trunks voicemail
services (1571 - voipfone).

I'm trying to persuade them to move away from the asterisk gui!


I've not used it myself - is it any good?


Seems a bit buggy at the moment, and without some of the freepbx
functionality.



I'm bound to think that any GUI that tries to capture the functionality of
Asterisk's text config files - which is practically a programming language
in itself - is going to be a disappointment on some level.

Think I'll stick with Freepbx at this end!

I'll check the contexts - looks like a good point to start.


Also, are the local users who can dial 1571 dialling 9 first, perchance? How
are you routing the call? @default doesn't look right. For example [vanilla
Asterisk], in my sip.conf, my Sipgate context is called 'sipgate' and my
voip.co.uk context is called 'voipcouk'. And to use the Sipgate trunk, I
dial 89 then the number. So if I 8910000, the leading 89 gets stripped off,
and Asterisk says:

Executing Dial("SIP/6012-00637c30", "SIP/[email protected]|60|tr") in new stack

Dial 9 for voip.co.uk trunk, 9 gets stripped off:

Executing Dial("SIP/6012-00637c30", "SIP/[email protected]|60|o") in new
stack

So your dial requests should look like [email protected], or whatever your
Voipfone trunk context is. You should be able to see what it's doing by
opening an Asterisk and making sure the verbosity is at least 3 ['set
verbose 3'].

Thats what I thought. All the local users can dial out without any
problems, without a 9.

The settings for the trunks are so simplistic in the gui though.....

When I get a chance I'll sit and have another watch of the cli...

Cheers

Matthew

 




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