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 '1571@default' 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

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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 '1571@default' 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?

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On Apr 13, 11:51 am, Desk Rabbit <nos...@example.com> wrote:
> Matt wrote: 
> 
> 
> 
> 
> 
> 
> 
>
> 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

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On Apr 13, 12:14 pm, "Matt" <matthew.humphr...@gmail.com> wrote:
> On Apr 13, 11:51 am, Desk Rabbit <nos...@example.com> wrote:
>
>
> 
> 
> 
> 
> 
> 
> 
> 
> 
>
> 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

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Matt wrote:
> On Apr 13, 11:51 am, Desk Rabbit <nos...@example.com> wrote: 
>
> 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".

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On Apr 13, 12:41 pm, Desk Rabbit <nos...@example.com> wrote:
> Matt wrote: 
> 
> 
>
> Thats fine and thats what I have here at both ends.
> 
>
> Ah! At this point you are on your own. Sorry.
> 
>
> 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!

[ Post a follow-up to this message ]

Matt wrote:
 
[vbcol=seagreen]
> 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 [vani
lla
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/10000@sipgate|60|tr") in new stack

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

Executing Dial("SIP/6012-00637c30", "SIP/08000850643@voipcouk|60|o") in new
stack

So your dial requests should look like 1571@voipfone, 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) (UnSoEsNpEaTm@ale.cx)
00:22:52 up  4:41,  2 users,  load average: 0.03, 0.06, 0.07
Yes. I'm just guessing.

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On Apr 14, 12:38 am, alexd <troffa...@hotmail.com> wrote:
> Matt wrote: 
>
> 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!
 
>
> Also, are the local users who can dial 1571 dialling 9 first, perchance? H
ow
> are you routing the call? @default doesn't look right. For example [va
nilla
> 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/10000@sipgate|60|tr") in new stac
k
>
> Dial 9 for voip.co.uk trunk, 9 gets stripped off:
>
> Executing Dial("SIP/6012-00637c30", "SIP/08000850643@voipcouk|60|o") in ne
w
> stack
>
> So your dial requests should look like 1571@voipfone, 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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