View Single Post
  #7 (permalink)  
Old 04-14-2007, 12:38 AM
alexd
Guest
 
Posts: n/a
Default Re: Linking Asterisk Servers - partial success

Matt wrote:

>> >>> 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).


> 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/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.


Reply With Quote