Displaying 7 results from an estimated 7 matches for "uridial".
2007 Feb 25
2
Dialling ZAP channel from analogue
...)
[in]
exten => uxbod,1,Dial(sip/uxbod,20)
exten => uxbod,2,VoiceMail(5001@incoming)
exten => uxbod,3,Hangup()
[outbound-local]
exten => _9NXXXXXX,1,Dial(${TRUNK}/${EXTEN:1})
exten => _9NXXXXXX,2,Congestion()
exten => _9NXXXXXX,102,Congestion()
[uri]
exten => _[a-z].,1,Macro(uridial,${EXTEN}@${SIPDOMAIN})
exten => _[A-Z].,1,Macro(uridial,${EXTEN}@${SIPDOMAIN})
exten => _X.,1,Macro(uridial,${EXTEN}@${SIPDOMAIN})
[macro-uridial]
exten => s,1,NoOp(Calling remote SIP peer ${ARG1})
exten => s,n,Dial(SIP/${ARG1},120,tr)
exten => s,n,Congestion()
When I try dialling...
2008 Feb 11
0
asterisk-users Digest, Vol 43, Issue 30
...t;>
>>> I use outgoing URI-dialing for my sip-phones as suggested in
>>> http://www.voip-info.org/wiki/view/Asterisk+tips+SIP+URI+Dial
>>>
>>> The relevant extensions look like this:
>>>
>>> [dial-uri]
>>> exten => _[a-z].,1,Macro(uridial,${EXTEN}@${SIPDOMAIN})
>>> exten => _[A-Z].,1,Macro(uridial,${EXTEN}@${SIPDOMAIN})
>>> exten => _X.,1,Macro(uridial,${EXTEN}@${SIPDOMAIN})
>>>
>>> [macro-uridial]
>>> exten => s,1,Set(dialuri=${CUT(ARG1,\;,1)})
>>> exten => s,n,Set(C...
2007 Feb 19
2
sip to sip ?
hi all
i've just setup an * box and want to test voip calling, initially from
sip user to sip user...
local sip users can call each other, no issues.
problem arises when i try and call a remote sip account, my * box
always returns "SIP/2.0 404 Not Found"
any ideas ?
2009 Jul 15
2
how to enable dial to alex@asterisk.blurb.com
Hi
The subject line says it all how do I enable this style of call.
Pointers to the dns setup and asterisk setup would be great
or even search words for google, as I am not sure how to search for this
type of request.
Alex
--
There is no instance of a country having benefited
from prolonged warfare
-- Sun Tzu - The Art of War
-------------- next part --------------
A
2006 May 02
8
Zapata Telephony interface and torisa module error
Looking at my log file I found the following error:
May 2 12:00:45 debian kernel: Zapata Telephony Interface Registered on major 196
May 2 12:00:45 debian kernel: No ISA tormenta card found at d0000
May 2 12:00:45 debian kernel: Zapata Telephony Interface Unloaded
May 2 12:00:45 debian insmod: /lib/modules/2.4.20-8smp/misc/torisa.o: init_module: Input/output error
May 2 12:00:45 debian
2010 Jul 05
1
SIP response 482 "Loop Detected"
Hi,
We have tried upgrading from 1.6.1.14 to 1.6.2.9 and have found that we are unable to URI dial our clients. We run a multi-tenant server and have set sip.conf to forward calls to a public context based on incoming domain name. This was all working before but not it is complaining of a loop back as the source and target server are the same.
Any ideas on how to overcome this problem as we dial
2009 Aug 27
2
Selective canreinvite in multi-tenant environment
Hello, all. In our multi-tenant environment, we would like to be able
to use the reinvite media redirection within Asterisk for calls within a
tenant but not between tenants. We would like inter-tenant calls to be
fully proxied by the Asterisk server. I think the answer is, "we
can't," but I thought I'd ask anyway.
I'd dearly like to remove the substantial traffic