We need incoming calls to simultaneously ring SIP phones, and a cell phone
which is called via a SIP or IAX trunk. When the cell phone answers we'd
like a brief prompt played (e.g. press # to accept call) and if # is pressed
connect the incoming call to the cell phone.
ZAP trunks have some of
Hello list
I don't get to compile h323. I have the mistake:
asteriskaudio.cxx: In destructor `virtual
PAsteriskSoundChannel::~PAsteriskSoundChannel()':
asteriskaudio.cxx:167: `baseChannel' undeclared (first use this function)
asteriskaudio.cxx:167: (Each undeclared identifier is reported only
Hi all
Two S100I-IAXY configured * the CVS-HEAD and following the IAXY´s
Configuration Guide v. 1.0 by Digium.
The first one S100I-IAXY have IP 10.0.0.5. (my home)
The second S100I-IAXY have IP 200.253.232.23. (my office)
I only obtain to establish a linking enters the two S100I-IAXY when I
I yesterday brought up to date the version of * the CVS and now I have a
problem.
I cannot effect the RELOAD that * it breaks.
Somebody can help or say as to load new users without stopping * ?
Thank´s
Excuse my English
Joao Carlos Moura
___
Thank´s.
It decided my problem.
Joao Carlos Moura
- Original Message -
From: Holger Schurig [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, July 20, 2004 11:42
Subject: Re: [Asterisk-Users] New CVS version
I yesterday brought up to date the version of * the CVS and now I
Hi all
what does the command NOTRANSFER in IAX.CONF?
where do i find asterisk´s commands?
In the website, VOIP-INFO.ORG I did not find anything regarded to NOTRANSFER
commands unfortunatly.
Thank you.
jmoura
___
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Hello
Use two Asterisk servers.
I registered Server2 in Server1. When I bind for an extension in the
Server1, the hard call some as and is interrupted.
My configuration:
iax.conf / Server 1
[20001]
type=friend
accountcode=20001
host=dynamic
secret=secret
context=sip
disallow=all
allow=gsm
