I am trying to setup Asterisk as the voicemail server for Cisco Call Manager. I
have just about everything working except for the message waiting indicator.
I have the following setup in context [ccm] in my extensions.conf file:
;MWI
exten => _2807XXX,1,SetCallerID(${EXTEN:3})
exten => _2807XXX,
read http://www.voip-info.org/tiki-index.php?page=Asterisk%
20Cisco%20CallManager%20Integration ?
I've followed these steps and I can make calls from a CCM client to
Asterisk, but the end point at the Asterisk side can't hear any audio.
On Mon, 2005-09-26 at 12:28 -0400, Brian J. Rathman
ndex.php?page=Asterisk%
> 20Cisco%20CallManager%20Integration ?
>
> I've followed these steps and I can make calls from a CCM client to
> Asterisk, but the end point at the Asterisk side can't hear any audio.
>
> On Mon, 2005-09-26 at 12:28 -0400, Brian J. Rathman wrote:
>
: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk to CCM
Hi Brian,
Calling search space is correct?
Dinesh.
On Mon, 26 Sep 2005, Brian J. Rathman wrote:
> I am currently trying to send calls from Asterisk to Cisco Call Manager 4.0
> and vice
I am currently trying to send calls from Asterisk to Cisco Call Manager 4.0 and
vice versa. I have a SIP trunk setup in CCM and I also have an entry in my
sip.conf file for CCM. Unfortunately, when I try and send a call to CCM I keep
getting:
SIP/10.0.0.1-9c18 is circuit-busy
== Everyone is
I just recently loaded a new Asterisk server running the CVS code from 12/3/04. I have been running a previous CVS version without any problems from 6/5/04. On the new server I have noticed an issue where when I place a call from one of my snom 190 phones, I see immediatly as I should:
-- Executin
I am having several users complain about not being able to use the # button when
dialing into IVR's, etc, because the # key prompts for transfering the call to another
extension. Is there a way to still provide transfer capability, but not use the # key?
I am using SNOM 200 phones so if anyone h
Has anyone been able to get 3way/Conference working with the snom200 and Asterisk.
According to the documentation for the phones the option should come up when you have
two lines active on the snom phone. Unfortunately, I don't see this option appear and
I am now beginning to wonder if this is a
-Original Message-
From: Anton Tinchev [mailto:[EMAIL PROTECTED]
Sent: Thursday, July 08, 2004 11:57 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Audiocodes -> Asterisk Implementation
Brian J. Rathman wrote:
> Anyone out there have the AudioCodes MP-108 working with Aster
Anyone out there have the AudioCodes MP-108 working with Asterisk? I am able to get
the channels to registers with Asterisk, but anytime I try and send a call I receive
these error messages:
Jul 6 15:12:10 DEBUG[1133742896]: chan_sip.c:771 __sip_ack: Stopping retransmission
on '[EMAIL PROTECTE
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