Thanks in advance
---
Originally "John Voss" said
>Has anyone had experience with Theo's chan_bluetooth with
>asterisk-1.2.0 on FC3.
>
>When I try to make a call to my home through the channel my phone
>shows calling
Has anyone had experience with Theo's chan_bluetooth with asterisk-1.2.0 on
FC3.
When I try to make a call to my home through the channel my phone shows calling
but my home phone never rings.
I get this on screen:
== Spawn extension (outgoing, 3234600, 2) exited non-zero on 'SIP/2299-9131'
comma between flash an the ()
- Original Message -
From: "John Voss"
To: asterisk-users@lists.digium.com
Subject: Has anyone tried using flash() in features.conf (applicationmap)
Date: Wed, 04 Jan 2006 22:51:59 -0500
I have the sample application working and the flash
I have the sample application working and the flash applications executes but I
get the following error on the console.
Feature Found: zapflash exten: zapflash
WARNING[32164]: res_features.c:948 feature_exec_app: Could not find
application (flash())
Flash works within the dialplan so the
Has anyone used this feature?
I have been trying to find documentation on it but can't. I have tried the one
example shown but can't get it to work.
Anyone had success?
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ip-info.org/wiki/view/NAT+and+VOIP
- Original Message -
From: "Dennis Gilmore"
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] RE:IConnecthere dial out problems
Date: Wed, 7 Dec 2005 21:49:01 -0600
Once upon a time Wednesday 07 December 2005 8:42 pm, Joh
Is there a document/wiki/web site that maps the various SIP.conf settings to
the structure of the actual IP packet?
If so please advise.
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On Wed, 7 Dec 2005 13:32:09 -0600 Dennis Gilmore wrote:
>Im having no issues with my outgoing calls but my incoming call registration
>keeps locking up. it seems after awhile it stops sending reregistration
>packets.
>my extentions.conf has
>; ** Dial Out iconnecthere
I can't seem to get my outgoing connections to work with IConnecthere. At one
time it did with v1.0
I can register and receive calls just fine. But can't make them.
Ultimately, the trace ends with a "400 Bad Request" error when you do a SIP
debug.
Has anyone got it to work with v1.2? Don't kno
From the docs on this command it should be available in 1.0.1.
I don't find it under the apps directory even after doing an update -d (which I
understand will add missing files or diretories)
I have also downloaded ver 1.0.5 and looked in its apps directory. It isn't
there either.
Any suggest
I am running on Core 3 also with a voicepulse account.
I found this document quite helpfulwww.voip-info.org/tiki-print.php?page=Asterisk+Fedora+Core+3
I did deviate in that I ran my make of Asterisk itself as follows
cd /usr/src/asterisk
make clean make linux26 make install make samples
Hope it
6 Jan 2005 08:35:50 -0700 (MST)From: Greg Hill <[EMAIL PROTECTED]>Subject: Re: [Asterisk-Users] Glophone/Voiceglo and AsteriskTo: Asterisk Users Mailing List - Non-Commercial Discussion<asterisk-users@lists.digium.com>Message-ID: <[EMAIL PROTECTED]>Content-Type: TEXT/PLAIN; ch
Has anyone managed to get Asterisk to work with Glophone/Voiceglo since this posting.
http://lists.digium.com/pipermail/asterisk-users/2004-February/036559.html
I've tried copying the config in this listing with no success.
One thing that I have noticed is that all the listings that I have found m
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