similar to: qudBRI and transfering calls with the latest RC2.

Hi All, I'm having a problem after reinstalling the operating system. Festival works fine for SIP, but when IAX users are calling the same extension they don't hear the festival and I see the next message on console: NOTICE[3996]: chan_iax2.c:2995 iax2_read: I should never be called! I googled and couldn't find a solution, if somebody can help.... neobase*CLI>

We upgraded our asterisk server to 1.2.18 last night to pick up the security update. Since then, any calls coming in on IAX2 links get dropped if they try to enter a MeetMe conference room. The log shows this: Apr 26 08:33:16 NOTICE[27362]: chan_iax2.c:3167 iax2_read: I should never be called! Hanging up. I've temporarily worked around it by switching our inbound provider to use SIP

I am recieving inbound calls to my asterisk box over IAX. And getting "spawn extension....exited non-zero" errors. Im not entirely sure what this means, and would appreciate any help in fixing my problem. FYI: ********** is the inbound phone number x.x.x.x is a remote asterisk box calling my own asterisk box. When I choose it to dial an internal extension using this dialplan: exten

Hey guys, I'm currently running Asterisk 1.2.18 Under Mandriva Linux. Three Facilities are hooked together via IAX2 (Trunked) over a OpenVPN connection on a 10mbit (uplink/downlink) internet connection. I was parked for around thirty seconds at a remote facility. All of a sudden, the call drops. The log entry was: Jun 8 11:34:14 NOTICE[10458]: channel.c:1918 ast_read: No/unknown

Hi all! I downloaded right mpg123, chabged path to mpg123 binary in app_mp3.c, rebuilt app_mp3.so, and got MusicOnHold to work. But MP3Player refuses to do properly: -- Accepting AUTHENTICATED call from x.x.x.x, requested format = 1024, actual format = 1024 -- Executing Answer("IAX2/maxhome@maxhome/3", "") in new stack -- Executing

I am running CVS HEAD (on a Linux-PPC machine.) My current dialplan generates an error at the console in asterisk when I attempt to issue the MP3Player command -- I can't figure out why it's not playing the actual audio file? The rest of the dialplan works great. Here's what I see in the console: -- Executing MP3Player("IAX2/income-in-01@IP",

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Hello ! I 'm having a machine running asterisk 1.6.0.10 with IAX and dahdi. The calls are going in and out from IAX2 to dahdi (chan_dahdi + libpri) and vice versa. After a period of time, I got the following scenario: NOTICE[860] chan_iax2.c: I should never be called! WARNING[752] channel.c: Exception flag set on 'IAX2/iax-peer-13262', but no exception handler WARNING[752] channel.c:

I received this before, and it is because you are using the wrong context in the iax.conf. For example: The context must match the username in the register statement. Iax.conf... register => username:secret@iax.host.net [username] type=friend context=iax-in user=username secret=secret auth=plaintext host=iax.hust.net ----------------------------------------------------------------

I did the following: - chan_capi-0.3.5/Makefile: uncommented CFLAGS+=-DDEFLECT_ON_CIRCUITBUSY - recompile asterisk + chan_capi - added /etc/asterisk/capi.conf: deflect=0800123456 ; some 0800 test number - in etc/asterisk/extensions.conf under [tcom-in]: exten => 98765,1,capiCD(0800123456) - made both b channels busy by outcalling on both lines (ISND BRI) - called my msn (98765) by mobile

Hi, I'm having trouble getting IAX calls through in one direction. On the receiving end I see:- Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 00014ms SCall: 00001 DCall: 00000 [1.142.164.215:4569] VERSION : 2 CALLED NUMBER : 2003 CALLING NUMBER : dave CALLING NAME : Dave Smith LANGUAGE : en CALLED CONTEXT :

Hello! I am semi-new to asterisk, I've been toying with it for about a month now. I'm using the current version from the CVS server (as of last week) and have succesfully connected it to Broadvoice, as well as a few softphones. I got a Nortel i2004 IP phone from a friend to setup at home as a homephone using Broadvoice, but going through my * box and using iptel.com for toll free calls.

Hello! Asterisk box receiving calls. Is there some way to get information about current calls from external or AGI application? I'm interested in: - duration, how long calls already in the system (billing and actual time); - source/destination phone numbers; - etc. In other words can I receive information which we are usually getting in CDRs during the time when the call is still active?

Version: 1.0.beta3 OS: Redhat AS 4 I use the cvs version of Dovecot to run my IMAP server. When I move some mails from one folder to another, only 1 or 2 mails are moved, while others were holded in the tmp directory of the target folder. So, what's the matter? I use becky and Outlook as client, and both of the two have the same problem. Thanks :) -- Xueron Nee <xueron at

Hello, I'm looking for a way to restore old behaviour (before Arkadia patch #0010668) of ForkCDR application in 1.4.18 I've done some research directly in the code (cdr.c & forkcdr.c), but can't find any flag. I am just f*c*ed or do you have something to suggest ? :) Thank you for help. Mathieu

Is H323 built into the current CVS? If so, could someone give me an idea of a simple config? thanks, darran -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030429/fffec1d1/attachment.htm

Dear all! W2k, R 2.5.1 I am working with an ongoing malting barley variety evaluation within Sweden. The structure is 25 cultivars tested each year at four sites, in field trials with three replicates and 'lattice' structure (the replicates are divided into five sub blocks in a structured way). As we are normally keeping around 15 varieties from each year to the next, and take in 10 new

The Asterisk.org development team has released Asterisk versions 1.2.30.3, 1.4.23-rc2, 1.6.0.2, 1.6.1-beta3, as well as Asterisk-Addons versions 1.6.0.1 and 1.6.1-rc2. These releases are available for immediate download from http://downloads.digium.com/. This update for Asterisk includes a fix for a regression introduced in Asterisk 1.2.30 and Asterisk 1.4.21.2 and has existed in the

All, aiming at releases later this or early next week, please test! Thanks, Jan

Hello, Is there some things removed from the kernel.org kernels before the get into xen-unstable? I was just about to compile a new dom0 kernel, and couldn''t find the option for Buslogic SCSI driver i xen-unstable kernel synced earlier this morning. I find this a bit strange, as I know the option is there in kernel.org''s 2.6.16-rc2. The config option was somehow lost in 2.6.15