similar to: res_musiconhold.c Bug - Patch to solve?

Hi to all, I recently upgraded a production machine to asterisk 1.4.25. It seems quite stable but after ~5 days of normal operation it core dumped with this result: (gdb) bt #0 0x00516402 in __kernel_vsyscall () #1 0x005b3d20 in raise () from /lib/libc.so.6 #2 0x005b5631 in abort () from /lib/libc.so.6 #3 0x005ebe6b in __libc_message () from /lib/libc.so.6 #4 0x005f3b16 in _int_free ()

Good day Asterisk users, If this is the wrong place to post this, my apologies. However, I'm trying to see where I can get a patch for the res_musiconhold.so module. I have an issue where if someone is placed on hold, or is placed in a queue, after any announcement is played in the queue, or if someone is put on hold, the call is resumed, then is put back on hold, if the same music is still

Hi all, I setup a new asterisk machine and all is working fine for maybe 10 days Today there was the problem that nobody can hear the music during you are waiting in a/the queue/s. Only silence was the answer. Then I want to shutdown asterisk with "stop now" and nothing happends After killing asterisk and restarting everything is ok but I'm realy interested in what happends .

Hi, using asterisk 1.2.5 with mysql in a centos 4.2 (2.6 kernel) no hardware interfaces installed gives me this error. Im a bit new to this so any help will be appreciated. == Parsing '/etc/asterisk/musiconhold.conf': Found Mar 26 00:58:49 WARNING[5171]: res_musiconhold.c:833 moh_register: Unable to open pseudo channel for timing... Sound may be choppy. [chan_oss.so] => (OSS

Good day Asterisk users, If this is the wrong place to post this, my apologies. However, I'm trying to see where I can get a patch for the res_musiconhold.so module. I have an issue where if someone is placed on hold, or is placed in a queue, after any announcement is played in the queue, or if someone is put on hold, the call is resumed, then is put back on hold, if the same music is still

Hey guys. Im trying to setup Music on Hold. If I transfer a call (with dial) I like to put the call on Music on hold.. Here's what I've tried so far: On my I extensions.conf exten =>1,1,WaitMusicOnHold(30) exten =>1,2,Dial(SIP/mateo,18) exten =>1,3,VoiceMail(1001) I have also added this line to [context].. So it looks like that: ;[context] musiconhold=default Additinaly,

Hi, I created an account but when I go to issues.asterisk.org <http://issues.asterisk.org/> It still asks for a client certificate. See this screen shot, hopefully it showswhat I mean. http://firestar-hosting.com/clientcert <http://firestar-hosting.com/clientcert>.png -------------- next part -------------- An HTML attachment was scrubbed... URL:

Apologies if this has been asked before. Does anyone know if I can simply recompile * 1.4.34 over 1.4.24.1? Ie. perform an upgrade from 1.4.24.1 to 1.4.34 by just rebuilding the source files for 1.4.34 over the top of the existing 1.4.24.1 files. Obviously, I will need to keep my config files (and sound files etc) - so I'll back them up first. Also, will I need to stop * to perform this

I'm testing an upgrade of an i686 production machine running 1.4.24.1 to 1.6.0.9. I've installed dahdi-linux-2.1.0.4. But: asterisk -cvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvv Asterisk 1.6.0.9, Copyright (C) 1999 - 2008 Digium, Inc. and others. Created by Mark Spencer <markster at digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.

hi,all i have just set up asterisk 1.6.0.7 rc1 with spandsp 0.0.5 pre4 to ReceiveFAX, link to a E1 (DE410P) using dahdi this can receive the fax from E1 successfully, but i see many error message in the log like this: [Mar 16 09:24:38] ERROR[23540] channel.c: ast_read() called with no recorded file descriptor. when i receive a 5 pages fax, i will see this error message over 200 lines..... it

Hi everybody, quite frequently I build customized RPMs with asterisk-1.4.20.1 including some special patches for it, to install the on CentOS 5. Now I was looking to upgrade to asterisk-1.4.24.1, but the RPM-build is not working anymore with my build environement. In version 1.4.22 the "Makefile" was modified and all the RPM-stuff was removed, same for the

I'm trying to upgrade from 1.4.24.1 to 1.6.0.9 over this weekend. I'm getting: [May 23 10:56:33] ERROR[26017]: chan_sip.c:2922 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data [May 23 10:56:33] ERROR[26017]: chan_sip.c:2922 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data [May 23 10:56:33] ERROR[26017]:

Hi all! A few days I have problems connecting to the Internet on my house and since then my local SIP extensions are no longer registered against the local Asterisk server. I'm using Asterisk 1.4.24.1. I was researching on the Internet and I found that it can be related to a bug of chan_sip, can it be? In this case, is there a possible workaround? Thanks in advance for your reply. Regards,

Hi, I've just discovered IMPORT function existence. It can be use to import values from channel's Variable section but unfortunately, il can't be use to access to values from Info section (I'm referring here to sections Info and Variables dumped by DumpChan application). Is there a way to work around this and access from one channel for instance to another channel's

Hi all, I run an Asterisk 1.4.24.1 and face problem with DTMF. When calling from my mobile phone -Nokia E65- in GSM, Asterisk present me a second tone so I can use the GW. For this I use: exten => s,1,NoOp(One of our workers (${CALLERID(number)}) is calling office) ;callerID is the one of the calling mobile phone exten => s,n,Background(silence/1) ; Nokia E65 send digits in

Hello, I'm getting a KErnel Pannic every time i restart the server, what could be happening? I just make: "shutdown -r now" and the server gets Kernel Panic. I'have to go on site and press the power button Here you have my sotware versions: Asterisk 1.4.24.1 DAHDI Tools Version - 2.1.0.2 DAHDI Version: 2.1.0.4 libpri version: 1.4.10.1 WANPIPE Release: 3.5.4 IS there

I had a working Asterisk 1.4.24.1 installation on SUSE 9 Linux but SIP only. I then downloaded and installed latest Zaptel and could not get Zaptel working. So I downloaded Asterisk again and re-installed. But still problems: Here is my ztcfg output: asterisk:/etc/asterisk # ztcfg -vvvvv Notice: Configuration file is /etc/zaptel.conf line 4: Unable to read Zaptel version information. Zaptel

Hiya, I sent this bugfix to the asterisk-dev mailing list, and modified it as I noticed side effects, but now it appears to be finished. Nobody seemed to notice it there, so I thought I'd post here, as it seems to be something that will be needed as people update to the latest CVS version. So...read on :) Ted programmer_ted@hotmail.com P.S. Read to the very end. The original bugfix

Hi, I'm using Asterisk 1.4.24.1 Is it possible (and recommended) to have realtime peers that are not cleared from memory when 'sip reload' is issued? According to https://issues.asterisk.org/view.php?id=14196 I thought having rtcachefriends=yes would be enough, but this does't seem to work. Thanks, Dan -------------- next part -------------- An HTML attachment was scrubbed...

I'm trying to upgrade from 1.4.24.1 to 1.6.0.9 with a TE120P card. I can't make any connection over the T1. From CLI: ERROR[26017]: chan_dahdi.c:14300 process_dahdi: Unknown signalling method 'pri_cpe' at line 37. cat chan_dahdi.conf cat chan_dahdi.conf [trunkgroups] [channels] language=en ;internationalprefix = 00 ;nationalprefix = 0 context=from-pstn switchtype=national