similar to: Asterisk 1.2 crash: gdb trace on core dump

Would it be sufficiant if I were to copy the makefile and res_config_ldap.c to the res/ directory of my running Asterisk and do make; make install? or do I have to do LIBS=-lldap export LIBS ./configure before that? My asterisk version is 1.2.6. Thanks in advance, Abhishek * * * * On 8/27/07, Gavin Henry <gavin.henry at gmail.com> wrote: > > I see it is res_config_ldap. You'd be

Hi All, I'm running 1.2.9.1 stable. I'm wondering has this patch been applied to stable release or is it still only in CVS. Will this file patch apply correctly to 1.2.9.1 stable? Which file do I patch? I'm guessing app_directory_realtime_1.6.1.patch <http://bugs.digium.com/file_download.php?file_id=4915&type=bug> and config.h.patch

Hello. I'm using asterisk 1.4.30. I've found this patch for app_queue.c : https://issues.asterisk.org/view.php?id=11700 Can I easily implement this by issuing : */wget 'https://issues.asterisk.org/file_download.php?file_id=17192&type=bug' -O - | patch -p0/* ?? Does this mean I have a "patched" asterisk ? (I ask this because some applications require a

I've recently finished an add-on module for CEL logging to MySQL, and it needs to be tested. The feature is being tracked at https://issues.asterisk.org/view.php?id=19058 And the patch is available at https://issues.asterisk.org/file_download.php?file_id=29110&type=bug Thank You, -Jonathan Penny -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello I use xen and I am trying to use the parallel port of Centos 5.2 host from a Windows XP (HVM) guest. I have tried to transfer control of the port by blacklisting lp, parport and parport_pc + adding: ioports = [ "0378-037a" ] to the xen configuration file. However, in the presence of this line, xm create replies with: Error: function takes exactly 4 arguments (3

Hi, I have an audio quality problem regarding IAX2. I have 2 Asterisk servers interconnected via 2 LAN trunks at 1Gbps (no nat, no firewall). One trunk is SIP and the other IAX2. Normally, I use IAX2 but have noticed easily reproducible audio quality problems (voice in/out is OK but there's a "third" noise overlapping with a "scratchy sound" as if it were some kind of

My Asterisk 1.2.40 process crashes regularly in the is_zero_or_null function at: return (*vp->u.s == 0 || (to_integer (vp) && vp->u.i == 0)); My gdb trace is at: http://pastebin.com/raw.php?i=hmhzZxye Other examples here: http://lists.digium.com/pipermail/asterisk-users/2010-March/245927.html Can anyone please help?

I wrote a connection pooling patch because asterisk is not usable with MSSQL without it. If you're using, or would like to use, MSSQL I recommend you to check it out. http://bugs.digium.com/file_download.php?file_id=8809&type=bug Just so you know, this is a diff against 1.2.1 and it's been only tested with 1.2.1 (because that's what we are using in production). To enable pooling,

Hi guys, I've an asterisk 1.2.5 runing as production system. Now it becomes very important to my customer an exact analysis of CDRs for their QoS to their customers. I've been analysing the CDRs, and i notice many entries like this: Calldate |Channel |Source | Clid | Dst | Disposition | Duration

Hello mailing list, I have been porting one of my Asterisk boxes to 1.4 and I have encountered a nasty DTMF problem. What happens is someone might come in to my IVR and enter "12345" and what will actually come through could be along the lines of "12234445". Sometimes it works, sometimes it doesn't. I had this problem with 1.2 back in November but was able to solve it

Hi folks, I repeat "as is" the title of a post someone did a few months ago, since I am facing the same problem and did not see one single answer to his post. Maybe I'll be a little bit more lucky. When I'm trying to issue an Atxfer AMI command, in the asterisk 1.8 branch, what happens is that some DTMF's are sent, like this : [Mar 22 15:46:27] DTMF[5910]: channel.c:3900

Hi, I am testing some IP phones (eg. GXP2000) and noticed that the "early dial" feature works fine with Asterisk 1.4 but not with 1.2. "early dial" is when digits are sent immediately, one by one, and Asterisk replies with a "484 Address Incomplete" and waits for the next digit until a match is found. This is a very useful feature where no dial patterns have to be

Hello List, I have been trying to setup T38 gatewaying with the following setup SIP ->Asterisk -> DAHDI TE410P with Libss7 -> TELCO I'm using asterisk Asterisk 1.8.3.2 and DAHDI Version: SVN-trunk-r9697M Echo Canceller: HWEC I'm aware there's no support for T38 gateway but I have been trying to get the patches https://issues.asterisk.org/view.php?id=13405 to work. It seems

Sorry for top-posting but my webmail forces me to. I added -W to the APPEND line as suggested but I'm still getting the same result: Booting... Altiris, inc. X86PC PreBoot, PXE-2.x Enhanced Build ID=402 PXEPreZero: Invalid PXE Server list format. and the client PC freezes right there. Here's the full content of my dhcp.conf: max-lease-time 86400; ddns-update-style interim;

On Wed, Mar 5, 2014 at 1:55 AM, Vieri <rentorbuy at yahoo.com> wrote: > Sorry for top-posting but my webmail forces me to. Odd. It's been a while since I used Yahoo but I didn't think I had that issue. GMail does default to top-posting but clicking the ellipsis to look at the previous email is enough. > I added -W to the APPEND line as suggested but I'm still getting

I''ve got some problems right now using the new model for relationships in fixtures (by label, not by id) and STI. What I''ve got, is a couple of models: Attachment --> FileDownload Version Version has_one file_download with file_download_id on the versions table In my fixtures, I have two FileDownloads in my attachments.yml like so: hemingway_alpha_zip: size: 100

Thanks to ww and his patch on bug #104, I have successfully implemented Asterisk behind NAT without using STUN or anything crazy. It's quite straight forward. Until this gets tested enough and put into CVS, you will have to patch your chan_sip.c file to do this. I'm sure within the next few days this will get put merged into CVS if no one finds any problems. I tried this on chan_sip.c

Hi, I would like to know if one can set the outgoing caller ID within Asterisk when calls are going out through: 1) an analog POTS line (I suppose not) 2) a telco BRI line (I don't think so) 3) a telco PRI line (maybe) 4) a voip provider (surely) Thanks, Vieri ____________________________________________________________________________________ Moody friends. Drama queens. Your

Hi, Suppose you have 2 identical Asterisk servers and 1 alias IP address that you assign to either one, according to system failures, etc. Also suppose that all SIP clients register requests go to the alias IP address. Imagine server1 fails and server2 gets the alias IP address. Correct me if I'm wrong but I would have to wait at least 60 seconds before most SIP clients re-register to

I just found an old bug report at bugs.digium.com with exactly the same problem. It's really too bad this bug wasn't addressed: http://bugs.digium.com/view.php?id=7315 --- On Fri, 1/9/09, Vieri <rentorbuy at yahoo.com> wrote: > I'm trying to figure out how to reload iax2 (without > breaking existing calls) so it can listen on a new IP > address (like "ip addr