similar to: ast_channel_alloc() losing pvt struct

Today my pbx stopped responding to my sip phones.. looking into the log, here what I got: Jun 23 15:50:05 WARNING[7176]: File rtp.c, Line 586 (ast_rtp_new): Unable to allocate socket: Too many open files Jun 23 15:50:05 WARNING[7176]: File chan_sip.c, Line 1308 (sip_alloc): Unable to create RTP session: Too many open files Jun 23 15:50:05 WARNING[7176]: File chan_sip.c, Line 4655

Dear all, we have encounter problem when starting asterisk in the foreground, "asterisk -vvvvgc" with more 100 SIP calls concurrently. we have set ulimit to the highest value. still has this problem. Is this the problem keeping asterisk in the foreground or this is a bug in SVN 1.2 16771? Apr 5 08:48:36 WARNING[14887]: channel.c:562 ast_channel_alloc: Channel allocation

Hello, we are curently benchmarking an asterisk system 1034 sip users are logged into this system and the test software is trying to establish 400 concurrent calls. In the CLI I see the following messages: Apr 19 14:20:51 WARNING[4045]: rtp.c:911 ast_rtcp_new: Unable to allocate socket: Too many open files Apr 19 14:20:51 WARNING[4045]: acl.c:306 ast_ouraddrfor: Cannot create socket Apr 19

Hi In asterisk channel ,I so number of variable regarding the Codec ,Can anyone explain what are those variable variable means.Below are the variables 1. chan->readformat 2. chan->writeformat 3. chan ->rawreadformat 4. chan ->rawwriteformat 5. chan->nativeformats Thanks Nikhil

I've got a brand new Eicon Diva Server BRI card and I want to configure it with Asterisk. I managed to get asterisk and zaptel to compile and install, I've compiled and installed the drivers for the Diva card and now I need to compile and install the chan_driver for chan_capi. Unfortunately this fails miserably. I get the following messages: I'm using: Kernel 2.6.16.37.4,

Hi, I'm developing a new channel driver for a device. I'm calling ast_channel_alloc from one of my functions, and as I step through the execution of ast_channel_alloc with gdb, everything is being initialized properly. However, once ast_channel_alloc returns, the pvt field of the data structure created is reset to null. I've verified that the structure created in the function and

All, maybe that's a similar issue like the one mentioned in [1], nevertheless I started a new thread. Using public folders: location = maildir:/var/vmail/public/:LAYOUT=fs:INDEXPVT=%h/.public-indexes When deleting a public maildir in filesystem (rm -rf ...) and copying back from backup, it's older than the users' dovecot.index.pvt files. (%h is in another directory). Then the

Hi My hardware PBX run asterisk on vxworks,Because the vxworks not support perl. Now I want to add a callback function to my pbx. now it can store Caller and Called party numbers in queue when Called party is busy Then I malloc a new ast_channel to call.It is should use ast_get_channel_by_exten_locked() or ast_channel_alloc() , my program as follow,But it isn't work, anyone know how to

Any ideas what this cryptic message means ? Despite what the message says, there is no "more information" to be found in the logs on either server ! mail.debug dovecot: dsync-server(<REMOVED>): Debug: sieve: file storage: Created storage directory /my/dir/structure/.dovecot.sieve/tmp mxd mail.debug dovecot: dsync-server(<REMOVED>): Debug: sieve: file storage: Relative path

Hi, I have a Intel Xeon Dual Core server, with 3 GB RAM, running Centos 5.0, Asterisk 1.4.4 and mysql 5.0. It is a kinda high-traffic box, with about 60 concurrent calls. The profile of calls on this box are: Incoming: via a Sangoma A101 via SIP from anothjer SIP server Outgoing all calls that come in are sent out via SIP to yet another SIP server. This morning I has this error: (edited)

This was a bug 1.4.4 It has now been fixed in Asterisk 1.4.5 Stuart Bennett wrote: > Hi Yusuf > > A friend of mine had the same problem with a high volume site.. The problem > lies with a limitation in Linux. Linux will only allow a certain amount of > open files at a time. You will need to add the following line before running > asterisk. > > ulimit -n 32768 > >

What happened to chan_h323.c in the asterisk cvs? I got ast_h323.cpp, ast_h323.h and chan_h323.h but no chan_h323.c. Hence chan_h323.so was not created so no h323 support in asterisk. Just wondering when to expect it again because I was stupid and didn't make a backup of the asterisk code before wiping the directory for a rebuild. -------------- next part -------------- An HTML attachment

Problem solved by going in manually. The log message appears for empty "public" folders. Say, you have a folder X with subfolder Y, where X does not contain any e-mail. The log message disappears if you drop an email into X, then remove it. Puf, gone! So, there seems to be a baby bug in how dovecot manages the index in this case. -------------- next part -------------- An HTML attachment

Hi, we currently work on a fresh setup coming from 2.2.13 going to 2.3.2.1 (0719df592) (Debian Dovecot Repo). We make heavily use of Public Folder with Per-user \Seen flag and Subscription. namespace { type = public separator = / prefix = Verteiler/ location = maildir:/var/vmail/public/example.com/verteiler:SUBSCRIPTIONS=subscriptions/%n:INDEXPVT=~/public/example.com/verteiler

Hi, Introduction: There are two domains, for example (aaa.com, bbb.com). In the aaa.com domain two users (bob at aaa.com, alex at aaa.com). In the bbb.com domain there are no users. In time full sync replication (replication_full_sync_interval) operations in logs appear errors: dovecot: dsync-local(bob at aaa.com): Error: Mailbox INBOX: Failed to set attribute

I downloaded asterisk-12.4.0 (asterisk-12-current.tar.gz, dated 10-JUL_2014) from http://downloads.asterisk.org/pub/telephony/asterisk/. The packaged configured OK. However, I'm getting a compile error: use of undeclared identifier 'pvt'. I've found similar reports for other versions (asterisk-11 and dongle; macports; a couple others), but nothing for 12.4. Any suggestions to

Hi all I have been looking at my asterisk CDR in the mysql database and some channel names are set to "**Unknown**" string. When I look at the code, everybody when calling ast_channel_alloc set a channel format like SIP/%s or Zap/%s Only app_voicemail.c doesn't when sending emails and I don't use voicemail. Why app_voicemail needs to allocate a channel to send emails ? And in

Both updated from CVS within the same script but:- channel.c:44:2: #error "You need newer zaptel! Please cvs update zaptel" channel.c: In function `ast_channel_alloc': channel.c:298: error: `ZT_TIMERPONG' undeclared (first use in this function) channel.c:298: error: (Each undeclared identifier is reported only once channel.c:298: error: for each function it appears in.)

It seems like asterisk-addons in SVN has been broken for the last few weeks: gcc -DHAVE_CONFIG_H -I. -I. -I. -I./ooh323c/src -I./ooh323c/src/h323 - DGNU -D_GNU_SOURCE -D_REENTRANT -D_COMPACT -c src/chan_h323.c -MT chan_h323.lo -MD -MP -MF .deps/chan_h323.TPlo -fPIC -DPIC -o .libs/ chan_h323.lo src/chan_h323.c: In function 'ooh323_new': src/chan_h323.c:250: error: too few arguments

Hi, This morning I had some failed calls. On the console (and in the log) I saw the error "Unable to allocate channel structure". Before I restarted the process, I checked it's memory usage in ps and glanced at my free memory in top. Asterisk was using a normal ammount of memory, about 40M. I don't think this was a system limit. This was running Asterisk v1.0.2. Below is