similar to: Error: Unable to allocate RTCP socket: Too many open files

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 > >

Hello, I m using asterisk version 1.6.2.0 beta. I m trying to test load on it, for which i m using WINSIP installed at two computers and facing two problems. Problem 1: I got 100 users registered to asterisk from each winsip and then initiates 100 calls from one winsip other winsip. But the problem is approx of 60 calls get mature and asterisk give error for the remaining like shown below.

Hi, We have been running asterisk in our telco interconnect box with ss7 and H323 configured. Everything ran find till now, however, today, it started crashing with the following messages: [Jul 28 14:56:55] WARNING[2968]: acl.c:541 ast_ouraddrfor: Cannot create socket [Jul 28 14:56:55] ERROR[2968]: chan_h323.c:962 __oh323_rtp_create: Unable to locate local IP address for RTP stream [Jul 28

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

Hello folks, since I started with asterisk 1.8.2 I got this messages in my console when finish a call. -- Executing [1610 at from-e1:1] Dial("SIP/xxx-00000027", "SIP/1610,60") in new stack == Using SIP RTP CoS mark 5 -- Called 1610 -- SIP/1610-00000028 is ringing -- SIP/1610-00000028 answered SIP/xxx-00000027 -- Locally bridging SIP/xxx-00000027 and

Hi all, While trying to track down a T.38 issue, I came across a series of log entries like this: ============================================================================ [Jan 26 10:23:31] WARNING[32508]: udptl.c:948 ast_udptl_new_with_bindaddr: Unable to allocate socket: Too many open files [Jan 26 10:23:31] ERROR[32508]: acl.c:488 ast_ouraddrfor: Cannot create socket

Hey Guys, Whenever i calling any extension i am getting following WARNING messages do you have any idea they coming from where? -Satish shirley*CLI> == Using SIP RTP CoS mark 5 -- Executing [7623 at from-sip:1] Macro("SIP/7527-00000008", "stdexten,7623,sip/7623&sip/7624") in new stack -- Executing [s at macro-stdexten:1] Dial("SIP/7527-00000008",

I am integrating a new server (Asterisk 1.8.2.3, DAHDI 2.4.0) with VOIP only trunks, and this server only has soft phones. When I dial an extension and the phone is not registered, I don't get any ring or progress indications, and eventually the Dial() times out and drops into voicemail (as expected). CLI output: -- Executing [s at macro-StdExten:6] Dial("IAX2/barneveld-2036",

Greetings list, One of our asterisk boxes has been spitting out the following error this morning: Oct 15 12:31:50 WARNING[22300]: acl.c:306 ast_ouraddrfor: Cannot create socket Looking at the list archives, it seems this is usually caused by insufficient file handles on very heavily loaded systems. Is there anything else that can cause it? For now I've killed and restarted the asterisk

Hi guys I have a site on Asterisk 1.8.11.0 running in Centos 6.5 that has about 150 concurrent callers. I keep getting these types of messages in the CLI: [May 21 11:39:21] WARNING[18469]: channel.c:1189 __ast_channel_alloc_ap: Channel allocation failed: Can't create alert pipe! Try increasing max file descriptors with ulimit -n [May 21 11:39:21] WARNING[18469]: chan_sip.c:7041 sip_new:

Following is my scenario to connect back to back PRI of two asterisk server. PRI cards are Sangoma A102D [Asterisk1]------------[PRI]-Cross Cable---------[Asterisk2] Asterisk1 ; Span 1 (MASTER) switchtype = national ; commonly referred to as NI2 context = from-pstn group = 0,24 echocancel = yes signalling = pri_net channel => 1-23 Asterisk2 ; Span 1 switchtype = national ; commonly

Hi, I am taking over an asterisk system from another person and having an issue where a sip trunk is restricting the outgoing codecs to just g.729 I am dialing in from a Cisco 7960. The Invite from the Cisco has the folowing M line: m=audio 17022 RTP/AVP 18 0 8 101. So it is allowing g.729, ulaw and alaw. Asterisk is tandeming the call out over a SIP trunk sip.conf tandem trunk config:

Hi All, I'm trying to upgrade asterisk server to 1.8.x from my asterisk 1.6, But when making A Call from SIP Client, I got cli Warning ... and no call has been made. My Sip Client is using lib java peers client http://peers.sourceforge.net/ with standard codec PCMU/PCMA [Jan 28 23:03:32] WARNING[1654]: chan_sip.c:8942 process_sdp: Unsupported SDP media type in offer: audio 0 RTP/AVP 0 8

Good Day All, I am facing a serious problem since I started to use asterisk. I don?t know if it is a bug or some one already solved this. Currently I am running 120 VoIP SIP channels on my asterisk server but each day 2, 3 calls got hanged in asterisk, and on asterisk CLI ?show channels? showing us as call UP but in real there is no call. When asterisk restarted the hanged calls removed from

Hi, i got Ubuntu 6.06 installed and theres a problem with asterisk. I've sucessfully installed it with the command: #apt-get install asterisk Then after installing FreePBX i get this error when restarting asterisk: root@hernandezz-laptop:/home/hernandezz# asterisk -rvvvvvvvvvv Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) After looking at the logs i

Hi, I did an update from dovecot 2.2.10 -> 2.2.13 and now I'm getting errors: May 14 10:40:00 imap(hansdampf): Error: open(/var/lib/dovecot/db/shared-mailboxes) failed: Permission denied Currently the permissions on shared-mailboxes rw-rw-rw but teh error still exists. Any hints? Thansk and Regards. G?tz -- G?tz Reinicke IT-Koordinator Tel. +49 7141 969 82 420 E-Mail goetz.reinicke

Hi, I have been (attempting) to transition a company from in-house dovecot 1.x to a hosted dovecot 2.2 setup. I am running the doveadm sync command, and for the four mailboxes have been blocked -- sync'ing seem to be copying the same mails, over and over (note, initially I was using doveadm backup but my reading has indicated that 'doveadm sync' is better) Example: # date

Ok guys, its official -- wxWindows will become wxWidgets (see below). Curt -----Original Message----- From: Julian Smart [mailto:julian@anthemion.co.uk] Sent: Friday, February 20, 2004 10:57 AM To: wx-users@lists.wxwindows.org; wx-dev@lists.wxwindows.org; wx-discuss@lists.wxwindows.org Subject: [wx-dev] wxWindows name change Dear wxFolk, As you may be aware, the project is changing its name

Hi, dont ask :) but I have two accounts for which I have to configure an auto reply for any e-mail send to it. As far as I understand Sieve yet, I do have to set sieve_vacation_min_period = 0 in the dovecot config and set the days value in the sieve rule. But as there is a limit of 1 day, I think this is not working. Right? Is there any other way to accomplish that? regards . G?tz -- G?tz

Hi, we still migrate a couple of users from mbox to maildir. Most of them are fine without any problem. Some crash and dont get all mails synced. first I sync backup the mbox to a maildir-NEW, than copy mbox to a new name, mv maildir-NEW to maildir. Now some problematic mailusers miss some mails from the old mbox, so I start a dsync mirror and BAM crash. e.g.: dsync(s108019): Panic: file