similar to: Asterisk 1.0.1 Too many open files

Hi, I'm having trouble explicitly forcing a codec between sip devices. Am I missing something or is this not really possible? I have a grandstream registering to asterisk, named sip0. Sip0 registers, via sip, to another asterisk box, sip1. When I place a call from the grandstream, it will travel through sip0 to sip1, where it is then placed to the PSTN. Nothing can reinvite, this path is

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 My asterisk is stoping. i am using asterisk with ser on same mechine here is the asterisk trace ------------------------------------------------ -- Setting call duration limit to 3000 seconds. Sep 2 15:58:12 WARNING[10334]: rtp.c:852 ast_rtp_new_with_bindaddr: Unable to allocate socket: Too many open files Sep 2 15:58:12 WARNING[10334]: chan_sip.c:2313 sip_alloc: Unable to create RTP

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 > >

Hi all. I have an old windows VM with an oldish cygwin that I use for the regression tests. Investigating one of the test failures, I see that it's for UsePrivilegeSeparation=sandbox, and it seems to be because setrlimit(RLIMIT_FSIZE, ...) is not supported. IMO, this isn't a big loss, since the most useful thing in the rlimit "sandbox" is the descriptor limits. Can anyone see

I'm reposting this to the list.. My spam filters didn't like the list host. :( If anyone was able to respond to the mail below, can you send it again please? Thanks. ------------------------------------------------------------------------- Hi, I'm having trouble explicitly forcing a codec between sip devices. Am I missing something or is this not really possible? I have a

Hello all, It used to work on calling my GTalk ID from another GTalk user. But now that I tried calling it again, the caller hears only a ringtone and disconnected after a few rings. The messages on my Asterisk-1.4.21.2 are the following: [Jan 29 10:37:51] ERROR[1303]: rtp.c:1945 ast_rtp_new_with_bindaddr: Unexpected bind error: Cannot assign requested address [Jan 29 10:37:51] WARNING[1303]:

I know I'm getting tired of looking at code, but why is the function register_verify defined in 2 different files? chan_iax2.c line 3860 static int register_verify(int callno, struct sockaddr_in *sin, struct iax_ies *ies) chan_sip.c line 4869 /*--- register_verify: Verify registration of user */ static int register_verify(struct sip_pvt *p, struct sockaddr_in *sin, struct sip_request *req,

Hello, CVS 07/31/03. Test with 130+ PSTN-to-SIP calls. Asterisk gets locked ... grep -e "Error" -e "eventually" p-console chan_sip.c line 1453 (sip_alloc): Error obtaining mutex: Device or resource busy chan_sip.c line 1453 (sip_alloc): Got it eventually... chan_sip.c line 1453 (sip_alloc): Error obtaining mutex: Device or resource busy chan_sip.c line 1453 (sip_alloc): Got

Hello all, When I generate an out-going call from *, the DTMF detection is not working ? ASTERISK --> GW AS5300 --> PSTN But the DTMF is working correctly when it's an incoming call. PSTN - -> GW AS5300 - -> ASTERISK Well, I tried with the 3 dtmfmode of asterisk inband, rfc2833 and info, no way !!! Is it normal that asterisk try to setup the outgoing-call using ULAW ? if I

Hi Joel, We essentially have the following directory structure: /images/yyyy/mm/dd/nn For each day of the year we get in the region of 120,000 images, which are split down into 100 subdirectories to prevent their being too many files in one directory. The directory /images is shared out as follows: [images] comment = Imaging Files path = /images public = yes writable = yes

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, The systrace and rlimit sandboxes have been committed and will be in snapshots dated 20110623 and later. This diff adds support for pre-auth privsep sandboxing using the OS X sandbox_init(3) service. It's a bit disappointing that the OS X developers chose such as namespace-polluting header and function names "sandbox.h", "sandbox_init()", etc. It already forced me to

Hello! Unfortunately we are seeing lately (2-3 times during a day) that asterisk seems to hang up somehow - no new calls can be made and sip show peers and other commands show no obvious problem. We then recompiled 1.2.7.1 with all the DEBUG_ turned on in the makefile and now we see the following messages: May 17 06:46:05 ERROR[8606]: ../include/asterisk/lock.h:236

Hi, we're in the process of setting up large-page support on IBM regattas, but for large-page support the users have to have a set of extra capabilities (CAP_BYPASS_RAC_VMM,CAP_PROPAGATE). This are configured on a per user basis by listing which capability each user have in /etc/security/user. Unfortunately they don't get set when the users log in via OpenSSH (3.1p1). Does anybody know

Hi, I noticed that the glusterfs client tries to set ulimit -n to 1M. When I run booster with non-privileged user, the following line appears several times in the log file: [2009-09-14 09:15:22] W [client-protocol.c:6010:init] brick-0-0-0: WARNING: Failed to set 'ulimit -n 1M': Operation not permitted When I run it with root, there's no such complaint even though

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

> First, src/vsnprintf.[ch] where not added to cvs, version checked out > off CVS does not build. You're right. I forgot to cvs add them. They are now in the repository. > Second, src/main.c *still* does not build on Linux 2.4.x based systems - > it is necessary to #include <sys/resource.h> somewhere. I could have sworn that I have the 2.4 headers installed on this

What does this mean? Asterisk 1.2.13 talking to 1.4.0. (response from 1.4.0.) Yuan Liu