similar to: Unable to create RTP session

My asterisk process produced the following errors this morning: Dec 8 10:44:07 WARNING[50315282]: rtp.c:829 ast_rtp_new_with_bindaddr: Unable to allocate socket: Too many open files Dec 8 10:44:07 WARNING[50315282]: chan_sip.c:2352 sip_alloc: Unable to create RTP session: Too many open files Dec 8 10:44:07 WARNING[50315282]: chan_sip.c:8024 sip_request: Unable to build sip pvt data for

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 > >

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, 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, 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.

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

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

I am having a problem when a SIP registration fails. I get the following messages in the log: Sep 25 18:51:02 NOTICE[1125329600]: File chan_sip.c, Line 2874 (sip_reg_timeout): Registration for '<user>@fwd.pulver.com@65.39.205.114' timed out, trying again Sep 25 18:51:02 NOTICE[1125329600]: File chan_sip.c, Line 5119 (handle_request): Registration from

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,

Pentium(R) D CPU 2.80Ghz 3 GB of Ram 250 Gb harddrive i not sure if i'm downloading the right one here as my mechine runs on 1386 scale could someone help us ouit to see if i am downloading the right one Mike CentOS-5.3-i386-bin-DVD.iso -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have searched all over the web and have not really found anything related to this error.... The only thing I found is related to a system stops responding on DTMF, which does not happen here... THe following is the output from the CLI: *CLI> 2004-08-09 17:36:29 DEBUG[229390]: chan_sip.c:2332 sip_alloc: Allocating new SIP call for 640E2D47-E98B11D8-8FDBE54A-5FB5A0CB@65.67.76.30

Dear Sir, I have implemented Samba PDC. Its working fine. But o do Highly Available, I have been trying to make it in 2 node cluster. Everything is running fine. But facing a problem, which I want to share. When I shift PDC to another cluster node. Everything is shifting fine. But my existing user can not log in. The can logged in again if I rejoined that mechine again to domain. I am explaining

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, all.... Honestly I'm a newbie in Linux. For a couple days, I've been trying to build my own PDC server. I've got enough manual resource for guidance, but somehow, untill now I can't log my Win2000 Profesional into the PDC server. I'm using Debian sid kernel 2.6.5, Samba 3.0.2a and a Win2000 box as client. I think there's nothing wrong with my smb.conf ( I guess...

Hi, I'm working on Asterisk and would like to use only Asterisk SIP signalling for my Voip application. I have written my own channel driver and want to integrate my own RTP with Asterisk. SIP signalling is working fine. But i could not find API's to get RTP Port and IP address to start without starting rtp session. The only way I found to receive/send rtp information is by

Hi, The Polycom 600 phones do not natively bridge with Asterisk. I've solved the problem, but I'm not sure how general it is, so I thought I'd ask this list for advice. It's necessary to use a recent Asterisk CVS for this, since there was a problem with session versions in earlier CVS builds. The problem now is the Via field. When the reinvite goes out, the branch number

I know asterisk isn't a real SIP proxy and is more of a multi-protocol pbx with limited SIP support, but... ... is it possible if you have a central registration server that handles all of your dialplan routing and several asterisk PSTN gateways that it routes calls to for an outbound SIP conversation using reinvites and NOT have the registrar box try and send ANY RTP traffic back to the

Wow, thanks Brian! Everything I saw said the patch was only needed on older releases. I've updated several times over the last week. I patched two systems today, one 3/11/05 and one 3/17/05 and now they both work. Should have posted here sooner! Brian G. On Thu, 2005-03-17 at 13:28, Brian Buhrow wrote: > Hello. I'm writing in response to your message to the ASterisk-users >

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]: