similar to: Number of max SIP calls.

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 > >

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! 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

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

Does anyone tried the Winsip sotware to test Asterisk? _________________________________________________________________ Discover the new Windows Vista http://search.msn.com/results.aspx?q=windows+vista&mkt=en-US&form=QBRE

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

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

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, A radio is being developped at school, and we wanted to avoid eating too much bandwidth, so we had a look at RTP. I first implemented it in icecast1 and it worked well (although we had to apply a patch to make RTP work with xmms). We are indeed saving a lot of bandwidth. I then adapted it to icecast2, here is the patch. The biggest trouble for now is the configurability. I added 3 global

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

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",

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

Dear ALL Please tell me how to configure Openldap in rhel 5 Please send me links & document Thanks in advance Ankit Jariwala 9725655020

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

Hello LLVM community, I was trying to port a target specific loop transformation pass (HexagonVectorLoopCarriedReusePass) to the New Pass Manager. However, I could not figure out a way to register this pass with opt. I can see that llvm/lib/Passes/PassRegistry.def is the registry for target independent passes. Can anyone point me to an example/API which can help me in registering this pass so

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

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

Hi All, I changed the CFG of a big function using a pass that runs on a function, I am having trouble debugging it. Is there someway to dump a module to a file from inside gdb? I have access to the Module pointer. Let me know if anything else is needed. -- *Ankit* -------------- next part -------------- An HTML attachment was scrubbed... URL: