Yusuf
2007-Jun-20 07:30 UTC
[asterisk-users] Error: Unable to allocate RTCP socket: Too manyopen 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 > > That will set the max open files to 32768 for you.. The default is 1024, so > I am sure there should be enough once setting 32768... I hope this helps.. > Think it is the same problem... Give it a bash.. > > Stuart Bennett > Technical Engineer > Electrodynamics Frontline Software (Pty) Ltd Nortel and Asterisk Software > Solutions > > http://www.electrodynamics.biz > > -----Original Message----- > From: asterisk-users-bounces at lists.digium.com > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Yusuf > Sent: 15 June 2007 10:34 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] Error: Unable to allocate RTCP socket: Too > manyopen files > > 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) > > Executing [0824537518 at inbound:37] Dial("Zap/11-1", > "SIP/0824537518 at 10.65.138.102|40|L(3600000)") in new stack > -- Setting call duration limit to 3600 seconds. > -- Called 0824537518 at 10.65.138.102 > -- Call on SIP/10.65.138.105-0a67bbd8 left from hold > -- SIP/10.65.138.105-0a67bbd8 answered SIP/sipCloverCSC-b7eba8a8 > -- Packet2Packet bridging SIP/sipClCSC-b7eba8a8 and > SIP/10.65.138.105-0a67bbd8 > [Jun 15 09:21:48] WARNING[5306]: channel.c:768 ast_channel_alloc: Channel > allocation > failed: Can't create alert pipe! > [Jun 15 09:21:48] WARNING[5306]: chan_sip.c:3783 sip_new: Unable to allocate > AST channel > structure for SIP channel > [Jun 15 09:21:48] NOTICE[5306]: chan_sip.c:13662 handle_request_invite: > Unable to > create/find SIP channel for this INVITE > -- SIP/iswitch-0a69fb70 is ringing > -- Call on SIP/iswitch-0a69fb70 left from hold > -- SIP/iswitch-0a69fb70 is making progress passing it to > SIP/sipClCSC-b7e2ec78 > -- Call on SIP/iswitch-0a569528 left from hold > -- SIP/iswitch-0a569528 answered Zap/9-1 > [Jun 15 09:21:49] WARNING[5306]: channel.c:768 ast_channel_alloc: Channel > allocation > failed: Can't create alert pipe! > [Jun 15 09:21:49] WARNING[5306]: chan_sip.c:3783 sip_new: Unable to allocate > AST channel > structure for SIP channel > [Jun 15 09:21:49] NOTICE[5306]: chan_sip.c:13662 handle_request_invite: > Unable to > create/find SIP channel for this INVITE > -- SIP/10.65.138.103-0a8c4000 is ringing > -- Call on SIP/10.65.138.103-0a8c4000 left from hold > -- SIP/10.65.138.103-0a8c4000 is making progress passing it to > SIP/sipClCSC-b7e62f28 > -- SIP/10.65.138.103-0a8c4000 is ringing > -- Call on SIP/10.65.138.103-0a8c4000 left from hold > -- SIP/10.65.138.103-0a8c4000 is making progress passing it to > SIP/sipClCSC-b7e62f28 > -- Call on SIP/10.65.138.103-0a8c4000 left from hold > -- SIP/10.65.138.103-0a8c4000 answered SIP/sipCloverCSC-b7e62f28 > -- Packet2Packet bridging SIP/sipCloverCSC-b7e62f28 and > SIP/10.65.138.103-0a8c4000 > == Spawn extension (iaxClover, 0722269331, 37) exited non-zero on > 'SIP/sipClCSC-b7e4cd58' > > -- Executing [0117973000 at inbound:52] GotoIf("Zap/1-1", "0 ? 60") in new > stack > -- Executing [0117973000 at inbound:53] Dial("Zap/1-1", > "SIP/iswitch/27117973000|40|L(3600000)") in new stack > -- Setting call duration limit to 3600 seconds. > -- Called iswitch/27117973000 > [Jun 15 09:22:04] WARNING[5306]: acl.c:378 ast_ouraddrfor: Cannot create > socket > [Jun 15 09:22:04] ERROR[5306]: rtp.c:1861 ast_rtp_new_with_bindaddr: Unable > to allocate > socket: Too many open files > [Jun 15 09:22:04] WARNING[5306]: chan_sip.c:4242 sip_alloc: Unable to create > RTP audio > session: Too many open files > [Jun 15 09:22:04] WARNING[5306]: acl.c:378 ast_ouraddrfor: Cannot create > socket > [Jun 15 09:22:05] WARNING[5306]: acl.c:378 ast_ouraddrfor: Cannot create > socket > [Jun 15 09:22:05] ERROR[5306]: rtp.c:1861 ast_rtp_new_with_bindaddr: Unable > to allocate > socket: Too many open files > [Jun 15 09:22:05] WARNING[5306]: chan_sip.c:4242 sip_alloc: Unable to create > RTP audio > session: Too many open files > [Jun 15 09:22:05] WARNING[5306]: acl.c:378 ast_ouraddrfor: Cannot create > socket > [Jun 15 09:22:06] WARNING[5306]: acl.c:378 ast_ouraddrfor: Cannot create > socket > [Jun 15 09:22:06] ERROR[5306]: rtp.c:1861 ast_rtp_new_with_bindaddr: Unable > to allocate > socket: Too many open files > > > So I stopped Asterisk. I am going to > > increase the ulimit, > also increasing the RTP range, from the default of 10000 - 20000. > I had SElinux on permissive, should I rather just disable it? > > Can anyone give me pointers as to what has gone wrong, and what I can do, > other than the > above to fix it? > > Also, as as aside, what it Packet2PAcket? Reading some of Olle's posts, I > gather there is > two types of brigding technologies? > > > > > > > > > > > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-- thanks, Yusuf