[asterisk-users] Error: Unable to allocate RTCP socket: Too manyopen files

Yusuf yusuf at ecntelecoms.com
Wed Jun 20 02:30:26 CDT 2007


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



More information about the asterisk-users mailing list