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

[Yusuf]

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?