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 [EMAIL PROTECTED]:37] Dial("Zap/11-1",
"SIP/[EMAIL PROTECTED]|40|L(3600000)") in new stack
-- Setting call duration limit to 3600 seconds.
-- Called [EMAIL PROTECTED]
-- 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 [EMAIL PROTECTED]:52] GotoIf("Zap/1-1", "0 ? 60") in new stack
-- Executing [EMAIL PROTECTED]: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?
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