[Asterisk-Users] Unable to create RTP session
Kamran Ahmad
p_kami at yahoo.com
Fri Sep 2 07:52:23 MST 2005
Hello
My asterisk is stoping. i am using asterisk with ser
on same mechine
here is the asterisk trace
------------------------------------------------
-- Setting call duration limit to 3000 seconds.
Sep 2 15:58:12 WARNING[10334]: rtp.c:852
ast_rtp_new_with_bindaddr: Unable to allocate socket:
Too many open files
Sep 2 15:58:12 WARNING[10334]: chan_sip.c:2313
sip_alloc: Unable to create RTP session: Too many open
files
Sep 2 15:58:12 WARNING[10334]: chan_sip.c:8202
sip_request: Unable to build sip pvt data for
'5000131 at 192.168.0.11:5060'
Sep 2 15:58:12 NOTICE[10334]: app_dial.c:1084
dial_exec: Unable to create channel of type 'SIP'
== Everyone is busy/congested at this time
-- Executing Hangup("SIP/192.168.0.11-436cd6d8",
"") in new stack
== Spawn extension (default, 5000131, 2) exited
non-zero on 'SIP/192.168.0.11-436cd6d8'
Sep 2 15:58:14 WARNING[10334]: chan_sip.c:695
retrans_pkt: Maximum retries exceeded on call
894517076 at 192.168.0.153 for seqno 1 (Non-critical
Response)
-- SIP/192.168.0.171:5060-5784 is making progress
passing it to SIP/192.168.0.11-437b5310
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