I am having a problem when a SIP registration fails. I get the following messages in the log: Sep 25 18:51:02 NOTICE[1125329600]: File chan_sip.c, Line 2874 (sip_reg_timeout): Registration for '<user>@fwd.pulver.com@65.39.205.114' timed out, trying again Sep 25 18:51:02 NOTICE[1125329600]: File chan_sip.c, Line 5119 (handle_request): Registration from '<sip:<user>@fwd.pulver.com>' failed for '192.168.1.249' Sep 25 18:51:02 NOTICE[1125329600]: File chan_sip.c, Line 4571 (handle_response): Failed to authenticate on REGISTER to '<sip:<user>@fwd.pulver.com>;tag=as75fc26a2' That continues until asterisk*CLI>sip show channels 65.39.205.114 <usr>@fwd. 3ff9e23356a 00103/00000 00000ms 0000ms UNKN 65.39.205.114 <usr>@fwd. 3ff9e23356a 00102/00000 00000ms 0000ms UNKN 504 active SIP channel(s) tail -4 /var/log/asterisk/messages Sep 25 18:31:00 WARNING[1125329600]: File rtp.c, Line 708 (ast_rtp_new): Unable to allocate socket: Too many open files Sep 25 18:31:00 WARNING[1125329600]: File chan_sip.c, Line 1479 (sip_alloc): Unable to create RTP session: Too many open files Sep 25 18:31:01 WARNING[1125329600]: File rtp.c, Line 708 (ast_rtp_new): Unable to allocate socket: Too many open files Sep 25 18:31:01 WARNING[1125329600]: File chan_sip.c, Line 1479 (sip_alloc): Unable to create RTP session: Too many open files I blackholed the route to see if any of them would time out. No dice. #asterisk -r Asterisk CVS-09/18/03-23:43:55, Copyright (C) 1999-2001 Linux Support Services, Inc. Written by Mark Spencer <markster@linux-support.net> ========================================================================Broken pipe At this point I need to restart asterisk.
When calling from Zap (E100P) to ATA186 (SIP) * hanged up...
below is 'show channels' command output:
Channel (Context Extension Pri ) State Appl. Data
SIP/565-adc3 (voip 1 ) Up AppDial (Outgoing
Line)
Zap/31-1 (incoming 565 2 ) Ringing Dial
SIP/565|60|r
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20040420/18dd567a/attachment.htm
i have configured a sip phone to make calls through a sip server but when i make call through the sip phone to the sip server every thing goes well and the call is done perfectly but on sip server it gives me these messages(i have 2 pc with different ips one with a sip phone and the another with an asterisk ): Aug 12 18:19:11 NOTICE[12149]: chan_sip.c:5284 register_verify: Peer 'wassim' is trying to register, but not configured as host=dynamic Aug 12 18:19:11 NOTICE[12149]: chan_sip.c:8730 handle_request_register: Registration from '<sip:wassim@195.112.214.98>' failed for '195.112.214.98' Aug 12 18:19:11 WARNING[12149]: chan_sip.c:7910 handle_response: Forbidden - wrong password on authentication for REGISTER for 'wassim' to '195.112.214.98' Aug 12 18:19:31 NOTICE[12149]: chan_sip.c:4380 sip_reg_timeout: -- Registration for 'wassim@195.112.214.98' timed out, trying again any body have an idea. __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com
Hi, On Upgrading do 1.2 I do have an PRoblem with my VOIP-Provider. Making outbound calls does result in "Error 400" - exept if I do call my own phonenumber. Usimg my SNOM190 directly or reverting to 1.0.9 does resolve the problem immediately. What has to be changed in SIP config to move from 1.0.9 to 1.2? Elmar
> On Upgrading do 1.2 I do have an PRoblem with my VOIP-Provider. > > Making outbound calls does result in "Error 400" - exept if I do call my > own phonenumber. >I dind find the solution th this problem in current CVS source, chan_sip.c has to be updated. Elmar