search for: 5126

...cense for 729 but I can't use it this is the message. == Registered translator 'g729tolinb' from format 8 to 6, cost 99999 == Registered translator 'lintog729b' from format 6 to 8, cost 18 == Parsing '/etc/asterisk/enum.conf': Found Asterisk Ready. *CLI> WARNING[5126]: File chan_sip.c, Line 1601 (process_sdp): No compatible codecs! WARNING[5126]: File chan_sip.c, Line 1601 (process_sdp): No compatible codecs! I want to use a snom 100 with this codec. THX -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pi...

I figured out that there is some sort of incompatibility with snom and asterisk's sip. For the first time the authentication looks like: NOTICE[5126]: File chan_sip.c, Line 4424 (handle_request): Failed to authenticate user <sip:800@157.181.25.113>;tag=yiubra2azl for SUBSCRIBE NOTICE[5126]: File chan_sip.c, Line 4486 (handle_request): Registration from '"Levi" <sip:800@157.181.25.113>' failed for '62.112.220.10...

Greetings I have been running * for about a month now. Configuration. (5) Cisco 79xx IP phones (1) XP100P Pentium III (300mhz) 192meg memory Redat 8.0 (updated) It seems to run for about 3-6 hours, then the process stops. I have noticed, that * does not stop, if I do NOT have it register to other sip servers. (FWD and PCH). Here is are the last few lines in the /var/log/asterisk/messages

Hello I'm tryng to install Asterisk and by now I got a first congfiguration working (0ne PBX box and 2 X-lite phone communicating with each other) The problem now is that I keep this annoying message every time: WARNING[5126]: File chan_sip.c, Line 435 (retrans_pkt): Maximum retries exceeded on call 43c827415f83584c0c9dc15b03ed6924@10.1.1.1 for seqno 102 (Request) Do you have a clue? My bet is like it is trying to send a message to a mailbox that doesn't exist Am I right?? How can I cancel this and stop...

...36, callwait = -1, thirdcall = -1 Oct 6 10:55:21 DEBUG[27664]: File chan_zap.c, Line 1996 (zt_setoption): Set option TDD MODE, value: OFF(0) on Zap/18-1 Oct 6 10:55:21 DEBUG[27664]: File chan_zap.c, Line 1028 (update_conf): Updated conferencing on 18, with 0 conference users Oct 6 10:56:15 DEBUG[5126]: File chan_sip.c, Line 657 (create_addr): Setting NAT on RTP to 0 Oct 6 10:56:15 DEBUG[5126]: File chan_sip.c, Line 548 (__sip_ack): Stopping retransmission on '0c2586220717bdda69153c53788bcf1d@172.20.1.67' of Request 102: Found **************************** Working call ****************...

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I am seeing some very peculiar things in the routines that REGISTER my * server with several accounts. I saw this on my console: . . . NOTICE[5126]: File chan_sip.c, Line 1878 (sip_reg_timeout): Registration timed out, trying again NOTICE[5126]: File chan_sip.c, Line 1878 (sip_reg_timeout): Registration timed out, trying again NOTICE[5126]: File chan_sip.c, Line 1878 (sip_reg_timeout): Registration timed out, trying again NOTICE[5126]: Fil...

...RNING[11276]: PRI: !! Got reject for frame 124, retransmitting frame 124 now, updating n_r! Jul 15 17:07:44 WARNING[11276]: PRI: !! Got reject for frame 124, retransmitting frame 125 now, updating n_r! Jul 15 17:11:56 WARNING[11276]: PRI: Read on 66 failed: Unknown error 500 Jul 15 23:08:37 WARNING[5126]: Maximum retries exceeded on call c36ae135242165c3@XXX.XXX.XX.XX for seqno 30406 (Response) Jul 16 05:39:08 NOTICE[11276]: PRI got event: 8 on span 1 Jul 16 06:25:04 NOTICE[5126]: Request to schedule in the past?!?! Jul 17 14:43:43 WARNING[11276]: Ring requested on channel 1 already in use on span...

...n make successful calls if I disable gsm with "disallow=gsm". As soon as I allow gsm the following appears at the console. There are much much more Lines with "File dsp.c, Line 1198" but I cut them for a better survey : --------- Log Start ------------- Asterisk Ready. WARNING[5126]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries exceeded on call 0487461566dd401a0759b84a70318c87@217.140.94.11 for seqno 103 (Request) WARNING[5126]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries exceeded on call 0487461566dd401a0759b84a70318c87@217.140.94.11 for seqno 102 (R...

...:5213 socket_read: Peer 'arlington' is now REACHABLE! May 23 12:58:19 NOTICE[7176]: chan_iax2.c:5616 iax2_poke_noanswer: Peer 'arlington' is now UNREACHABLE! May 23 12:58:29 NOTICE[7176]: chan_iax2.c:5213 socket_read: Peer 'arlington' is now REACHABLE! May 23 13:06:07 NOTICE[5126]: chan_sip.c:6901 sip_poke_noanswer: Peer '6666' is now UNREACHABLE! May 23 13:06:34 NOTICE[7176]: chan_iax2.c:5616 iax2_poke_noanswer: Peer 'arlington' is now UNREACHABLE! May 23 13:06:44 NOTICE[7176]: chan_iax2.c:5213 socket_read: Peer 'arlington' is now REACHABLE! May 23...

...llow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp Content-Length: 266 [ ... ] Sending to 192.168.1.190 : 5060 (non-NAT) [ ... ] Capabilities: us - 524302, them - 285/0, combined - 12 Non-codec capabilities: us - 1, them - 0, combined - 0 DEBUG[5126]: File chan_sip.c, Line 3965 (check_user): Setting NAT on RTP to 0 Reliably Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required [ ... ] ACK sip:2@192.168.1.10 SIP/2.0 [ ... ] DEBUG[5126]: File chan_sip.c, Line 565 (__sip_ack): Stopping retransmission on 'b5288d54-a46c-9e16-ff7c-...

...pet of what I'm getting when I try to call the ATA -- Executing Dial("SIP/-080e8b58", "SIP/0@cisco1|20|Ttr") in new stack -- Called 0@cisco1 -- SIP/cisco1-f5b3 is ringing == Spawn extension (default, 9342199, 1) exited non-zero on 'SIP/-080e8b58' NOTICE[5126]: File chan_sip.c, Line 1814 (process_sdp): Content is 'multipart/mixed;boundary=uniqueBoundary', not 'application/sdp' -- Executing Dial("SIP/-080e8b58", "SIP/0@cisco1|20|Ttr") in new stack -- Called 0@cisco1 -- SIP/cisco1-426e is ringing == Spawn...

...33:41 2005 } >ERROR: ReadOthersDskInfo(): ReadFile(/oradata/dbf/quorum.dbf) failed(5) - (0) bytes read, tid = main:1024 file = oem.c, line = 1396 {Sun Apr 17 10:33:41 2005 } >ERROR: UpdateDskNodeInfo(): WriteFile(/oradata/dbf/quorum.dbf) failed(5) - (0) bytes read, tid = DiskPingThread:5126 file = oem.c, line = 1325 {Sun Apr 17 10:33:41 2005 } I upgraded ocfs to 1.0.13 (from 1.0.11)... -Stopped ocfs -Removed the ocfs module -Upgraded to ocfs 1.0.13 $rpm -qa | grep ocfs ocfs-2.4.9-e-1.0.13-1 ocfs-tools-1.0.10-1 ocfs-2.4.9-e-enterprise-1.0.13-1 ocfs-support-1.0.10-1 -Started ocfs...

...nf) to work with an Adtech Tipcome SIP Phone s-150? Using a conf like this: exten => 100,1,Dial,SIP/100@adtech (in extension.conf) and [adtech] type=friend username=123 secret=123 host=dynamic defaultip=192.168.0.10 (in sip.conf) I have "Registering Error!" on display and NOTICE[5126]: File chan_sip.c, Line 3600 (handle_request): Registration from '<sip:100@192.168.0.1:5060>' failed for '192.168.0.10' on *. Thanks Corrado Ventura

Hello all I already install and able to use Asterisk as voicemail. But after Asterisk is ready, we found lot of following warning(WARNING 5126). Is there any solusion of stopping this warning. Please help me. [Warning} | | | WARNING[1024]: File cdr_pgsql.c, Line 194 (my_load_module): Unable to load config for PostgreSQL CDR's: cdr_pgsql.conf =...

...trace:opengl:wine_glDepthRange (0.000000, 1.000000) trace:opengl:wine_glVertexAttribPointerARB (0, 3, 5126, 1, 24, 0xc942020) trace:opengl:wine_glVertexAttribPointerARB (4, 4, 5121, 1, 24, 0xc94202c)...

Hi all, I'm setting up my first * install and have it peering with another * machine using IAX across the internet which provides our pstn gateway. So far I have the IAX "friend" set up correctly but when I make a test call from an external phone, I get: WARNING[5126]: File chan_iax.c, Line 648 (get_timelen): Don't know how to calculate timelen on 8 packets I have set up a test default context as follows: [default] exten => pstnnumber,1,Answer exten => pstnnumber,2,Playback,demo-thanks exten => pstnnumber,3,Hangup Which doesnt appear to be load...

...e my signalling set to ks for both sides, can make and receive calls just fine. But I never get CallerID on incoming calls. I get the following messages: Aug 11 11:55:35 ERROR[360464]: callerid.c:193 callerid_feed: fsk_serie made mylen < 0 (-26) Aug 11 11:55:35 WARNING[360464]: chan_zap.c:5126 ss_thread: CallerID feed failed: Success Aug 11 11:55:35 WARNING[360464]: chan_zap.c:5168 ss_thread: CallerID returned with error on channel 'Zap/4-1' Or something very similar. Does anybody have any idea how to debug this so I can find out what needs to happen for my CallerID to work...

...----- # () ASCII ribbon campaign - Against html E-mail # /\ # # MotD follows: # If brute force doesn't solve your problems, then you aren't using enough. -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 5126 bytes Desc: not available URL: <http://lists.centos.org/pipermail/centos/attachments/20081127/982809af/attachment-0003.bin>

...webservices#cat tinc-up #!/bin/bash ifconfig $INTERFACE 192.168.98.2 netmask 255.255.255.0 HERE'S the frustrating part: When I star the service, I get (what appears) to be a good startup on both sides: In the management workstation (Mint202): Mar 11 15:59:28 michael-desktop tinc.webservices[5126]: Trying to connect to webservices (173.165.161.166 port 655) Mar 11 15:59:28 michael-desktop tinc.webservices[5126]: Connected to webservices (173.165.161.166 port 655) Mar 11 15:59:28 michael-desktop tinc.webservices[5126]: Connection with webservices (173.165.161.166 port 655) activated On the...