search for: 98310

...k/capi.conf: deflect=0800123456 ; some 0800 test number - in etc/asterisk/extensions.conf under [tcom-in]: exten => 98765,1,capiCD(0800123456) - made both b channels busy by outcalling on both lines (ISND BRI) - called my msn (98765) by mobile phone. => asterisk says: Nov 16 13:28:53 ERROR[98310]: chan_capi.c:1953 capi_handle_msg: received a call waiting CONNECT_IND Nov 16 13:28:57 ERROR[98310]: chan_capi.c:1953 capi_handle_msg: received a call waiting CONNECT_IND Nov 16 13:29:05 ERROR[98310]: chan_capi.c:1953 capi_handle_msg: received a call waiting CONNECT_IND Nov 16 13:29:09 ERROR[98310...

I'm having sporadic problems with my X100P card. When trying trying to call out I get a message on the console that the channel is busy even though I have bounced asterisk. It seems the card grabs the line and doesn't let go. When I try to call the number I get a busy signal. If I perform a rmmod wcfxo and restart asterisk everything retruns to normal. I have a TDP400P installed along with

...6717890,1) -- Executing Dial("SIP/i2004-a783", "SIP/13136717890@broadvoice") in new stack -- Called 13136717890@broadvoice -- SIP/broadvoice-9afd is making progress passing it to SIP/i2004-a783 -- SIP/broadvoice-9afd answered SIP/i2004-a783 Sep 21 11:31:49 NOTICE[98310]: chan_sip.c:7519 handle_request: Unknown SIP command 'PING' from '69.14.242.210' Sep 21 11:31:50 WARNING[98310]: chan_sip.c:680 retrans_pkt: Maximum retries exceeded on call cf27d_ff22b1386c@192.168.0.106 for seqno 1 (Non-critical Response) Sep 21 11:31:59 NOTICE[98310]: chan_sip.c...

Hi, I upgraded my two asterisk boxes today to the latest cvs (up from 5/3/04). These two boxes talk to eachother via sip, not iax. Since the upgrade, I get the error "Failed to authenticate on INVITE" trying to make calls to/from either box. Removing the secret from each box's sip config seems to work but is utterly braindead. Has anyone seen this? - Eric

I am totally new to this project, I have been trying to set it up now for almost a whole day, and can not understand what is wrong. The following error appears when my phone tries to register: Oct 6 19:08:30 NOTICE[98310]: chan_sip.c:7519 handle_request: Registration from '<sip:2201@192.168.0.5;user=phone>' failed for '192.168.0.253' What exactly does this error mean, apart from that registration failed. Anyone any ideas why it failed? My phone is a IP2006 ( http://www.solwise.co.uk/voip-phon...

...-- Executing Dial("Zap/8-1", "Zap/g3/20|80") in new stack -- Called g3/20 -- Zap/7-1 is ringing -- Zap/7-1 answered Zap/8-1 -- Attempting native bridge of Zap/8-1 and Zap/7-1 -- Started music on hold, class 'default', on Zap/7-1 Jul 23 13:14:18 NOTICE[98310]: channel.c:1252 ast_read: No/unknown event '0' on timer for 'IAX2/2100@221.11.10.30:4569/4'? Jul 23 13:14:18 NOTICE[98310]: chan_iax2.c:2309 iax2_read: I should never be called! == Spawn extension (officeplan, 20, 1) exited non-zero on 'Zap/8-1' -- Hungup 'Zap/8...

...: 2 CALLED NUMBER : 2003 CALLING NUMBER : dave CALLING NAME : Dave Smith LANGUAGE : en CALLED CONTEXT : from-dave USERNAME : g6crv FORMAT : 2 CAPABILITY : 65283 ADSICPE : 2 DATE TIME : 160927376 Dec 23 17:52:34 NOTICE[98310]: chan_iax2.c:5390 socket_read: Rejected connect attempt from 1.142.164.215 Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REJECT Timestamp: 00003ms SCall: 00001 DCall: 00001 [1.142.164.215:4569] CAUSE : No authority found Rx-Frame Retry[ No] -- OSeqno: 00...

...a sample rate of 128 kHz and output it to the Icecast server. My question is, what are the minimum and maximum sample rates that Ices will continue to work with? Thank you, Heidi Young Software Developer SAIC - Sea Technology and Services Division 1007 Scott Ave. Suite A Bremerton, WA 98310 (360) 415-5768 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/icecast/attachments/20040831/224c096b/attachment.htm

...oaded yesterday's version, just to be sure) on a test system with no digium cards in it, so I have installed ztdummy (see logs and screenshots below) as a timing source. When I call the music on hold extension from a Sipura Sip connected analog phone, I hear nothing and start getting Warning[98310]: chan_sip.c:674 retrans_pkt: Maximum retries exceeded on call cb70fdb1-df-408047@192.168.1.17 for seqno 102 (Non-critical Response) As well, I set up a meetme conference, and dial it, the first user (also a Sipura sip phone) gets 'there are no other users on the conference..", which is O...

...ered translator 'g729tolin' from format G729A to SLINR, cost 2 == Registered translator 'lintog729' from format SLINR to G729A, cost 12 I was hoping by letting it ring out, I would get a voice-mail message, but that doesn't work either... *CLI> Jul 13 11:29:08 WARNING[98310]: chan_sip.c:2696 process_sdp: No compatible codecs! -- Executing Dial("SIP/67.23.212.25-0814f830", "SIP/334|20") in new stack -- Called 334 -- SIP/334-26f8 is ringing -- Nobody picked up in 20000 ms -- Executing VoiceMail("SIP/67.23.212.25-0814f830...

...0.252.240.2|30|tr") in new stack -- MGCP mgcp_request(aaln/2@10.252.240.2) -- MGCP cw: -1, dnd: 0, so: 1, sno: 0 -- MGCP mgcp_new(MGCP/aaln/2@10.252.240.2-0) created in state: Down -- Called aaln/2@10.252.240.2 -- MGCP/aaln/2@10.252.240.2-0 is ringing Jun 29 09:46:58 NOTICE[98310]: chan_mgcp.c:2268 handle_response: Terminating on result 400 from aaln/2@10.252.240.2-0 == No one is available to answer at this time -- Endpoint 'aaln/2@10.252.240.2-1' observed 'hf' -- Swapping 1 for 0 on aaln/2@10.252.240.2 -- MGCP Muting 1 on aaln/2@10.252.240.2...

...d license 'G729-700241AB' providing 5 channels == Found total of 5 G.729 licenses == Registered translator 'g729tolin' from format G729A to SLINR, cost 2 == Registered translator 'lintog729' from format SLINR to G729A, cost 12 *CLI> Jul 13 11:29:08 WARNING[98310]: chan_sip.c:2696 process_sdp: No compatible codecs! -- Executing Dial("SIP/67.23.212.25-0814f830", "SIP/334|20") in new stack -- Called 334 -- SIP/334-26f8 is ringing -- Nobody picked up in 20000 ms -- Executing VoiceMail("SIP/67.23.212.25-0814f830...

...> 10 include => 11 [10] exten => 10,1,Dial(SIP/10,45) exten => 10,2,Hangup [11] exten => 11,1,Dial(SIP/11,45) exten => 11,2,Hangup I've tried a "qualify=yes" in the sip.conf, but then the Box will loose connection after a few seconds: Aug 21 13:16:10 NOTICE[98310]: chan_sip.c:7653 sip_poke_noanswer: Peer '10' is now UNREACHABLE! I would be really pleased for any help :) Regards, Marco canreinvide=no ___________________________________________________________ Gesendet von Yahoo! Mail - Jetzt mit 100MB Speicher kostenlos - Hier anmelden...

Is this a new bug, or am I doing something wrong here... ? Same config files as I have always been using, maybe there is a config file change needed? I have been running CVS for a while and just upgraded to 1.0.0 Now I get an RTP error with MGCP, MGCP dies afterwards. Sep 23 11:28:33 WARNING[98310]: rtp.c:711 ast_rtp_offered_from_local: rtp structure is null -- MGCP mgcp_new(MGCP/aaln/2@63.252.240.2-1) created in state: Down -- Endpoint 'aaln/2@63.252.240.2-1' observed 'hu' Thanks Duane Cox

...: 2 CALLED NUMBER : 2003 CALLING NUMBER : dave CALLING NAME : Dave Smith LANGUAGE : en CALLED CONTEXT : from-dave USERNAME : g6crv FORMAT : 2 CAPABILITY : 65283 ADSICPE : 2 DATE TIME : 160927376 Dec 23 17:52:34 NOTICE[98310]: chan_iax2.c:5390 socket_read: Rejected connect attempt from 1.142.164.215 Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REJECT Timestamp: 00003ms SCall: 00001 DCall: 00001 [1.142.164.215:4569] CAUSE : No authority found Rx-Frame Retry[ No] -- OSeqno: 00...

...type=PATH msg=audit(1404144754.513:46370): item=0 name="/var/tmp/" inode=2 dev=fb:01 mode=041777 ouid=0 ogid=0 rdev=00:00 obj=system_u:object_r:tmp_t:s0 audispd: node=myhost.somewhere type=PATH msg=audit(1404144754.513:46370): item=1 name="/var/tmp/sqlite_4h8lCyF8htbWvZb" inode=98310 dev=fb:01 mode=0100600 ouid=8 ogid=12 rdev=00:00 obj=system_u:object_r:tmp_t:s0 audispd: node=myhost.somewhere type=EOE msg=audit(1404144754.513:46370): ... REPEATED MANY TIMES ... audispd: node=myhost.somewhere type=SYSCALL msg=audit(1404144754.533:46373): arch=c000003e syscall=87 success=yes exit...