search for: 196621

...connection needs to be established. It registers with gatekeeper ok though. Here are the symptoms. If the call initiated by SIP device, asterisk replies to it "Trying" and then silently crashes (it launched as asterisk -vvvvcd). In debug log I can see the following: Jul 16 18:11:52 DEBUG[196621]: File pbx.c, Line 1123 (pbx_extension_helper): Launching 'Dial' Jul 16 18:11:52 DEBUG[196621]: File chan_oh323.c, Line 1393 (oh323_request): In oh323_request. Jul 16 18:11:52 DEBUG[196621]: File chan_oh323.c, Line 1394 (oh323_request): type=oh323, format=4, data=<phone number>. Jul 1...

...sk spews out on the console: Jan 13 15:45:32 WARNING[180236]: chan_zap.c:5834 zt_pri_error: PRI: Read on 43 failed: Unknown error 500 PRI got event: 5 Jan 13 15:45:36 WARNING[180236]: chan_zap.c:5834 zt_pri_error: PRI: Read on 43 failed: Unknown error 500 PRI got event: 6 Jan 13 15:45:36 WARNING[196621]: chan_zap.c:4781 handle_init_event: Detected alarm on channel 1: Red Alarm Jan 13 15:45:36 WARNING[196621]: chan_zap.c:4781 handle_init_event: Detected alarm on channel 2: Red Alarm Jan 13 15:45:36 WARNING[196621]: chan_zap.c:4781 handle_init_event: Detected alarm on channel 3: Red Alarm (repe...

Hi- I've posted this on the bugs list, but I'd also like to see if others have had similar problems when connecting via E1 trunks (E400P). I'm getting numerous errors like the following during inbound calls to my E1 channels. These occur when the system is under medium load: WARNING[196621]: File chan_zap.c, Line 5404 (zt_pri_error): PRI: !! Got reject for frame 78, retransmitting frame 78 now, updating n_r! WARNING[196621]: File chan_zap.c, Line 5404 (zt_pri_error): PRI: !! Got reject for frame 78, retransmitting frame 79 now, updating n_r! WARNING[196621]: File chan_zap.c, Line 540...

Hi.... I'm having a extrange problem.... I cant register with Iaxtel or call to digium... But i cant make or recive IAX calls... ( I made some one with irc users ) Any idea why? At my logs i have this from iaxtel: NOTICE[196621]: File chan_iax2.c, Line 2832 (register_verify): No registration for peer 'xmarts' (from 192.168.0.11) NOTICE[196621]: File chan_iax2.c, Line 4389 (socket_read): Registration of 'arabe' rejected: Registration Refused And when i tray to call Digium i get: -- Called guest@misery.dig...

This morning, I received a very strange error message on the Asterisk console. The error occurs when I try to access iconnect WARNING[196621]: File chan_sip.c, Line 386 (__sip_xmit): sip_xmit of 0x80d0854 (len 649) to 213.137.73.178 returned -1: Bad file descriptor I also get this error when I try to reload: WARNING[16384]: File chan_sip.c, Line 5355 (reload_config): Unable to get IP address for BusinessOne.telantek.com, SIP disabled...

Hi, I have a problem with asterisk-0.5.0 which I don't understand. The monitor says when making a call: *CLI> -- Executing Dial("SIP/roger-c456", "Modem/ttyI0:BYEXTENSION|60|tTm") in new stack -- Called ttyI0:1234567890 WARNING[196621]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 2 frames WARNING[196621]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 2 frames and the last 2 lines are repeated approx a 100 times every second. Regardless which one of my 2 snom phones is used. I have a...

...extension (simstream-sip, t, 2) exited non-zero on 'SIP/7831-76f0' #################################### Trace og outgoing call ############################## #################################### Error Message I often get ############################################# Jun 11 11:11:18 WARNING[196621]: chan_zap.c:6070 zt_pri_error: PRI: received TEI check request for TEI = 127 #################################### Error Message I often get ############################################# ######################################### Zapata.conf ################################# [channels] ; ; ISDN q...

...TRUNKTWO}${CALLEDNO:1}) I have changed password for first trunk to simulate trunk failure. Trunk one tries to dial, it gets rejected by remote peer and then it just defaults to look for timeout (exten => t, Something). CLI displays this after dial command for trunk one: Mar 31 15:59:37 WARNING[196621]: chan_iax2.c:4901 socket_read: Call rejected by 213.166.5.130: No authority found -- Hungup 'IAX2[magout]/16384' == No one is available to answer at this time Mar 31 15:59:47 WARNING[344084]: pbx.c:1839 ast_pbx_run: Timeout, but no rule 't' in context 'dial-mobile-peak...

...p and isdn work. The i4l channel has MSN=26. I also put incomingmsn=26,27 in modem.conf. extensions.con: [incoming-isdn] exten => s,1,Dial(SIP/701,20,Ttr) ;will make my extension 701 ring, while exten => 26,1,Dial(SIP/701,20,Ttr) ;will cause an error message in the Cli interface: WARNING[196621]: pbx.c:1814 ast_pbx_run: Channel 'Modem[i4l]/ttyI0' sent into invalid extension 's' in context 'default', but no invalid handler -- Hungup 'Modem[i4l]/ttyI0' Of course, I would like to be able to distinguish between called MSN's so as to be able to exp...

...7 WARNING[16384]: chan_zap.c:9567 setup_zap: Ignoring loadzone Sep 30 10:51:07 WARNING[16384]: chan_zap.c:9567 setup_zap: Ignoring loadzone Sep 30 10:51:07 WARNING[16384]: chan_zap.c:9567 setup_zap: Ignoring defaultzone [New Thread 163851 (LWP 28296)] .[New Thread 180236 (LWP 28297)] [New Thread 196621 (LWP 28298)] .. -- Logging into database with user asterisk, password xxxxxx, and database asteriskvmusers ........[New Thread 213006 (LWP 28300)] ........................................Sep 30 10:51:12 WARNING[98311]: chan_sip.c:681 retrans_pkt: Maximum retries exceeded on call 333ab105721da...

Thanks for the info so far. I am still trying to asterisk'ize my ML9.2 firewall box and can't get the external SIP registration to work. If I hook up my Sipura directly to the WAN it registers OK. This is the message I get from asterisk: Mar 8 21:03:07 NOTICE[196621]: chan_sip.c:3140 sip_reg_timeout: Registration for '263872@192.246.69.223' timed out, trying again If tried all combinations of firewall and asterisk settings (as well as turning the firewall completely off). I don't understand why this would now work. ; SIP Configuration for Aste...