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...dropped when this happens. Spans are not necessarily in use when this happens. Mar 30 12:01:41 WARNING[81926]: chan_zap.c:6017 zt_pri_error: PRI: Read on 72 failed: Unknown error 500 Mar 30 12:01:41 NOTICE[81926]: chan_zap.c:6733 pri_dchannel: PRI got event: 4 on span 2 Mar 30 12:01:41 WARNING[98311]: chan_zap.c:4938 handle_init_event: Detected alarm on channel 32: Yellow Alarm Mar 30 12:01:41 WARNING[98311]: chan_zap.c:1151 zt_disable_ec: Unable to disable echo cancellation on channel 32 Mar 30 12:01:41 WARNING[98311]: chan_zap.c:4938 handle_init_event: Detected alarm on channel 33: Yellow...

...e their IAX2 jitter buffers turned off? 4. Is there any significance to the "Username (none)" for one of the peers? The Asterisk has both peer and user names for both machines. The caller name shows up, but the callee name is always "(None)" Ideas anyone? Thanks. DEBUG[98311]: File chan_iax2.c, Line 4649 (socket_read): Sending VNAK DEBUG[98311]: File chan_iax2.c, Line 4649 (socket_read): Sending VNAK DEBUG[98311]: File chan_iax2.c, Line 4649 (socket_read): Se...

Hi, I am running * with E100P board. At least every our I got an Error 500 message and ISDN-PRI restarts: Jan 10 12:53:02 WARNING[98311]: File chan_zap.c, Line 5758 (zt_pri_error): PRI: Read on 27 failed: Unknown error 500 Jan 10 12:54:27 WARNING[98311]: File chan_zap.c, Line 5758 (zt_pri_error): PRI: Read on 27 failed: Unknown error 500 Jan 10 13:33:01 WARNING[98311]: File chan_zap.c, Line 5758 (zt_pri_error): PRI: Read on 27 f...

...k -- Set Response Timeout to 5 == CDR updated on Zap/5-1 -- Executing Dial("Zap/5-1", "Zap/g1/1901") in new stack -- Called g1/1901 == D-Channel on span 1 up == D-Channel on span 1 up == D-Channel on span 1 up -- Zap/1-1 is ringing Jul 13 11:50:15 WARNING[98311]: chan_zap.c:6070 zt_pri_error: PRI: received TEI check request for TEI = 73 Jul 13 11:50:17 WARNING[98311]: chan_zap.c:6070 zt_pri_error: PRI: received TEI check request for TEI = 73 == D-Channel on span 1 up == D-Channel on span 1 up == D-Channel on span 1 up -- Zap/1-1 answered Zap/5-1...

hello to asterisk start WARNING {98311] : File chan_sip.c, line 388 (retrans_pkt) : Maximum retries exceeded on call xxxxxxx...xxxxxxxxxx@192.168.0.1 for seqno 102 (Request) with a call from x-lite 38113@fwd.pulver.com WARNING {98311] : File chan_sip.c, line 2002 (__transmit_response): Unable to determine sequence number from ''...

...yone pushing keys - I hear the music on Hold - as does the calling party. Are we somehow managing to sound like the tone for a '#' ???? My BT100 phone is set up for DTMF=info This appears to happens quite randomly. Suggestions? I'm also getting quite a few... May 12 19:51:52 WARNING[98311]: chan_sip.c:542 retrans_pkt: Maximum retries exceeded on call 048449ee0dd198e304320aa505148a76@160.124.48.24 for seqno 152 (Non-critical Request) May 12 19:51:58 WARNING[98311]: chan_sip.c:542 retrans_pkt: Maximum retries exceeded on call 048449ee0dd198e304320aa505148a76@160.124.48.24 for seqno 15...

I am trying to get the handytone 286 to make a very simple call to * and having problems. It registers with * just fine, but when I place a call (to echo test, for example), the RTP stream seems to have problems opening. Here is there error I get in *: WARNING[98311]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries exceeded on call 20d1c411-e210-5f3d-3f88-19035c8fcb26@192.168.2.6 for seqno 0 (Response) When doing traces with ethereal, I see successful SIP and SDP handshakes, but when * sends handytone RTP packets, I see a ICMP Port Unreachable messag...

...lo. Although I can hear the demo etc. now, I notice during asterisk launch I get :- [chan_oss.so] => (OSS Console Channel Driver) == Console is full duplex == Registered channel type 'Console' (OSS Console Channel Driver) == Parsing '/etc/asterisk/oss.conf': Found WARNING[98311]: File chan_oss.c, Line 232 (sound_thread): Read error on sound device: Resource temporarily unavailable I'm running mandrake 9.1 with the ALSA sound driver. In the control center | Hardware | HardDrake | soundcard | help it suggest this is preferred. Is that warning important ? How do I...

...capability=2147483518;version=1;adsicpe=2' -- Called bartosz:password@iaxtel.com/17007591228@iaxtel -- Call accepted by 12.37.165.130 (format GSM) -- Format for call is GSM NOTICE[147466]: File chan_iax2.c, Line 4060 (socket_read): Rejected connect attempt from 12.37.165.130 WARNING[98311]: File chan_iax.c, Line 4042 (socket_read): Received mini frame before first full voice frame WARNING[98311]: File chan_iax.c, Line 4042 (socket_read): Received mini frame before first full voice frame When somebody is trying to call me I am getting also NOTICE [147466]: File chan_iax2.c, Line 40...

...==================================== [ Booting.....[New Thread 49156 (LWP 28288)] [New Thread 65541 (LWP 28289)] .[New Thread 81926 (LWP 28291)] .... -- SIP Seeding '4000' at 4000@192.168.0.202:5060 for 1800 -- SIP Seeding '4001' at 4001@192.168.0.21:5060 for 3600 [New Thread 98311 (LWP 28292)] ...[New Thread 114696 (LWP 28293)] [New Thread 131081 (LWP 28294)] .[New Thread 147466 (LWP 28295)] 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 loadzone Sep 30 10:51:07 WARNING[16384]...

...65541]: File chan_zap.c, Line 5793 (pri_dchannel): Ring requested on channel 1 already in use on span 1. Hanging up owner. Nov 25 10:49:15 WARNING[81926]: File chan_zap.c, Line 5793 (pri_dchannel): Ring requested on channel 1 already in use on span 2. Hanging up owner. Nov 25 10:49:25 WARNING[98311]: File chan_zap.c, Line 5793 (pri_dchannel): Ring requested on channel 1 already in use on span 3. Hanging up owner. Nov 25 10:49:25 WARNING[114696]: File chan_zap.c, Line 5793 (pri_dchannel): Ring requested on channel 1 already in use on span 4. Hanging up owner. A little while back I also...

...:40:50 NOTICE[16384]: Registered Config Engine odbc > Sep 13 05:40:50 NOTICE[16384]: res_config_odbc loaded. > Sep 13 05:40:53 WARNING[16384]: Unable to get our IP address, Skinny disabled > Sep 13 05:40:53 WARNING[16384]: Unable to open /dev/dsp: No such device > Sep 13 07:48:14 NOTICE[98311]: Registration from '<sip:022@192.168.1.92>' failed for '192.168.1.93' > Sep 13 08:30:29 NOTICE[98311]: Registration from '<sip:021@192.168.1.92>' failed for '192.168.1.93' > Sep 13 08:30:29 NOTICE[98311]: Registration from '<sip:021@192.168....

Has anybody out there tried to use two different iconnecthere accounts with Asterisk? What I want to do is use a second account if the first is busy. I have tried the following: exten=>_91NXXNXXXXXX,1,StripMSD,1 exten=>_1NXXNXXXXXX,2,Dial,SIP/BYEXTENSION@iconnect ;iconnect is the first account exten=>_1NXXNXXXXXX,3,Dial,SIP/BYEXTENSION@iconnect2 ;iconnect2 is the second account But that

I've built the latest CVS of asterisk -- not the zaptel or libpri directories, just the asterisk directory. asterisk installs successfully, but there are severe problems. I built this system in the past and ran it, but now building it again fails. This is the CVS as of this morning, 2003-06-13, but I had problems on 06-11/12 as well. After make; make install; make samples; make config, I

...of DTMF eg in Voicemail. I'm running the latest CVS as of now. I'm getting the error... *CLI> -- Executing VoiceMailMain("SIP/phone1-e0dd", "") in new stack -- Playing 'vm-login' (language 'en') **Here I push a button** May 9 18:26:18 WARNING[98311]: chan_sip.c:5027 receive_info: Unable to retrieve DTMF signal from INFO message from 5e9f165cd1aea27c@159.144.58.141 By re-installing an older (cvs checkout -r v1-0_stable asterisk) version - everything works fine again... thats with NO config changes at all.. Has someone removed some support for...

...ION : 2 CALLED NUMBER : 398xxx CALLING NUMBER : 511 CALLING NAME : edguy@pulver.com LANGUAGE : en USERNAME : iaxfwd FORMAT : 4 CAPABILITY : 4 ADSICPE : 2 DATE TIME : 148517838 Asterisk*CLI> Jun 26 12:36:10 NOTICE[98311]: chan_iax2.c:4439 socket_read: Rejected connect attempt from 65.39.205.121 Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REJECT Timestamp: 00001ms SCall: 00003 DCall: 00040 [65.39.205.121:4569] CAUSE : No authority found Asterisk*CLI> Tx-Frame Retry[...

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