[asterisk-users] PSTN issues
Hope some can help me. I have a PSTN coming into TDM400 into Asterisk. We also have direct telephones connected to the PSTN bypassing the Asterisk. When a call comes in on the PSTN the direct connected phones ring first and if no one picks up , Asterisk picks and get routed to internal sip phones. I am not able to find what I should tune to make the calls always go through asterisk without the direct telephones ringing. Things used to work right, suddenly, I have this problem after a recent storm. Thanks, -braman -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] call not routed
After a power interruption, asterisk doesn't seem to be routing calls and there seems to be a premature timeout and hangups occurring. I am clueless where to look. Can someone in the know, look at the following log and enlighten me if there's a problem, or if it looks normal. From the calling phone, it keeps ringing as if never picked up. Thanks soo much. -braman == xx=calling number x'ed out for confidentiality prupose== Mar 25 09:48:13 VERBOSE[3685] logger.c: -- Starting simple switch on 'Zap/4-1' Mar 25 09:48:14 DEBUG[3685] pbx.c: Function result is 'DEWOLFE ENG xx' Mar 25 09:48:14 VERBOSE[3685] logger.c: -- Executing NoOp(Zap/4-1, CallerID=DEWOLFE ENG xx) in new stack Mar 25 09:48:14 VERBOSE[3685] logger.c: -- Executing Answer(Zap/4-1, ) in new stack Mar 25 09:48:14 DEBUG[3685] chan_zap.c: Took Zap/4-1 off hook Mar 25 09:48:14 DEBUG[3685] chan_zap.c: Enabled echo cancellation on channel 4 Mar 25 09:48:14 DEBUG[3685] chan_zap.c: Engaged echo training on channel 4 Mar 25 09:48:14 VERBOSE[3685] logger.c: -- Executing PlayTones(Zap/4-1, ring) in new stack Mar 25 09:48:14 DEBUG[3685] channel.c: Scheduling timer at 160 sample intervals Mar 25 09:48:14 VERBOSE[3685] logger.c: -- Executing NVFaxDetect(Zap/4-1, 6) in new stack Mar 25 09:48:14 DEBUG[3685] app_nv_faxdetect.c: Preparing detect of fax (waitdur=6ms, sildur=1000ms, mindur=100ms, maxdur=-1ms) Mar 25 09:48:14 DEBUG[3685] channel.c: Generator got voice, switching to phase locked mode Mar 25 09:48:14 DEBUG[3685] channel.c: Scheduling timer at 0 sample intervals Mar 25 09:48:14 DEBUG[3685] app_nv_faxdetect.c: Start of voice token! Mar 25 09:48:15 DEBUG[3685] app_nv_faxdetect.c: Found unqualified token of 0 ms Mar 25 09:48:15 DEBUG[3685] app_nv_faxdetect.c: Start of voice token! Mar 25 09:48:21 VERBOSE[3685] logger.c: -- Executing Goto(Zap/4-1, timeconditions|1|1) in new stack Mar 25 09:48:21 VERBOSE[3685] logger.c: -- Goto (timeconditions,1,1) Mar 25 09:48:21 VERBOSE[3685] logger.c: -- Executing GotoIfTime(Zap/4-1, 08:00-12:00|mon-fri|*|*?ivr-4|s|1) in new stack Mar 25 09:48:21 VERBOSE[3685] logger.c: -- Goto (ivr-4,s,1) Mar 25 09:48:21 VERBOSE[3685] logger.c: -- Executing Set(Zap/4-1, LOOPCOUNT=0) in new stack Mar 25 09:48:21 VERBOSE[3685] logger.c: -- Executing Answer(Zap/4-1, ) in new stack Mar 25 09:48:21 VERBOSE[3685] logger.c: -- Executing Wait(Zap/4-1, 1) in new stack Mar 25 09:48:22 VERBOSE[3685] logger.c: -- Executing Set(Zap/4-1, TIMEOUT(digit)=10) in new stack Mar 25 09:48:22 VERBOSE[3685] logger.c: -- Digit timeout set to 10 Mar 25 09:48:22 VERBOSE[3685] logger.c: -- Executing Set(Zap/4-1, TIMEOUT(response)=10) in new stack Mar 25 09:48:22 VERBOSE[3685] logger.c: -- Response timeout set to 10 Mar 25 09:48:22 VERBOSE[3685] logger.c: -- Executing BackGround(Zap/4-1, custom/RyderBrookWelcome) in new stack Mar 25 09:48:22 DEBUG[3685] channel.c: Scheduling timer at 0 sample intervals Mar 25 09:48:22 DEBUG[3685] channel.c: Scheduling timer at 160 sample intervals Mar 25 09:48:22 VERBOSE[3685] logger.c: -- Playing 'custom/RyderBrookWelcome' (language 'en') Mar 25 09:48:42 DEBUG[3685] channel.c: Scheduling timer at 58 sample intervals Mar 25 09:48:42 DEBUG[3685] channel.c: Scheduling timer at 0 sample intervals Mar 25 09:48:42 DEBUG[3685] channel.c: Scheduling timer at 0 sample intervals Mar 25 09:48:53 VERBOSE[3685] logger.c: -- Timeout on Zap/4-1 Mar 25 09:48:53 VERBOSE[3685] logger.c: == CDR updated on Zap/4-1 Mar 25 09:48:53 VERBOSE[3685] logger.c: -- Executing Goto(Zap/4-1, loop|1) in new stack Mar 25 09:48:53 VERBOSE[3685] logger.c: -- Goto (ivr-4,loop,1) Mar 25 09:48:53 DEBUG[3685] pbx.c: Expression result is '1' Mar 25 09:48:53 VERBOSE[3685] logger.c: -- Executing Set(Zap/4-1, LOOPCOUNT=1) in new stack Mar 25 09:48:53 DEBUG[3685] pbx.c: Expression result is '0' Mar 25 09:48:53 VERBOSE[3685] logger.c: -- Executing GotoIf(Zap/4-1, 0?hang|1) in new stack Mar 25 09:48:53 DEBUG[3685] pbx.c: Not taking any branch Mar 25 09:48:53 VERBOSE[3685] logger.c: -- Executing Goto(Zap/4-1, ivr-4|s|begin) in new stack Mar 25 09:48:53 VERBOSE[3685] logger.c: -- Goto (ivr-4,s,4) Mar 25 09:48:53 VERBOSE[3685] logger.c: -- Executing Set(Zap/4-1, TIMEOUT(digit)=10) in new stack Mar 25 09:48:53 VERBOSE[3685] logger.c: -- Digit timeout set to 10 Mar 25 09:48:53 VERBOSE[3685] logger.c: -- Executing Set(Zap/4-1, TIMEOUT(response)=10) in new stack Mar 25 09:48:53 VERBOSE[3685] logger.c: -- Response timeout set to 10 Mar 25 09:48:53 VERBOSE[3685] logger.c: -- Executing BackGround(Zap/4-1, custom/RyderBrookWelcome) in new stack Mar 25 09:48:53 DEBUG[3685] channel.c: Scheduling timer at 160 sample intervals Mar 25 09:48:53 VERBOSE[3685] logger.c: -- Playing 'custom/RyderBrookWelcome' (language 'en') Mar 25 09:48:59 DEBUG[3685] chan_zap.c: Exception on 13, channel 4 Mar 25 09:48:59 DEBUG[3685] chan_zap.c: Got event On hook(1) on channel 4 (index 0) Mar 25 09:48:59 DEBUG[3685] chan_zap.c: disabled
Re: [asterisk-users] pstn calls not picked up
Zaptel seems to be running. Channel status: Channel: 4 File Descriptor: 13 Span: 1 Extension: Dialing: no Context: from-pstn Caller ID: Calling TON: 0 Caller ID name: Destroy: 0 InAlarm: 0 Signalling Type: FXS Kewlstart Radio: 0 Owner: Zap/4-1 Real: Zap/4-1 Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: yes Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: no Pulse phone: no Echo Cancellation: 128 taps, currently ON Actual Confinfo: Num/0, Mode/0x Actual Confmute: No Hookstate (FXS only): Offhook Is this right ? Verbosity is at least 3 Channel: 1 File Descriptor: 12 Span: 1 Extension: Dialing: no Context: from-internal Caller ID: Calling TON: 0 Caller ID name: Destroy: 0 InAlarm: 0 Signalling Type: FXO Kewlstart Radio: 0 Owner: None Real: None Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: no Pulse phone: no Echo Cancellation: 128 taps, currently OFF Actual Confinfo: Num/0, Mode/0x Actual Confmute: No Hookstate (FXS only): Onhook Verbosity is at least 3 Hope the above helps in helping me. Thanks, -braman On Wed, Mar 24, 2010 at 1:45 AM, ABBAS SHAKEEL shakeel.abbas@gmail.comwrote: Hello, Please Confirm if the dahdi/Zaptel service is running . check your channels status. On Wed, Mar 24, 2010 at 9:29 AM, Balu Raman brama...@gmail.com wrote: I have a PSTN line coming into FXO port 4 on a TDM400P. Incoming calls are not being picked up. I don't find anything unusual in asterisk log. I am clueless where I should look. I also find zapata-additional.conf empty. The trouble started when the system was accidentally shut down and rebooted. Any help ? How do I diagnose if the TDM400P is not fried ? Thanks, -braman -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Shakeel Abbas -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] pstn calls not picked up
I have a PSTN line coming into FXO port 4 on a TDM400P. Incoming calls are not being picked up. I don't find anything unusual in asterisk log. I am clueless where I should look. I also find zapata-additional.conf empty. The trouble started when the system was accidentally shut down and rebooted. Any help ? How do I diagnose if the TDM400P is not fried ? Thanks, -braman -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Festival issues
I have used this [EMAIL PROTECTED] for a while now, 2 years. Only recently, I am trying Festival and on invoking festival --server I get these errors : /usr/share/festival/bin/festival: /usr/lib/libstdc++.so.5: version `CXXABI_1.2' not found (required by /usr/share/festival/bin/festival) /usr/share/festival/bin/festival: /usr/lib/libstdc++.so.5: version `GLIBCPP_3.2' not found (required by /usr/share/festival/bin/festival) I have libstdc++.so.6 on the system and symbolically linking did not fix it either. The festival package is asteriskathome-festival-1.96.zip. Thanks for your help. - balu raman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Refrigerator Alarms
Hi, I want asterisk to call a person on the phone for monitoring the refrigerator storing vaccines. I am clueless where to look. Can someone clue me in ? Thanks, balu raman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Refrigerator Alarms
Omar, I am hoping that there may be some temp sensor interface that can be routed to a pc and if the temp falls out of a range, I can have this event call someone. I know what to do in asterisk to make a call. I have to do some research. may be, someone has already done a similar thing. Has to be event driven. Thanks, balu raman On 10/17/07, Omar A. Sabek [EMAIL PROTECTED] wrote: Balu, Do you want events passed to Asterisk from the refrigerator? Or does a reminder type phone call need to be placed on an interval? Please be more specific, since this sounds like a special purpose refrigerator, does it have any way of passing events to an external device? Omar A. Sabek On 10/17/07, Balu Raman [EMAIL PROTECTED] wrote: Hi, I want asterisk to call a person on the phone for monitoring the refrigerator storing vaccines. I am clueless where to look. Can someone clue me in ? Thanks, balu raman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] .call file problem
No, Atis means 'make it writable'. .call should be removable after execution. - balu raman On 7/31/07, Nitesh Divecha [EMAIL PROTECTED] wrote: Thanks Atis, Yes and the .call executes fine... but after 60 seconds it executes again automatically without any application executing it. Cheers, Nitesh Atis wrote: Is your .call file writable by asterisk? $ chmod 777 sample.call On 7/31/07, Nitesh Divecha [EMAIL PROTECTED] wrote: Hello All, Something strange I found that my .call file is running twice... Just after 60 sec it will run again, without any application invoking it. This is my .call file: - = Channel: SIP/xo-out/19097773456 Callerid: 9097773456 MaxRetries: 3 RetryTime: 30 WaitTime: 15 Context: custom-900 Extension: 900 Priority: 1 I am running Asterisk 1.2.18 on CentOS 4.5. Anyone can help? Cheers, Nitesh ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone having trouble with claling US Domestic on Sellvoip?
Can you tell me, why sellvoip rocks ? I am looking to sign up with someone. Thanks, balu raman On 3/25/07, Stephen Bosch [EMAIL PROTECTED] wrote: Salvatore Giudice wrote: Nothing has changed in my Asterisk configuration and now outbound US is getting nothing, but 403's. Anyone else having the same problem? Inbound calls to my DID's are working fine. Clearly, sellvoip rocks! -stephen ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
