Re: [asterisk-users] Xorcom Bri and asterisk crashes
Thanks for the input, but I still don't seem to have any luck with the devices locking up. I've even rebuilt a new system on new hardware and a new xorcom device but still no good. Once the device locks up that's it the only way to get zaptel and asterisk back up is to turn them off and restart the server. The command you have me rmmod xpp_usb; /usr/share/zaptel/xpp_fxloader reset Works great and will reload the firmware as long as the devices are frozen. Once they lock up this command will not reload the firmware and brings up the following errors. 'xpp_fxloader'[16065]: Resetting FPGA Firmware on /dev/bus/usb/005/016 'xpp_fxloader'[16067]: /usr/sbin/fpga_load: ERROR (247): bulk_write failed: error reaping URB: No such device 'xpp_fxloader'[16067]: /usr/sbin/fpga_load: ERROR (810): Renumeration to default failed: errno=-19 'xpp_fxloader'[16067]: /usr/sbin/fpga_load: ERROR (203): Releasing interface: usb: could not release intf 0: No such device 'xpp_fxloader'[16342]: fpga_load failed remoivng with status 237 I'm running Fedora 7, Kernel 2.6.21-1.3194.fc7 Will hopefully be upgrading the kernel tonight if I can get some downtime to do so. As for more traces, I can do that, but being reasonably new to this I will need some help getting them for you. On Thu, Jul 05, 2007 at 09:14:12AM +1000, Nathan Dennis wrote: We have recently install an asterisk solution with about 60 physical extensions. While the system is running it runs reasonably well (Still have a few teething problems) but twice now they have experienced a degradation in voice quality and dropped calls and then finally asterisk completely crashes out. Restarting asterisk will work for a little while and it will crash again, each time less time will pass before a crash out. The first time I didn't have much logging so I didn't get anything to work with. I have since turned on debugging and following is the logs from the time of the last crash. Can anyone point out where the problem may lay, suggested updates or changes? Jul 4 11:56:51 DEBUG[20042] chan_sip.c: Auto destroying call 'aca7e8d7fc914018 at 192.168.12.164 http://lists.digium.com/mailman/listinfo/asterisk-users ' Jul 4 11:56:54 DEBUG[20042] chan_sip.c: Stopping retransmission on '6eeb52b53a414a6975facbc22ca10686 at 192.168.10.12 http://lists.digium.com/mailman/listinfo/asterisk-users ' of Request 102: Match Found Jul 4 11:56:55 VERBOSE[20050] logger.c: -- Accepting voice call from '' to '40312688' on channel 0/2, span 5 Jul 4 11:56:55 DEBUG[20050] chan_zap.c: Enabled echo cancellation on channel 14 Jul 4 11:56:55 DEBUG[20295] pbx.c: Function result is 'CID withheld' Jul 4 11:56:55 VERBOSE[20295] logger.c: -- Executing Set(Zap/14-1, CALLERID(name)=Old Main Line:CID withheld) in new stack Jul 4 11:56:55 VERBOSE[20295] logger.c: -- Executing Goto(Zap/14-1, mainq|q|1) in new stack Jul 4 11:56:55 VERBOSE[20295] logger.c: -- Goto (mainq,q,1) Jul 4 11:56:55 DEBUG[20295] pbx.c: Function result is 'false' Jul 4 11:56:55 VERBOSE[20295] logger.c: -- Executing Set(Zap/14-1, NightMode=false) in new stack Jul 4 11:56:55 DEBUG[20295] pbx.c: Expression result is '0' Jul 4 11:56:55 VERBOSE[20295] logger.c: -- Executing GotoIf(Zap/14-1, 0?afterhoursq|q|1) in new stack Jul 4 11:56:55 DEBUG[20295] pbx.c: Not taking any branch Jul 4 11:56:55 VERBOSE[20295] logger.c: -- Executing GotoIfTime(Zap/14-1, 8:00-17:30|mon-fri|*|*|?businesshours) in new stack Jul 4 11:56:55 VERBOSE[20295] logger.c: -- Goto (mainq,q,5) Jul 4 11:56:55 VERBOSE[20295] logger.c: -- Executing Set(Zap/14-1, __ALERT_INFO=http://www.example.com http://www.example.com/ ;info=MainQ) in new stack Jul 4 11:56:55 VERBOSE[20295] logger.c: -- Executing Queue(Zap/14-1, mainq1|twr|||10) in new stack Jul 4 11:56:55 DEBUG[20295] chan_zap.c: Requested indication 3 on channel Zap/14-1 Jul 4 11:56:55 VERBOSE[20295] logger.c: -- Called Local/700 at callagents http://lists.digium.com/mailman/listinfo/asterisk-users Jul 4 11:56:55 VERBOSE[20298] logger.c: -- Executing Set(Local/700 at callagents-bc5a http://lists.digium.com/mailman/listinfo/asterisk-users ,2, Extension=700) in new stack Jul 4 11:56:55 VERBOSE[20298] logger.c: -- Executing Set(Local/700 at callagents-bc5a http://lists.digium.com/mailman/listinfo/asterisk-users ,2, __ALERT_INFO=http://www.example.com http://www.example.com/ ;info=MainQ) in new stack Jul 4 11:56:55 VERBOSE[20298] logger.c: -- Executing Dial(Local/700 at callagents-bc5a http://lists.digium.com/mailman/listinfo/asterisk-users ,2, SIP/700||tw) in new stack Jul 4 11:56:55 DEBUG[20298] chan_sip.c: Setting NAT on RTP to 524288 Jul 4 11:56:55 DEBUG[20298] chan_sip.c: Setting NAT on VRTP to 524288 Jul 4 11:56:55 DEBUG[20298] chan_sip.c: Outgoing Call for 700 Jul 4 11:56:55 VERBOSE[20298] logger.c: -- Called 700 Jul 4 11:56:55 DEBUG[20042] chan_sip.c: (Provisional) Stopping
Re: [asterisk-users] Which features are lost when canreinvite is turned on ?
You mean I'm heading to NAT issues ? And what about Record-Route options ? Will it really help to be notified of call endings ? ___ --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
[asterisk-users] CTI application controling Asteridk
Hi, I have two boxes : - A asterisk server. - A Python Server doing CTI and call control. If a call come on the Asterisk, a sound will be played continually Then, If somebody want to pick up this call, he will click on a Webpage (using the Python server) that will ask the Asterisk box to join this call to this user. In fact, the asterisk as to do something ( like play a file) and at the same wait for a interuption to go on the next step (call someone). Do you have a idea on how I can do this properly ? Thanks a lot for your help, Thomas ___ --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] Fax and Asterisk
- Lee Howard [EMAIL PROTECTED] wrote: Andrew Nowrot wrote: I am trying to build reliable fax solution with asterisk, iaxmodem and hylafax. I am attempting to do this on Compaq DL-360 with 2 pentium 3 1.2 GHz (512 cache) and 2GB of RAM. I am using a Sangoma A101. After installing the newest zaptel and wanpipe-3.1.0 beta I did zttest and it didn't give me good results: 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% Are you having trouble with fax? Rumor is it that the Sangoma hardware isn't as needy this way as is the Diguim. I'm not sure about that, though. That should not be relevant anymore. You should see approximately the same reliability with both cards, as long as the card and timing is configured correctly. Matthew Fredrickson ___ --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] Xorcom Bri and asterisk crashes
On Mon, Jul 09, 2007 at 04:02:06PM +1000, Nathan Dennis wrote: Thanks for the input, but I still don't seem to have any luck with the devices locking up. The trace you posted before mentioned tasklets. Those are not in use in the Astribank driver code (unless you set the optional parameter pcm_tasklet of the driver xpp, which is almost always discourged) and IIRC are not used in the main zaptel code. Some other zaptel drivers use tasklets, and also ztdummy uses them. But I still suspect that the error is elsewhere. Again, oops / BUG() traces would be helpful to trace the issue. I've even rebuilt a new system on new hardware and a new xorcom device but still no good. Once the device locks up that's it the only way to get zaptel and asterisk back up is to turn them off and restart the server. The command you have me rmmod xpp_usb; /usr/share/zaptel/xpp_fxloader reset Works great and will reload the firmware as long as the devices are frozen. Once they lock up this command will not reload the firmware and brings up the following errors. 'xpp_fxloader'[16065]: Resetting FPGA Firmware on /dev/bus/usb/005/016 Here xpp_fxloader has identified that the device 005 016 has an Astribank that could use resetting. It calls fpga_load: 'xpp_fxloader'[16067]: /usr/sbin/fpga_load: ERROR (247): bulk_write failed: error reaping URB: No such device fpga_load has managed to open the device, but has failed to write. I'm not sure if this is the first write or not. 'xpp_fxloader'[16067]: /usr/sbin/fpga_load: ERROR (810): Renumeration to default failed: errno=-19 'xpp_fxloader'[16067]: /usr/sbin/fpga_load: ERROR (203): Releasing interface: usb: could not release intf 0: No such device 'xpp_fxloader'[16342]: fpga_load failed remoivng with status 237 I'm running Fedora 7, Kernel 2.6.21-1.3194.fc7 Will hopefully be upgrading the kernel tonight if I can get some downtime to do so. As for more traces, I can do that, but being reasonably new to this I will need some help getting them for you. On Thu, Jul 05, 2007 at 09:14:12AM +1000, Nathan Dennis wrote: We have recently install an asterisk solution with about 60 physical extensions. While the system is running it runs reasonably well (Still have a few teething problems) but twice now they have experienced a degradation in voice quality and dropped calls and then finally asterisk completely crashes out. Restarting asterisk will work for a little while and it will crash again, each time less time will pass before a crash out. The first time I didn't have much logging so I didn't get anything to work with. I have since turned on debugging and following is the logs from the time of the last crash. Can anyone point out where the problem may lay, suggested updates or changes? Jul 4 11:56:51 DEBUG[20042] chan_sip.c: Auto destroying call 'aca7e8d7fc914018 at 192.168.12.164 http://lists.digium.com/mailman/listinfo/asterisk-users ' Jul 4 11:56:54 DEBUG[20042] chan_sip.c: Stopping retransmission on '6eeb52b53a414a6975facbc22ca10686 at 192.168.10.12 http://lists.digium.com/mailman/listinfo/asterisk-users ' of Request 102: Match Found Jul 4 11:56:55 VERBOSE[20050] logger.c: -- Accepting voice call from '' to '40312688' on channel 0/2, span 5 Jul 4 11:56:55 DEBUG[20050] chan_zap.c: Enabled echo cancellation on channel 14 Jul 4 11:56:55 DEBUG[20295] pbx.c: Function result is 'CID withheld' Jul 4 11:56:55 VERBOSE[20295] logger.c: -- Executing Set(Zap/14-1, CALLERID(name)=Old Main Line:CID withheld) in new stack Jul 4 11:56:55 VERBOSE[20295] logger.c: -- Executing Goto(Zap/14-1, mainq|q|1) in new stack Jul 4 11:56:55 VERBOSE[20295] logger.c: -- Goto (mainq,q,1) Jul 4 11:56:55 DEBUG[20295] pbx.c: Function result is 'false' Jul 4 11:56:55 VERBOSE[20295] logger.c: -- Executing Set(Zap/14-1, NightMode=false) in new stack Jul 4 11:56:55 DEBUG[20295] pbx.c: Expression result is '0' Jul 4 11:56:55 VERBOSE[20295] logger.c: -- Executing GotoIf(Zap/14-1, 0?afterhoursq|q|1) in new stack Jul 4 11:56:55 DEBUG[20295] pbx.c: Not taking any branch Jul 4 11:56:55 VERBOSE[20295] logger.c: -- Executing GotoIfTime(Zap/14-1, 8:00-17:30|mon-fri|*|*|?businesshours) in new stack Jul 4 11:56:55 VERBOSE[20295] logger.c: -- Goto (mainq,q,5) Jul 4 11:56:55 VERBOSE[20295] logger.c: -- Executing Set(Zap/14-1, __ALERT_INFO=http://www.example.com http://www.example.com/ ;info=MainQ) in new stack Jul 4 11:56:55 VERBOSE[20295] logger.c: -- Executing Queue(Zap/14-1, mainq1|twr|||10) in new stack Jul 4 11:56:55 DEBUG[20295] chan_zap.c: Requested indication 3 on channel Zap/14-1 Jul 4 11:56:55 VERBOSE[20295] logger.c: -- Called Local/700 at callagents http://lists.digium.com/mailman/listinfo/asterisk-users Jul 4 11:56:55 VERBOSE[20298] logger.c: -- Executing Set(Local/700 at callagents-bc5a
Re: [asterisk-users] installing * from source
On Sun, Jul 08, 2007 at 05:58:18PM -0400, EdPimentl wrote: Have you also consider adding adding the uBuntu steps in addition to CentOS? -E Ubuntu steps, due to popular demand: apt-get install asterisk zaptel-source m-a a-i zaptel Untested yet. Should work on 7.04 . Bug reports are welcomed. Same instructions work on Debian Stable. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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
[asterisk-users] Monitor events?
Hi all, I would like to know if there is any possibility to send an event when a call is monitored? For both start and stop monitor. There is no event sent on asterisk 1.2 for that monitor case. I did not find any changes regrding that on 1.4. Am I wrong? Is it even possible to send an event when a monitor starts or stop ? Or is this a bad idea. Regards, Daniel ___ --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
[asterisk-users] Background transfers with callback
Hello list, I have successfully set up Asterisk, but girls from our office complain to me that when they hit Flash to transfer a call and pick the number, they need to wait until the call is answered, and only then they could hangup. On the analog PBX we had before the transfer was in background, and if called party did not answer the call, then the call returned to the girl in the office, so she could inform our customer than nobody could answer the call now etc. I would like to achieve such functionality in Asterisk. I have red about blind and atxfer and I believe this is what I want: http://bugs.digium.com/view.php?id=8413 is this already in Asterisk 1.4.4 ? How can I make it the default behaviour when Flash is pressed? Or maybe I need something else? -- .: Jakub Głazik, .: email jabber: zytekatnuxi.pl ___ --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
[asterisk-users] Monitor events?
Hi all, I would like to know if there is any possibility to send an event when a call is monitored? For both start and stop monitor. There is no event sent on asterisk 1.2 for that monitor case. I did not find any changes regrding that on 1.4. Am I wrong? Is it even possible to send an event when a monitor starts or stop ? Or is this a bad idea. Regards, Daniel ___ --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
[asterisk-users] DTLS availablity?
Is DTLS available for Asterisk on any Linux distro? I am most interested in Centos ___ --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] Which features are lost when canreinvite is turned on ?
If you manage to get everything working with canreinvite=yes ( i suppose u figure out nat issues ) then you cant play music on hold , can't record calls , and can't do most of pbx stuff asterisk is capable of .. but dont worry asterisk doesnt disable all this features if canreinvite=on .. like if you have call recording enabled in configuration and also have canreinvite=yes then asterisk wont send reinvite's and media stream will pass thorugh asterisk . For most of pbx canreinvite should be kept off unless you have latency issues , or you are just connecting 2 pbx systems and doing something like billing in between and not touching media stream . On 09/07/07, Olivier [EMAIL PROTECTED] wrote: You mean I'm heading to NAT issues ? And what about Record-Route options ? Will it really help to be notified of call endings ? ___ --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] Need Advice/Suggestion
One of my client requested that he wants to manually shift the dial plan like above as he has flexiable timing sometime he finishes at 3:00pm some time 8pm. I can not give him freepbx access. How about ignoring the time element completely and just telling the client to divert his/her phone before they leave the office? Depending on the device, they can either do that locally on the device, or alternatively, you can program a couple of short codes into your dialplan to allow the client to enable/disable divert. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it/chris.html This email is made from 100% recycled electrons ___ --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
[asterisk-users] Very bad TDMF tone !
hello, all of asteriskers: i am using tdm400P in my office. i tested that TDMF generated by asterisk is so bad. the sound is very soft and quality is so bad. i am using asterisk 1.2.18. most of time, the # key can not be detected correctly. Does anyone has that problem? please give me a hit for that problem! thanks! zhu - 抢注雅虎免费邮箱3.5G容量，20M附件！ ___ --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] Need Advice/Suggestion
Or, if you can have a trigger of some type. If you have say, a database, that stores the current night service status, then you can query that to determine if you should send the call to the after hours steps, or to dial into the phone. Then set up another extension that the internal people can dial to trigger that service. Rob Chris Bagnall wrote: One of my client requested that he wants to manually shift the dial plan like above as he has flexiable timing sometime he finishes at 3:00pm some time 8pm. I can not give him freepbx access. How about ignoring the time element completely and just telling the client to divert his/her phone before they leave the office? Depending on the device, they can either do that locally on the device, or alternatively, you can program a couple of short codes into your dialplan to allow the client to enable/disable divert. Regards, Chris ___ --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] Very bad TDMF tone !
lizhong zhu wrote: hello, all of asteriskers: i am using tdm400P in my office. i tested that TDMF generated by asterisk is so bad. the sound is very soft and quality is so bad. i am using asterisk 1.2.18. most of time, the # key can not be detected correctly. Does anyone has that problem? please give me a hit for that problem! The only time I've heard of that problem is when VoIP is involved. I've never heard of this problem when the call is all analog, like with a TDM400P. ___ --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] Which features are lost when canreinvite is turned on ?
Olivier wrote: Hi, My setup is : PSTN - ISTP Network --- Router - Asterisk -- SIP Phones Phones are located in the same location. I'm thinking about installing new phones in other locations (small agency, home workers), registering those phones to the same Asterisk server. As every location has DSL access, I think I should have those phones directly exchanging RTP data with ITSP media gateway, without passing through Asterisk server, with canreinvite = yes option. Before, trying this, I'm wondering which features I would loose in the process ? Will I keep the ability to : - record CDR, - listen to DTMF tones - ... What do you think ? Regards ___ --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 You would never lose CDR's because of this feature, and your DTMF should be out of band (in sip messages) anyway. A re invite really just makes the audio connect directly between the sip endpoints in a connection, the sip proxies still receive messages. To understand this better you should read this document: http://www.ietf.org/rfc/rfc2543.txt Hope this helps, Anthony ___ --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] Very bad TDMF tone !
- Original Message - From: Eric ManxPower Wieling [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, July 09, 2007 4:40 PM Subject: Re: [asterisk-users] Very bad TDMF tone ! lizhong zhu wrote: hello, all of asteriskers: i am using tdm400P in my office. i tested that TDMF generated by asterisk is so bad. the sound is very soft and quality is so bad. i am using asterisk 1.2.18. most of time, the # key can not be detected correctly. Does anyone has that problem? please give me a hit for that problem! The only time I've heard of that problem is when VoIP is involved. I've never heard of this problem when the call is all analog, like with a TDM400P. ___ To add to what Eric said if you have an FXS port on yoir box try to call out and see what happens. You should not have the same issue. Are you using asterisk 1.2.X ? ___ --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] OpenSer/Asterisk PBX solution
I have some clients using Enswitch (Paid solution). They are real happy with it. - Original Message - From: Bob Gibson To: asterisk-users@lists.digium.com Sent: Wednesday, June 27, 2007 7:37 PM Subject: [asterisk-users] OpenSer/Asterisk PBX solution We have been working a OpenSer/Asterisk solution to replace our Avaya PBXs. The OpenSer is to provide scalability and the Asterisk to provide rich features. I know this has been many times for calling card platforms but I'm not sure if anyone has a good scalable solution they are using on their virtual PBX or in a CPE PBX environment? If so I would like to talk to them about buy their deploying, testing and buying their solution? Bob G. [EMAIL PROTECTED] -- We've Got Your Name at Mail.com Get a FREE E-mail Account Today - Choose From 100+ Domains -- ___ --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] Monitor events?
Daniel Gradecak wrote: Hi all, I would like to know if there is any possibility to send an event when a call is monitored? For both start and stop monitor. There is no event sent on asterisk 1.2 for that monitor case. I did not find any changes regrding that on 1.4. Am I wrong? Is it even possible to send an event when a monitor starts or stop ? Or is this a bad idea. Regards, Daniel ___ --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 There are no events generated when the monitor stops and starts, but since you are implicitly recording in your dialplan one way or another you can just add a userevent step before recording and after. ___ --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] Google acquires Grand Central
I don't see the point of the service provided by GrandCentral. Party A calls party B through GrandCentral. Party B know party A's number and calls party A back, now party A can call party B directly, and party A has party B's directly number. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Panton Sent: Saturday, July 07, 2007 6:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Google acquires Grand Central On 4 Jul 2007, at 17:57, Stephen Bosch wrote: Jaswinder Singh wrote: Think about voicesense which will sense what you are talking and pop in a *relevant* voice ad to spice up conversation :P . If this happens I am going back to tin cans and string. Hmm, time to get that IAX encryption working along wit ZRTP Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --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] Monitor events?
Hi Anthony, are you sure the monitor is started and sotoped via the dialplan ? Anthony Francis wrote: Daniel Gradecak wrote: Hi all, I would like to know if there is any possibility to send an event when a call is monitored? For both start and stop monitor. There is no event sent on asterisk 1.2 for that monitor case. I did not find any changes regrding that on 1.4. Am I wrong? Is it even possible to send an event when a monitor starts or stop ? Or is this a bad idea. Regards, Daniel ___ --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 There are no events generated when the monitor stops and starts, but since you are implicitly recording in your dialplan one way or another you can just add a userevent step before recording and after. ___ --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] Monitor events?
Anthony Francis wrote: There are no events generated when the monitor stops and starts, but since you are implicitly recording in your dialplan one way or another you can just add a userevent step before recording and after. You can also start monitoring through the Manager API in which case you could also generate corresponding user events. It is also possible to map monitoring to dtmf digits in features.conf. In that case generating user events would be hard. So a better solution is probably to add events directly to res_monitor.c so that they fire automatically. =Stefan -- reuter network consulting Neusser Str. 110 50760 Koeln Germany Telefon: +49 221 1305699-0 Telefax: +49 221 1305699-90 E-Mail: [EMAIL PROTECTED] Jabber: [EMAIL PROTECTED] Steuernummern 215/5140/1791 USt-IdNr. DE220701760 signature.asc Description: OpenPGP digital signature ___ --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] Google acquires Grand Central
GrandCentral isn't about hiding your number, it's about reachability. Grand Central gives you a single number that rings your home, office, cell, etc... And provides a single voicemail box for all of those numbers. As Asterisk users, these features do not seem very ground breaking to us, as most of us have got this setup for ourselves already. But for someone with no telephony experience or equipment, it's a great product to have. AR On 7/9/07, Wai Wu [EMAIL PROTECTED] wrote: I don't see the point of the service provided by GrandCentral. Party A calls party B through GrandCentral. Party B know party A's number and calls party A back, now party A can call party B directly, and party A has party B's directly number. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Panton Sent: Saturday, July 07, 2007 6:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Google acquires Grand Central On 4 Jul 2007, at 17:57, Stephen Bosch wrote: Jaswinder Singh wrote: Think about voicesense which will sense what you are talking and pop in a *relevant* voice ad to spice up conversation :P . If this happens I am going back to tin cans and string. Hmm, time to get that IAX encryption working along wit ZRTP Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --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 -- Alex Robar [EMAIL PROTECTED] ___ --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] List delays
John Faubion wrote: Is it just me? After the mail list server upgrade, the average delivery time for messages to the users list is between 4 and 5 days. The Dev I've seen several people mention it taking a few days to send messages. I've usually seen mine in a few minutes. We'll see about this one... sent July 4th at 09:54 CDT (15:54 UTC) Received July 9th, 11:01 EST Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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] Monitor events?
On 7/9/07, Daniel Gradecak [EMAIL PROTECTED] wrote: are you sure the monitor is started and sotoped via the dialplan ? If you're using Monitor() or MixMonitor(), then just add a UserEvent() call just before it in the dialplan. If you're doing monitoring of queues, it's a bit trickier - you have to watch for Join events to see what calls are being enqueued, then when you see a Link event for that call, you can assume (based on local policy) that the monitoring has started (assuming there was no Leave event in the meantime - the logic in your AMI client has to match the logic in your dialplan that deals with queues obviously) If you're talking about automon, there's no support for that, but a cursory examination of the code doesn't show any reason why it couldn't be added. Look at builtin_automonitor() in res_features.c. -- j. ___ --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] Monitor events?
Hi Stefan, actually you probably know i am using your java-asterisk :) Yes the best solution i found till now it was to add those events to res_monitor.c. I wonder why it was not yet done, may be there was a reason or nobody needed it yet. Anyhow this would be a cool feature that others should benefit from too. If i do the patch is that possible for those events to be added in next versions of asterisk ? Regards, Daniel Stefan Reuter wrote: Anthony Francis wrote: There are no events generated when the monitor stops and starts, but since you are implicitly recording in your dialplan one way or another you can just add a userevent step before recording and after. You can also start monitoring through the Manager API in which case you could also generate corresponding user events. It is also possible to map monitoring to dtmf digits in features.conf. In that case generating user events would be hard. So a better solution is probably to add events directly to res_monitor.c so that they fire automatically. =Stefan ___ --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
[asterisk-users] Problems sending more than 2 SMS with asterisk / smsq
When i send more than one messages shortly after the other, my log (/var/spool/asterisk/sms ) looks like this and only two of four messages arrive. What am i doing wrong ? I am using an AVM B1 PCI with chan-capi and 1.4.4. and also, when sending with smsq -x only two of the messages are handled. (i thought, asterisk itself handles the queues ? ) Here the log: 2007-07-09T15:04:14 YOM04 0 - 0172xxx test11 2007-07-09T15:04:15 ?OM05 0 - 0172xxx test12 2007-07-09T15:07:51 YOM06 0 - 0172xxx test13 2007-07-09T15:07:53 ?OM07 0 - 0172xxx test14 sorry - i am a total newbie at asterisk. -- Mit freundlichen Grüssen Matthias Huber ___ --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] Polycom 301 - Problem with AMI Originated Calls
Lee Jenkins wrote:
Hi all,
I'm having an odd problem with my polycom 301. I am initiating a call
to it with AMI Originate() function:
Action: Originate
Channel: local/[EMAIL PROTECTED]
Context: to_meetme
Exten: s
Priority: 1
Variable: dropped_conf=111
The to_meetme context is very simple:
[to_meetme]
exten=s,1,MeetMe(${dropped_conf},id)
If I specify every other device I have to test:
* Grandstream 101
* XLite Client
* My Cell Phone
It works as expected. But with the Polycom, the phone will ring and the
usual ANSWER REJECT FORWARD soft buttons are painted on the display, but
hitting the answer button seems to fail to do anything other than
silence ringing.
SHOW CHANNELS shows the polycom as ringing still although the polycom
has stopped ringing (audibly at least).
Of course, all other calls originate through the dialplan are answered
with no problem.
It appears that it is something with my Polycom configuration. It seems
like the polycom is having a problem with calls that do not contain
correct CID info.
In the originate command, I added some lines to populate the CALLERID(x):
Action: Originate
Channel: local/[EMAIL PROTECTED]
Context: to_meetme
Exten: s
Priority: 1
Variable: CALLERID(num)=${DEV_NAME}|CALLERID(name)=Operator
Async: true
After explicitly setting the Caller ID info, the polycom then accepts
the call correctly.
Anyone know off hand what setting might be creating this behavior?
Thanks again,
--
Warm Regards,
Lee
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Re: [asterisk-users] ipv6 patch
On 01:06, Mon 02 Jul 07, Hans Witvliet wrote: On Sat, 2007-04-07 at 10:57 +0200, Michiel van Baak wrote: Read http://svn.digium.com/view/asterisk/team/blanchet/v6/README-IPV6.txt?view=markup before running this code. Before taking a plunch into the code Marc Blanchet wrote that he's making code ip-version independant. How much of these improvements have already made it into the 1.4 branche? Hans none as far as I can see. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called 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] Google acquires Grand Central
- Alex Robar [EMAIL PROTECTED] wrote: ___ --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 GrandCentral doesn't do anything you can't do with asterisk. What it does do is put those features within reach of an average person by providing a superb user interface for the end user, which allows them to self-administer all of these wonderful features. ___ --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] Google acquires Grand Central
On Mon, 9 Jul 2007, Wendell Hamilton wrote: GrandCentral doesn't do anything you can't do with asterisk. What it does do is put those features within reach of an average person by providing a superb user interface for the end user, which allows them to self-administer all of these wonderful features. Indeed, the success of their accomplishment -- as with most aspects of successful business -- seems to have been in effectively *productising* these solutions, and developing viable business processes and workflow in order to make them viably scale to the masses of end-users. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --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] Monitor events?
James FitzGibbon wrote: On 7/9/07, *Daniel Gradecak* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: are you sure the monitor is started and sotoped via the dialplan ? If you're using Monitor() or MixMonitor(), then just add a UserEvent() call just before it in the dialplan. If you're doing monitoring of queues, it's a bit trickier - you have to watch for Join events to see what calls are being enqueued, then when you see a Link event for that call, you can assume (based on local policy) that the monitoring has started (assuming there was no Leave event in the meantime - the logic in your AMI client has to match the logic in your dialplan that deals with queues obviously) If you're talking about automon, there's no support for that, but a cursory examination of the code doesn't show any reason why it couldn't be added. Look at builtin_automonitor() in res_features.c. -- j. ___ --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 Adding a call to manager_event() in res_monitor.c and rebuilding asterisk is a trivial mod to get the result you desire. ___ --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] List delays
I received your message just a few minutes after you sent it; however, it sometimes takes 3-4 days before I see messages I post coming back to me on the list. --Don Don Kelly PCF Corp Real Support for your Virtual Office 651 842-1000 888 Don Kell(y) 651 842-1001 fax -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Faubion Sent: Wednesday, July 04, 2007 9:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] List delays Is it just me? After the mail list server upgrade, the average delivery time for messages to the users list is between 4 and 5 days. The Dev I've seen several people mention it taking a few days to send messages. I've usually seen mine in a few minutes. We'll see about this one... sent July 4th at 09:54 CDT (15:54 UTC) John ___ --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
[asterisk-users] Setting Appearance on Outbound Calls?
What do I need to do to set the outbound appearance on a call so that it shows up as Unavailable or Private? ___ --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
[asterisk-users] allow third party registration/invitaion
Hi all, I'm running some performance tests over my Asterisk,to simple the test,I want to configure Asterisk to allow third party registration and invitation,so that Asterisk would not check the to head when challenge registartion and from head when challenge invitation,and I can only create one account and register several clients with different numbers using this account.Does Asterisk support that?How to configure?Please advise,thanks a lot. ___ --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 Waiting curiosity...
Is your incoming context using chanisavail, while your internal-dialing context is not, and just sends the call, without checking? Mojo Michael Wareman wrote: Hi, I have (to me) an interesting problem. There are 3 physical extensions, 11, 12 and 13. All hang off Sipura adapters. There is also extension 10 which simply uses 'Dial(SIP/11SIP/12SIP/13)' to call all phones in the house. Incoming calls from outside get sent to 10 in order that they can be answered from any phone.. Now - if (say) 11 is on a call externally, and 12 calls 11 - 11 get's the call waiting beeps, and can 'flash' over to the new incoming call. No problem there. However, if 12 instead calls 10, in the log I see the Dial command sees 11 as 'In Use' and the call never causes the call waiting beep in 11. Any way to change this? Many thanks, Michael. ___ --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] Polycom multiple registrations
The 430's have two line appearances. I'm trying to get the second line registered to a different extension but for some reason it's not allowing me to do this. The first line will register fine but the second line never seems to register no matter how I swap the device ID's and permissions around. I've played with the lineKeys and callsperlinekey settings to no avail. For what you want to do, you'll have to set lineKeys to 1 for both of your registrations. callsPerLineKey can be anything from 1 to up to (I think) 6, your preference. Can you share the reg ... / statement from your phone.cfg file? Also your sip.conf? - Noah ___ --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
[asterisk-users] Basic asterisk Autodialer?
I'm looking for an easy way to make asterisk perform as a basic (broadcast)autodialer. Basically all I want to do is give it a list of phone #'s and a pre-recorded message and have it call each one and play the message or leave it on the person's answering machine. The people I'll be calling are all our customers, etc. so I don't need to do any do-not-call checking. Just call and play the message and move on. Trying to find a way to notify a couple hundred customers that their service has been changed. Anyone have any easy ways to do this? I already have a functioning asterisk server with a POTS interface, etc. thanks Shawn ___ --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] Asterisk Help
Hi Arun - I need help in configuring a auto dialer system using Asterisk. I'm holding my customers number in MySQL want to fetch 10 numbers one time and dial if gets connected and answered by customer wants to play a sequence of message I've tried here is my code to place calls but in this I see no of failure calls are more than 50%. so please advise. $oSocket = fsockopen($strHost, 5038, $errnum, $errdesc) or die(Connection to host failed); fputs($oSocket, Action: login\r\n); fputs($oSocket, Username: $strUser\r\n); fputs($oSocket, Secret: $strSecret\r\n\r\n); fputs($oSocket, Action: Originate\r\n); fputs($oSocket, Channel: $strChannel\r\n); fputs($oSocket, CallerId: $strCallerId\r\n); fputs($oSocket, Context: $strContext\r\n); fputs($oSocket, Exten: $strExten\r\n); fputs($oSocket, Priority: $strPriority\r\n\r\n); fputs($oSocket, Action: Logoff\r\n\r\n); It would be very difficult to determine what's wrong without seeing more information. For a wild guess, I'd say that the individual MySQL records probably have formatting differences that don't match the format of your code. If so, the data that's pulled in may be bad by the time it gets to your dial statements. Just a random guess. Can you send some CLI output, or turn on logging and give us a little more info? - Noah ___ --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] Basic asterisk Autodialer?
[EMAIL PROTECTED] wrote: I'm looking for an easy way to make asterisk perform as a basic (broadcast)autodialer. Basically all I want to do is give it a list of phone #'s and a pre-recorded message and have it call each one and play the message or leave it on the person's answering machine. The people I'll be calling are all our customers, etc. so I don't need to do any do-not-call checking. Just call and play the message and move on. Trying to find a way to notify a couple hundred customers that their service has been changed. Anyone have any easy ways to do this? I already have a functioning asterisk server with a POTS interface, etc. thanks Shawn ___ --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 This can be done with an easy script and call files However The problem you'll run into is with waiting for an answering machine. To my knowledge, there is no way to listen for the beep. So you have 2 options. First, you have it dial the other person and when it connects just let her rip and hopefully it wasn't an answering machine. The situation is okay when a person is picking up at the other end. Or second, you could have it wait for connect, then have it wait 10 seconds, then play. In most cases, this is usually okay. The problem there is that if its a person who picks up, they will hear dead error for 10 secs. You could have have a long message that says please hold for message from 'your company' then have it wait for 10 seconds, then play. This kind of covers both bases, but no guarantees people aren't just going to hang up on it. Hopefully this helps a little. ___ --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] Setting Appearance on Outbound Calls?
Hi Matt - What do I need to do to set the outbound appearance on a call so that it shows up as Unavailable or Private? In most cases, I think you'd need to arrange this with your provider. If you want to do it on a call-by-call basis (in the US), dial *67 before you dial the number. If you have Caller ID blocked permanently, dial *82 to unblock for a given call. There may be other ways to do it, though. One of my clients has a Verizon PRI. If I set the CallerID to an invalid number and call another Verizon landline, it will show up as unavailable. If I do the same and call just about any cell phone, the receiving phone will show the invalid number even if it's something like 000. On this PRI, I think it always works to set it to a validly formatted, but fake number. - Noah ___ --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] Basic asterisk Autodialer?
Shawn,
Just call and play the message and move on. Trying to find a way to
notify a couple hundred customers that their service has been changed.
Anyone have any easy ways to do this? I already have a functioning
asterisk server with a POTS interface, etc.
Set up a dial plan extension that has the effect of calling each
number. Rather than building a dialplan entry for each number, just
make it require the prepending of a special prefix that is then
stripped off, not entirely unlike how trunk groups are used in TDM
routing:
exten = _666NX,1,Macro(dialer-macro,${EXTEN:3})
Or do it all without the use of macros, whatever. Have it Dial() the
customer at ${EXTEN:3}, Background() or Playback() a recording, then hang
up.
Then, load your target numbers into a text file. Write a script to
iterate through them, and trigger the dial plan extension through the
Asterisk Manager API:
http://www.voip-info.org/wiki-Asterisk+manager+API
It's a simple, TCP-based CLI service and is by far the easiest way to do
this. If you have a sound card in the source machine you might even be
able to get away with calling 'console dial [EMAIL PROTECTED]' from the Asterisk
CLI (asterisk -r -x -c), but I think the Manager approach is cleaner.
Let me know if you need some help getting this up and running, I've done
such things before and have some code readily available. If so, contact
me off-list.
Hope that helps,
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: +1-678-954-0670
Direct : +1-678-954-0671
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Re: [asterisk-users] Very bad TDMF tone !
i am using tdm400P in my office. i tested that TDMF generated by asterisk is so bad. the sound is very soft and quality is so bad. i am using asterisk 1.2.18. most of time, the # key can not be detected correctly. Does anyone has that problem? please give me a hit for that problem! The only time I've heard of that problem is when VoIP is involved. I've never heard of this problem when the call is all analog, like with a TDM400P. 1. If asterisk is detecting DTMF, the parameter relaxdtmf= can affect DTMF detection. 2. Have you checked your handsets on both ends of the call? Some handsets try to filter out DTMF tones. 3. Is voice quiet on your calls, too, or is it just DTMF? It's possible to affect overall signal levels in zapata.conf. Can you post the relevant portion of your zapata.conf? - Noah ___ --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] Early Media Handling
Hi Arun - using php script and Asterisk manager I'm dialing numbers and once gets connected send to an exten in my dial plan that plays an automated message but some time without answering even it goes to my exten. How can I handle early media in Asterisk that is I want only when user answer the call it should goto my specified extension. You could put in an explicit Answer() and then a Wait(1). That will force asterisk to answer the call first, wait 1 second, and then move on through the extension's priorities. - Noah ___ --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
[asterisk-users] Digium cards for sale in Pakistan
Hello Users, I have 2x2 port T1/E1 cards for sale in Pakistan. Cards are in warrenty and going cheap as i have purchased additional cards. regards ___ --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] DTLS availablity?
Is DTLS available for Asterisk on any Linux distro? Nope. I've read that the reSIProcate SIP stack has DTLS support. - Noah ___ --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] Basic asterisk Autodialer?
Call files and app_amd (Answering Machine Detection) come to mind. app_amd can take a little time to tune, but you can get it to be pretty reliable in most cases. See: http://www.voipinfo.org/wiki/index.php?page=Asterisk+cmd+AMD http://www.voipinfo.org/wiki/view/Asterisk+auto-dial+out Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ --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] Queue Status
Arun Kumar wrote: Hi I already tried asterisk manager but Im not able to get status for each queue member. thanks That must be a problem with your configuration. I get QueueMemberStatus on my AMI interface (1.2): Event: QueueMemberStatus Privilege: agent,all Queue: support Location: SIP/112 Membership: dynamic Penalty: 0 CallsTaken: 2 LastCall: 1184016974 Status: 1 Paused: 0 -- Warm Regards, Lee ___ --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] List delays
On Wed, 2007-07-04 at 09:57 -0500, John Faubion wrote: Is it just me? After the mail list server upgrade, the average delivery time for messages to the users list is between 4 and 5 days. The Dev I've seen several people mention it taking a few days to send messages. I've usually seen mine in a few minutes. We'll see about this one... sent July 4th at 09:54 CDT (15:54 UTC) John Received july the nineth! (unlike msg's from the suse-lists or aurora-lists, that do arive within minutes...) hw -- pgp-id: 926EBB12 pgp-fingerprint: BE97 1CBF FAC4 236C 4A73 F76E EDFC D032 926E BB12 Registered linux user: 75761 (http://counter.li.org) ___ --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] ipv6 patch
On Sun, 2007-07-01 at 18:27 -0500, Russell Bryant wrote: Hans Witvliet wrote: Before taking a plunch into the code Marc Blanchet wrote that he's making code ip-version independant. How much of these improvements have already made it into the 1.4 branche? None, and they never will make it into the 1.4 branch. We *only* add bug fixes to release branches. New features only go into the development tree for inclusion in the next major release. In this case, that would be Asterisk 1.6. Sorry, wrong question. I intended to ask, wether it would remain for the time being a bleeding-edge-patch, or already included into the svn-tree. Either way, i presume that i shouldn't hold my breath while waiting for the first 1.6 ;-)) (six-months, a year?) Hans -- pgp-id: 926EBB12 pgp-fingerprint: BE97 1CBF FAC4 236C 4A73 F76E EDFC D032 926E BB12 Registered linux user: 75761 (http://counter.li.org) ___ --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
[asterisk-users] Meetme delay?
I recently installed 1.4.5 and I've noticed a recurrence of a problem that I thought was solved long ago, namely a very long (2-4 seconds) delay on meetme calls. That means with two people in the conference room, it takes 2-4 seconds for what one person says to reach the other person. Is anyone else having this problem, and if so, is there a fix or solution? TIA Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 ___ --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
[asterisk-users] Asterisk 1.2.21, 1.4.7 and Libpri 1.2.5, 1.4.1 released
The Asterisk development team is proud to announce a new batch of releases. There are new releases of Asterisk and Libpri for both the 1.2 and 1.4 series. The development team has been working especially hard on fixing bugs in our existing release branches. These releases are regular maintenance releases that include various bug fixes. The ChangeLog in each release tarball contains details on what bugs have been fixed. The contents of the ChangeLog can be viewed through our svn repository viewer. http://svn.digium.com/view/asterisk/tags/1.2.21/ChangeLog?view=markup http://svn.digium.com/view/asterisk/tags/1.4.7/ChangeLog?view=markup http://svn.digium.com/view/libpri/tags/1.2.5/ChangeLog?view=markup http://svn.digium.com/view/libpri/tags/1.4.1/ChangeLog?view=markup The releases are available for download from ftp.digium.com. They are available as both tarballs and patches against the previous release. Thank you for your support! ___ --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] OpenSer/Asterisk PBX solution
Thank you for your input it is very helpful - Original Message - From: Dovid B To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OpenSer/Asterisk PBX solution Date: Mon, 9 Jul 2007 17:05:57 +0300 I have some clients using Enswitch (Paid solution). They are real happy with it. - Original Message - From: Bob GibsonTo: [EMAIL PROTECTED]: Wednesday, June 27, 2007 7:37 PMSubject: [asterisk-users] OpenSer/Asterisk PBX solution We have been working a OpenSer/Asterisk solution to replace our Avaya PBXs.The OpenSer is to provide scalability and the Asterisk to provide rich features.I know this has been many times for calling card platforms but I'm not sure if anyone has a good scalable solution they are using on their virtual PBX or in a CPE PBX environment?If so I would like to talk to them about buy their deploying, testing and buying their solution? Bob [EMAIL PROTECTED] -- We've Got Your Name at Mail.com Get a FREE E-mail Account Today - Choose From 100+ Domains ___ --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 -- We've Got Your Name at http://www.mail.com! Get a FREE E-mail Account Today - Choose From 100+ Domains ___ --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] ipv6 patch
Hans Witvliet wrote: I intended to ask, wether it would remain for the time being a bleeding-edge-patch, or already included into the svn-tree. Either way, i presume that i shouldn't hold my breath while waiting for the first 1.6 ;-)) (six-months, a year?) As far as I know, the patch is ready for use. It has not yet been merged into asterisk trunk, but I don't think there are technical reasons for that. It's just a matter of someone else taking a final look over it, and merging it in. As for a time frame for 1.6, I still don't know. It will be announced when there is one. -- Russell Bryant Software Engineer Digium, Inc. ___ --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] List delays
There is definitely something wrong with this list. I have my emails sorted by date, and every day, the emails do not just come on top, but get slotted in. Today (10 July 2007), I received about 6 emails from 29th of June, couple from 30th, up until the 5th of July, nothing of today's, or, well, for the last 5 days. Admin, get your act together ! ;) -- This message has been scanned for viruses and dangerous content by Mail Call antivirus software, and is believed to be clean. ___ --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] DTLS availablity?
Noah Miller wrote: Is DTLS available for Asterisk on any Linux distro? Nope. I've read that the reSIProcate SIP stack has DTLS support. I found out that DTLS is in openSSL 0.9.8. This is available with Redhat/Centos 5. So the code is there. Perhaps just configuring it to some ports may force use of DTLS? Just like you do with TLS. Well in 2 weeks is IETF, and I can ask Eric directly; its his RFC. ___ --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] Help. Cannot compile version 1.4.6 with the following error
Wai Wu wrote: Hi all, I need the zap channels going, but got the following error. What do I need to change in my configuration? Thnx. chan_zap.c: In function `zap_send_keypad_facility_exec': chan_zap.c:2309: warning: implicit declaration of function `pri_keypad_facility' chan_zap.c: In function `pri_dchannel': chan_zap.c:9292: structure has no member named `call' make[1]: *** [chan_zap.o] Error 1 make: *** [channels] Error 2 You need to download and install the latest libpri first. -- Joshua Colp Software Developer Digium, Inc. - The Genuine Asterisk Experience (TM) ___ --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
[asterisk-users] how to register several clients with different number but using single authentication account ?
Hi all, I'm running some performance tests over my Asterisk,to simplify the test,I want to configure Asterisk to allow several clients registered with different nubmers but using single authentication account,so that Asterisk would not check the to from head when challenging registartion invitation,.for example,I want Asterisk to allow following registration : . To:sip:[EMAIL PROTECTED] From:sip:[EMAIL PROTECTED]:tag=8a3d7e61 .. Authorization:Digest Username=5678,realm=192.168.1.42 ,nonce=4692235235jefsdfq23423fc,uri=sip:192.168.1.42, response=234234234324sfdsf,algorithm=MD5 Does Asterisk support that?How to configure?Please advise,thanks a lot. ___ --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
[asterisk-users] Asterisk 1.4.7 and MOH
I just installed the newly released Asterisk 1.4.7 and I cannot get music on hold. I am using the default settings with the wav files. Here is what I get on the cli from any sip phone: -- Executing [EMAIL PROTECTED]:1] NoCDR(SIP/1120-084e6010, ) in new stack -- Executing [EMAIL PROTECTED]:2] Answer(SIP/1120-084e6010, ) in new stack -- Executing [EMAIL PROTECTED]:3] MusicOnHold(SIP/1120-084e6010, default) in new stack [Jul 9 23:40:44] WARNING[24238]: channel.c:2964 set_format: Unable to find a codec translation path from ulaw to unknown [Jul 9 23:40:44] WARNING[24238]: res_musiconhold.c:702 moh_alloc: Unable to set channel 'SIP/1120-084e6010' to format 'unknown' -- Started music on hold, class '¤Ü', on channel 'SIP/1120-084e6010' [Jul 9 23:40:44] WARNING[24238]: res_musiconhold.c:575 moh0_exec: Unable to start music on hold (class 'default') on channel SIP/1120-084e6010 I do not know why it is not using the default moh. If I try to reload the moh here is what I get: pbx*CLI moh reload pbx*CLI 2 classes reloaded. == Destroying musiconhold processes == Parsing '/etc/asterisk/musiconhold.conf': Found [Jul 9 23:40:51] WARNING[24230]: res_musiconhold.c:638 get_mohbyname: Music on Hold class 'default' not found [Jul 9 23:40:51] WARNING[24230]: res_musiconhold.c:638 get_mohbyname: Music on Hold class 'prueba' not found pbx*CLI The classes are in the file and here is the output from moh show classes: pbx*CLI moh show classes Class: prueba Mode: files Directory: /var/lib/asterisk/moh Class: default Mode: files Directory: /var/lib/asterisk/moh Could this be a bug? -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 ___ --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 fail from audiocode to sip trunk
What error are you getting on the Audio Codes side ? Set verbose to 5 on the
Audio codes box and try running Syslog.
- Original Message -
From: satish patel
To: asterisk-users@lists.digium.com
Sent: Tuesday, June 26, 2007 2:14 PM
Subject: [asterisk-users] call fail from audiocode to sip trunk
Dear ALL
I have audiocode MP -124 with configure in asterisk Endpoint
configuration means every analog phone register in asterisk now thing is that i
have one more SIP trunk with mediant 2000
[auodiocode-mp-124]-[ * ]--[mediant 2000]-E1
When i call from audiocode MP -124 phone i got this error
-- Executing Dial(SIP/20-0889c4d8, SIP/mediant/1) in new stack
-- Called mediant/1
-- SIP/mediant-088a1a18 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Congestion(SIP/20-0889c4d8, ) in new stack
== Spawn extension (mysip, 111, 2) exited non-zero on 'SIP/20-0889c4d8'
-- Executing Dial(SIP/24-0889c4d8, SIP/mediant/0) in new stack
-- Called mediant/0
my extension.conf file is
exten = 43,1,Answer
exten = 43,2,Dial(SIP/43)
exten = 43,3,Hangup
exten = 777,1,Answer()
exten = 777,2,Dial(SIP/777)
exten = 777,3,Hangup()
exten = 888,1,Answer()
exten = 888,2,Dial(SIP/888)
exten = 55,1,Dial(SIP/55)
exten = 66,1,Dial(SIP/66)
exten = _11.,1,Dial(SIP/mediant/${EXTEN:2})
exten = _11.,2,Congestion
what is the problem
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Re: [asterisk-users] callback and bridge problem
Are you behind NAT ? Do you have canreinvite=yes ?
- Original Message -
From: Adam KOSA [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, June 25, 2007 6:37 PM
Subject: [asterisk-users] callback and bridge problem
Hi guys,
sorry for the long e-mail, i'm only trying to give as much information
as i think is relevant to my problem (console log, sip.conf and
extension.conf parts).
i've been practicing with callback for a while, but i'm at a dead end.
I hope somebody can help me to move on.
i have troubles getting two calls bridged together. Scenario is the
following:
- asterisk calls my cell via a SIP provider called neophone
- my cell rings, i pick up, and i find myself in:
[internal]
; callback is directed here
exten = s,1,WaitExten,50
include = voicemail-context
include = internal_extensions-context
include = dialout_prefix-context
because my call file looks like this:
Channel: SIP/[EMAIL PROTECTED]
Context: internal
Extension: s
Priority: 1
where 0620222 is my cell.
- after picking up, i dial 9520630111 where 952 is the dialing
prefix, 0630... is another cell. 952 is a prefix for another
registered account at the same provider (one account is allowed to place
one call at a time).
After this as you can see, the second number (..) is dialed.
However when i pick up the phone, the call hangs up.
This also happens when i use another prefix (another provider, even
PSTN) for the second call too.
The relevant part from asterisk console is at the end of this e-mail, i
don't really understand the warning messages.
- configs:
In sip.conf, the configuration for the two SIP accounts are:
register = 0621380:[EMAIL PROTECTED]
register = 0621381:[EMAIL PROTECTED]
[neophonex]
type=friend
host=sip.neophonex.hu
context=dialout_prefix-context
username=0621380
authname=0621380
fromuser=0621380
secret=password
callerid=0621380
fromdomain=sip.neophonex.hu
disallow=all
allow=alaw
allow=g723
dtmfmode=inband
nat=no
[neophonex-out]
type=friend
host=sip.neophonex.hu
context=dialout_prefix-context
username=0621381
authname=0621381
fromuser=0621381
secret=password
callerid=0621381
fromdomain=sip.neophonex.hu
disallow=all
allow=alaw
allow=g723
dtmfmode=inband
nat=no
extension.conf:
exten = _952.,1,Playback(kapcsolas,noanswer)
exten = _952.,n,Set(CALLERID(name)=0621380)
exten = _952.,n,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
I have tried every possible setting i know about, but still, when i call
outside, via 'turning around' in asterisk, both cells hung up when
answering the call. I have tried calling a regular landline phone
number but still hanging up.
Both accounts are valid, registered and have enough credit to dial
outside its voice network.
The only way the call does not hung up is when i dial extensions within
asterisk.
The asterisk log:
-- Called [EMAIL PROTECTED]
-- Call on SIP/neophonex-out-081a9cc0 left from hold
-- SIP/neophonex-out-081a9cc0 is making progress passing it to
SIP/neophonex-081ab240
[Jun 25 16:57:07] WARNING[18232]: chan_sip.c:11839
handle_response_invite: Re-invite to non-existing call leg on other UA.
SIP dialog '[EMAIL PROTECTED]'. Giving up.
-- Call on SIP/neophonex-out-081a9cc0 left from hold
-- SIP/neophonex-out-081a9cc0 answered SIP/neophonex-081ab240
-- Native bridging SIP/neophonex-081ab240 and
SIP/neophonex-out-081a9cc0
[Jun 25 16:57:10] WARNING[18232]: chan_sip.c:11839
handle_response_invite: Re-invite to non-existing call leg on other UA.
SIP dialog '[EMAIL PROTECTED]'. Giving up.
== Spawn extension (internal, 9520630111, 3) exited non-zero on
'SIP/neophonex-081ab240'
[Jun 25 16:57:10] NOTICE[18440]: pbx_spool.c:351 attempt_thread: Call
completed to SIP/[EMAIL PROTECTED]
Please help me to figure out why the calls are hung up.
Thanks
Adam
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[asterisk-users] ZAP TDM and DTMF issue
Hi, I'm curious if there is any other option beside relaxdtmf in zapata , or any where else to tune dtmf detection on TDM400 fxo boards. in one of our sites provider is giving us 4 analog lines out of Adtran router and Asterisk often recognize DTMF wrong. Obviously playing with relaxdtmf was not helpfull. What do we know anout 1.2 and 1.4 DTMF handling diffrences? At this time i'm using 1.2 but i can change to 1.4 if i see a motivation. _ Local listings, incredible imagery, and driving directions - all in one place! Find it! http://maps.live.com/?wip=69FORM=MGAC01 ___ --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
