Re: [asterisk-users] Inquiry:Asterisk sound files
No . I don't receive any error message after converting from *.wav to *.gsm but the new announcements cannot be heared (when trying for playback). On Tue, Sep 8, 2009 at 6:37 AM, Roel Sarmiento technomage.scratchbu...@gmail.com wrote: Do you have an error message? On Tue, Sep 8, 2009 at 1:29 PM, hadi motamedi motamed...@gmail.comwrote: Dear All Can you please do me favor and let me know why my converted sound files are not being played and heared on my Asterisk ? Please find attached my sound files . Actually , I had them recorded as *.wav files and I tried to convert them to *.gsm as the followings : #sox FR3.wav FR3.gsm Then I put my special announcements under /var/lib/asterisk/sounds but these converted announcement files cannot be heared on my Asterisk . Can you please let me know what is the problem ? Regards H.Motamedi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Inquiry:Asterisk sound files
check the file formats first if .wav is listed there and if it is, then check the translation if its activated. On Tue, Sep 8, 2009 at 2:05 PM, hadi motamedi motamed...@gmail.com wrote: No . I don't receive any error message after converting from *.wav to *.gsm but the new announcements cannot be heared (when trying for playback). On Tue, Sep 8, 2009 at 6:37 AM, Roel Sarmiento technomage.scratchbu...@gmail.com wrote: Do you have an error message? On Tue, Sep 8, 2009 at 1:29 PM, hadi motamedi motamed...@gmail.comwrote: Dear All Can you please do me favor and let me know why my converted sound files are not being played and heared on my Asterisk ? Please find attached my sound files . Actually , I had them recorded as *.wav files and I tried to convert them to *.gsm as the followings : #sox FR3.wav FR3.gsm Then I put my special announcements under /var/lib/asterisk/sounds but these converted announcement files cannot be heared on my Asterisk . Can you please let me know what is the problem ? Regards H.Motamedi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Inquiry:Asterisk sound files
Thank you . Please be informed that the *.wav files cannot be played on my Asterisk so I had to convert to *.gsm file format .I tried to convert to *.gsm by making use of sox but the new announcement cannot be heard . On Tue, Sep 8, 2009 at 7:22 AM, Roel Sarmiento technomage.scratchbu...@gmail.com wrote: check the file formats first if .wav is listed there and if it is, then check the translation if its activated. On Tue, Sep 8, 2009 at 2:05 PM, hadi motamedi motamed...@gmail.comwrote: No . I don't receive any error message after converting from *.wav to *.gsm but the new announcements cannot be heared (when trying for playback). On Tue, Sep 8, 2009 at 6:37 AM, Roel Sarmiento technomage.scratchbu...@gmail.com wrote: Do you have an error message? On Tue, Sep 8, 2009 at 1:29 PM, hadi motamedi motamed...@gmail.comwrote: Dear All Can you please do me favor and let me know why my converted sound files are not being played and heared on my Asterisk ? Please find attached my sound files . Actually , I had them recorded as *.wav files and I tried to convert them to *.gsm as the followings : #sox FR3.wav FR3.gsm Then I put my special announcements under /var/lib/asterisk/sounds but these converted announcement files cannot be heared on my Asterisk . Can you please let me know what is the problem ? Regards H.Motamedi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Using asterisk as the recording server
On Mon, Sep 07, 2009 at 01:47:57PM -0400, Steve Totaro wrote: On Mon, Sep 7, 2009 at 10:09 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Mon, Sep 07, 2009 at 07:44:07AM -0400, Steve Totaro wrote: On Mon, Sep 7, 2009 at 5:58 AM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Mon, Sep 07, 2009 at 01:15:12AM -0400, Steve Totaro wrote: On Mon, Sep 7, 2009 at 1:03 AM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Sun, Sep 06, 2009 at 11:06:39PM -0400, Steve Totaro wrote: On Sun, Sep 6, 2009 at 10:47 PM, Research resea...@businesstz.com wrote: Hello team; While am aware and active user of astersk monitor function for recording, i would like to know if i can use asterisk as a pure recording server(like nice or witness) for some other PABX's extensions (both inbound, outbound and internal). Setup PSTN---Legacy PABX(with analogy n digital extensions)--- asterisk(record Legacy PABX extensions.) Sam Is there any SIP or other VoIP in the mix? If so, you should take a look at OrecX. http://oreka.sourceforge.net (Open Source) They also have a paid version. Another method to do that is to make the Asterisk monitor output dummy SIP calls rather than sound files. Oreka/Orex can listen to those. Looking for volunteers to test that: http://svn.digium.com/svn/asterisk/team/tzafrir/monitor-rtp/ http://svn.digium.com/svn/asterisk/team/tzafrir/monitor-rtp-14/ http://svn.digium.com/svn/asterisk/team/tzafrir/monitor-rtp/configs/monitor.conf.sample This allows recording non-VoIP links, VoIP links where tapping is not convinient, or more selective recording of VoIP calls. Is this similar or the same as the portion of my post that you snipped? Different in many ways, which is why I snipped it. Sangoma RTP Tap will allow you to record TDM calls, again using OrecX but minus the VoIP. (Actually: recorded calls are sent as RTP streams to the Orex/Oreka server) This records outside of Asterisk. Thus it lacks information available in Asterisk (who really called who). OTOH, it is Asterisk-specific. We actually considered implementing something similar to the Sangoma interface in our driver but realised that doing it in Asterisk would probably be more useful. The overheade seems reasonable. Sorry, I fail to see the difference besides Sangoma implemented it in their Wanpipe drivers and you are attempting copy their idea and do it in Asterisk. Your quote This allows recording non-VoIP links, VoIP links where tapping is not convenient (edited to fix your spelling mistake), or more selective recording of VoIP calls. Isn't that more or less the same thing I said that you snipped, Sangoma RTP Tap will allow you to record TDM calls, again using OrecX but minus the VoIP. And what if the call does not go through a TDM card? And ore importantly: how can you tell who is the caller and who is the callee? The rtp-tap interface basically tells you that channel X had a call at time Y. I am sure it is pretty trivial to figure out who channel X and Y are based on the channel, time, CID, DID Just a wee bit of code... Which means you have to keep a separate DB of that (I know such DB exists: the CDR) and get that data from it. Extra work to do. Some people prefer to avoid it. If it does not go through a TDM card, and is VoIP, then port mirroring works just fine. Sipcallid is a very simple way to match callers to callees. VoIP mirroring implies you have control over the network infrastructure. What if you install the PBX in a hostile network where the network administrator doesn't like you sniffing other network traffic? Not to mention that it is extra setup. So we add a different option. One that depends on Asterisk sending the relevant data, and uses the existing monitoring infrastructure in Asterisk: simply use Monitor and StopMonitor to enable/disable monitoring. This is something Asterisk admins should be familiar with. I snip content that is not relevant to my reply. Whoever reads this list already read about the Sangoma interface previously. I had nothing to say about it. It was not related to that new branch. Not everyone who reads the list, reads all the posts, give me a break. It was related to the thread. My target audince in posts to asterisk-users is (surpirse-surpirse) the readers of asterisk-users. I generally do expect them to follow the list[1]. Your motives and alliances have and always will be for Xorcom and Digium. That is the only reason why you helped me with that BRI install in the US, so you could poke around
Re: [asterisk-users] TE420P configuration
/etc/dahdi/system.conf file is auto generated do we need to change in this file as we do for zaptel ? Any working examples ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Shared Call Appearance - Polycom Phones
Does asterisk support doing shared call appearances with polycom 550 and 650 phones? I know that a line can be changed to type shared on the polycom but is it possible to put a call on hold with one phone and resume the held call with another? Does this work by default or is there some special configuration required? --Jared ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Inquiry:Asterisk sound files
I'm not quite sure but i think if you converted the file ex: file.wav using sox it should produce something like file.ulaw, file.alaw, file.gsm. Check if its there, then check the translation if you have the codec activated, it worked for me before. On Tue, Sep 8, 2009 at 2:40 PM, hadi motamedi motamed...@gmail.com wrote: Thank you . Please be informed that the *.wav files cannot be played on my Asterisk so I had to convert to *.gsm file format .I tried to convert to *.gsm by making use of sox but the new announcement cannot be heard . On Tue, Sep 8, 2009 at 7:22 AM, Roel Sarmiento technomage.scratchbu...@gmail.com wrote: check the file formats first if .wav is listed there and if it is, then check the translation if its activated. On Tue, Sep 8, 2009 at 2:05 PM, hadi motamedi motamed...@gmail.comwrote: No . I don't receive any error message after converting from *.wav to *.gsm but the new announcements cannot be heared (when trying for playback). On Tue, Sep 8, 2009 at 6:37 AM, Roel Sarmiento technomage.scratchbu...@gmail.com wrote: Do you have an error message? On Tue, Sep 8, 2009 at 1:29 PM, hadi motamedi motamed...@gmail.comwrote: Dear All Can you please do me favor and let me know why my converted sound files are not being played and heared on my Asterisk ? Please find attached my sound files . Actually , I had them recorded as *.wav files and I tried to convert them to *.gsm as the followings : #sox FR3.wav FR3.gsm Then I put my special announcements under /var/lib/asterisk/sounds but these converted announcement files cannot be heared on my Asterisk . Can you please let me know what is the problem ? Regards H.Motamedi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] features.conf : feature map == getting feature to work
just a hint. you might have # assigned the moh in feature.conf and #3 to starting the recording. check your feature.conf and makesure that # isn't assigned to anything. erik de wild Tripple-o Your Asterisk migration partner the Netherlands Verstuurd vanaf mijn iPhone Op 7 sep 2009 om 20:40 heeft jonas kellens jonas.kell...@telenet.be het volgende geschreven:\ Hi there, I need some help with a 'custom' feature. I have following feature defined in features.conf : [applicationmap] opnemencallee = #3,self/callee,Monitor,wav,/var/samba/profiles/ jonaskl/recording,m In my dialplan : [from-HostAst] exten = s,1,Set(__DYNAMIC_FEATURES=opnemencallee) exten = s,n,Dial(SIP/grandstream,30) I want the callee to be able to press #3 to be able to record the conversation but when I press these keys on my Grandstream phone, the following is displayed on the CLI : [Sep 7 20:33:49] WARNING[10870]: res_musiconhold.c:665 get_mohbyname: Music on Hold class '/var/samba/profiles/jonaskl/ recording' not found Don't know where this comes from... I have tried the same with *3. Same output on the CLI. Yes, I have restarted Asterisk after changes in features.conf. It's not my Grandstream or the DTMF-input because *8 for picking up a ringing phone works well... When I set : opnemencallee = #*3,self/callee,Monitor,wav,/var/samba/profiles/ jonaskl/recording,m and I press #*3, nothing happens... No output on the CLI. There's not much info. I followed the instructions on voip-info.org (which are the same as in features.conf). The module res_features is loaded. Greetingz, Jonas. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] asterisk and link spa942 provisioning
Hellos, I need to send personal directory from asterisk to the ersonal directory of the linksys spa 942. Is this possible? -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke mutuku.wordpress.com Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve better customer satisfaction and sales ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Inquiry:Problem with Call Parking
Dear All I sent you a message regarding my problem with Asterisk Call Parking feature and you told me that needs to check the polycom sip.cfg file . But my Asterisk doesn't have sip.cfg file . Can you please let me know how can I overcome ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Asterisk CLI commands not running !!!!!
Hello, I am using Asterisk 1.4.22 in a debian 4.0 with a kernel version 2.6.18-6-amd64 (SMP). The processor type is Intel(R) Xeon(R) CPU E5420 @ 2.50GHz. Sometimes, I get a strange behavior from asterisk: The CLI commands does not work and Asterisk cannot receive calls. The output of every CLI command is that command is not known (no such command). Please help me resolve this problem: what can be the cause of it? is it Asterisk or my system? and what have I to do to eliminate this problem? Thks in advance. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] features.conf : feature map == getting feature to work
Erik, I have placed everything in features.conf in comment ( ; ). Still when I run show features, I get this : clarkconnect*CLI show features Builtin Feature Default Current --- --- --- Pickup*8 *8 Blind Transfer# # Attended Transfer One Touch Monitor Disconnect Call * * Park Call clarkconnect*CLI Dynamic Feature Default Current --- --- --- opnemencaller no def #* opnemencallee no def #* clarkconnect*CLI Call parking *CLI Parking extension : 700 Parking context : parkedcalls Parked call extensions: 701-750 So there might be indeed a mix-up. It seems to me that the default features, like *8 for pickup, cannot be disabled. Even when in comment they still work ! I have restarted Asterisk after changes in features.conf. Jonas. On Tue, 2009-09-08 at 09:17 +0200, Erik de Wild wrote: just a hint. you might have # assigned the moh in feature.conf and #3 to starting the recording. check your feature.conf and makesure that # isn't assigned to anything. erik de wild Tripple-o Your Asterisk migration partner the Netherlands Verstuurd vanaf mijn iPhone ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Inquiry:Asterisk sound files
Thank you . Please find below my original and converted sound files attributes on my Asterisk : #file FR1.wav FR1.wav: RIFF (little-endian) data, WAVE audio, ITU G.711 A-law, mono 8000 Hz #file FR1.gsm FR1.gsm: data Can you please let me know what is the problem as the sox does not generate error message when converting ? On Tue, Sep 8, 2009 at 7:57 AM, Roel Sarmiento technomage.scratchbu...@gmail.com wrote: I'm not quite sure but i think if you converted the file ex: file.wav using sox it should produce something like file.ulaw, file.alaw, file.gsm. Check if its there, then check the translation if you have the codec activated, it worked for me before. On Tue, Sep 8, 2009 at 2:40 PM, hadi motamedi motamed...@gmail.comwrote: Thank you . Please be informed that the *.wav files cannot be played on my Asterisk so I had to convert to *.gsm file format .I tried to convert to *.gsm by making use of sox but the new announcement cannot be heard . On Tue, Sep 8, 2009 at 7:22 AM, Roel Sarmiento technomage.scratchbu...@gmail.com wrote: check the file formats first if .wav is listed there and if it is, then check the translation if its activated. On Tue, Sep 8, 2009 at 2:05 PM, hadi motamedi motamed...@gmail.comwrote: No . I don't receive any error message after converting from *.wav to *.gsm but the new announcements cannot be heared (when trying for playback). On Tue, Sep 8, 2009 at 6:37 AM, Roel Sarmiento technomage.scratchbu...@gmail.com wrote: Do you have an error message? On Tue, Sep 8, 2009 at 1:29 PM, hadi motamedi motamed...@gmail.comwrote: Dear All Can you please do me favor and let me know why my converted sound files are not being played and heared on my Asterisk ? Please find attached my sound files . Actually , I had them recorded as *.wav files and I tried to convert them to *.gsm as the followings : #sox FR3.wav FR3.gsm Then I put my special announcements under /var/lib/asterisk/sounds but these converted announcement files cannot be heared on my Asterisk . Can you please let me know what is the problem ? Regards H.Motamedi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Inquiry:Asterisk sound files
is there an error on the asterisk cli when you're playing the sound file? On Tue, Sep 8, 2009 at 4:19 PM, hadi motamedi motamed...@gmail.com wrote: Thank you . Please find below my original and converted sound files attributes on my Asterisk : #file FR1.wav FR1.wav: RIFF (little-endian) data, WAVE audio, ITU G.711 A-law, mono 8000 Hz #file FR1.gsm FR1.gsm: data Can you please let me know what is the problem as the sox does not generate error message when converting ? On Tue, Sep 8, 2009 at 7:57 AM, Roel Sarmiento technomage.scratchbu...@gmail.com wrote: I'm not quite sure but i think if you converted the file ex: file.wav using sox it should produce something like file.ulaw, file.alaw, file.gsm. Check if its there, then check the translation if you have the codec activated, it worked for me before. On Tue, Sep 8, 2009 at 2:40 PM, hadi motamedi motamed...@gmail.comwrote: Thank you . Please be informed that the *.wav files cannot be played on my Asterisk so I had to convert to *.gsm file format .I tried to convert to *.gsm by making use of sox but the new announcement cannot be heard . On Tue, Sep 8, 2009 at 7:22 AM, Roel Sarmiento technomage.scratchbu...@gmail.com wrote: check the file formats first if .wav is listed there and if it is, then check the translation if its activated. On Tue, Sep 8, 2009 at 2:05 PM, hadi motamedi motamed...@gmail.comwrote: No . I don't receive any error message after converting from *.wav to *.gsm but the new announcements cannot be heared (when trying for playback). On Tue, Sep 8, 2009 at 6:37 AM, Roel Sarmiento technomage.scratchbu...@gmail.com wrote: Do you have an error message? On Tue, Sep 8, 2009 at 1:29 PM, hadi motamedi motamed...@gmail.com wrote: Dear All Can you please do me favor and let me know why my converted sound files are not being played and heared on my Asterisk ? Please find attached my sound files . Actually , I had them recorded as *.wav files and I tried to convert them to *.gsm as the followings : #sox FR3.wav FR3.gsm Then I put my special announcements under /var/lib/asterisk/sounds but these converted announcement files cannot be heared on my Asterisk . Can you please let me know what is the problem ? Regards H.Motamedi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit:
Re: [asterisk-users] invalid extension
you should check dialstatus and gotoif. if you use both in the proper way ( see the wiki) then you have the dialplan behaviour you are looking for. erik de wild Tripple-o Your Asterisk migration partner the Netherlands Verstuurd vanaf mijn iPhone Op 7 sep 2009 om 21:26 heeft Miguel Molina mmol...@millenium.com.co het volgende geschreven:\ Administrator TOOTAI escribió: Hello, with Asterisk 1.6.1.6 I try to hangup a call if called extension is not existing. For this purpose I would use the internal i extension but seems not to work. [MyContext] exten = s,1,NoOp(Call is treated as it should) exten = s,n,NoOp(next step) exten = s,n,NoOp(aso ...) exten = _[a-zA-Z].,1,Goto(s,1); accept exten LEN 1 alpha exten = _X.,1,Goto(s,1); accept exten LEN 1 numeric exten = i,1,NoOP(sorry, extension doesnt exist) ; all 1 digits exten exten = i,n,Hangup ; refused, end of call What I have when calling a one digit extension -in this case h- is: == Using SIP RTP CoS mark 5 [Sep 7 18:51:03] NOTICE[6084]: chan_sip.c:18523 handle_request_invite: Call from '' to extension 'h' rejected because extension not found. == Using SIP RTP CoS mark 5 Should it not go to i extension? If I call the i or s extension it's going well. Am I missing something? Hi, The 'i' extension only works in applications like Background(), WaitExten() and everything that uses DTMF to route extensions within a context. As you can see in your call, it won't work directly because asterisk by default will reject a call that doesn't match in the context or included contexts you defined for the user. Because the call is not accepted there's no need for a hangup (in a SIP environment). If you want to explicitly hangup calls using the dialplan, for your case add a one-digit catch all and leave your good calls with a 2-digit minimum: exten = _X,1,NoOP(sorry, extension doesnt exist) ; all 1 digits exten exten = _X,n,Hangup exten = _XX.,1,Goto(s,1); accept exten LEN 1 numeric That will be enough to hangup what you want to, adjusting it to your needs. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Using asterisk as the recording server
using mixmonitor might not be such a good idea. afaik the mixing of the recordings of the two channels starts after ending the call causing a high cpu load. if you have recordings going on all the time moving the 2 files that has to be mixed to a dedicated mixing server might be a good idea. after mixing it should be stored in a retrievable way. Erik de Wild Tripple-o Your Asterisk migration partner the Netherlands Verstuurd vanaf mijn iPhone Op 8 sep 2009 om 00:25 heeft Miguel Molina mmol...@millenium.com.co het volgende geschreven:\ I imagine this setup will need those two communicating entities to be part of the pabx. But support extension 100 of PABX A (legacy) calls 101 on the same platform. I want asterisk connected to PABX A via E1/T1 to know about that call and start recording (tap) without bridging or being part of that conversation Hi, Asterisk won't work as a recording server if the call doesn't go through it. In the IP world it means that both media (RTP) and signalling must pass through asterisk, and in the E1/T1 digital or analog world it means that the call must be bridged through asterisk. A simple dialplan would explain it: exten = s,1,Answer() ;Asterisk receives the call, from the lecagy PBX or from the external link (this should be two different contexts) exten = s,n,MixMonitor(blah) ; Records the conversation, exten = s,n,Dial(Tech/peer/number...,30,rtTwhatever) ; and sends the call back to the legacy PBX or to an external link If you want to record 100% calls, you would have to route every call through asterisk, even internal PBX calls. Even if you want to tap your legacy PBX to a non-asterisk recording server like the ones suggested before in this thread, the calls must go through a link to make tapping possible and you should seek an alternate solution to the internal calls within your legacy PBX. The beauty of asterisk and open source IP-PBXs relies on the native recording capabilities which makes things really easy. When you see that asterisk works and that can do the recordings and much more, you would start thinking on making asterisk your main PBX solution and leaving that legacy PBX for minimal uses. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] features.conf : feature map == getting feature to work
8 sep 2009 kl. 10.17 skrev jonas kellens: Erik, I have placed everything in features.conf in comment ( ; ). Still when I run show features, I get this : clarkconnect*CLI show features Builtin Feature Default Current --- --- --- Pickup*8 *8 Blind Transfer# # Attended Transfer One Touch Monitor Disconnect Call * * Park Call clarkconnect*CLI Dynamic Feature Default Current --- --- --- opnemencaller no def #* opnemencallee no def #* clarkconnect*CLI Call parking *CLI Parking extension :700 Parking context :parkedcalls Parked call extensions: 701-750 So there might be indeed a mix-up. It seems to me that the default features, like *8 for pickup, cannot be disabled. Even when in comment they still work ! I have restarted Asterisk after changes in features.conf. The *8 is indeed a hard-coded feature in many channels, but only works if you've enabled callgroups and pickupgroups in the channel configurations, and it's a dialstring, not a DTMF code. The blind transfer functionality is enabled by the 'tT' options to dial(), like the disconnect call feature. The dynamic features is something you've enabled in features.conf. They require dialplan intervention to work. Call parking are again extensions, not DTMF codes. To work, requires you to include the context in the dialplan for a device/line. So apart from the dynamic features, what you see here is the Asterisk defaults that can be changed in features.conf, but will always be there. They only work if you enable them in other files. /O ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] CallerID app for Symbian?
Hi, we're using a GSM-Gateway on asterisk to forward incoming calls to the cellphones, but, of course, the cellphones always display the callerid from the gateway. Does anyone know a symbian app that could (on an incoming call) connect via grps/3G to a database behind the asterisk and fetch the real callerid and do a calleridname-lookup on a number? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] features.conf : feature map == getting feature to work
When I enable the automon-feature (*1) the callee can start recording the conversation. No problem there. But I can't get my user-defined features to work. I have setup the following test-feature in features.conf : [applicationmap] testfeat = *3,self/callee,Playback,tt-weasels I have the following in my dialplan : [from-HostAst] exten = s,1,Set(__DYNAMIC_FEATURES=testfeat) exten = s,n,NoOp(...) exten = s,n,NoOp(...) exten = s,n,Dial(SIP/grandstream,30) When pressing *3 on the Grandstream (the callee), the CLI shows nothing : [Sep 8 11:22:14] -- Executing [...@from-hostast:1] Set(IAX2/hostedasterisk-12746, __DYNAMIC_FEATURES=testfeat) in new stack [Sep 8 11:22:14] -- Executing [...@from-hostast:2] NoOp(IAX2/hostedasterisk-12746, ...) in new stack [Sep 8 11:22:14] -- Executing [...@from-hostast:3] NoOp(IAX2/hostedasterisk-12746, ...) in new stack [Sep 8 11:22:14] -- Executing [...@from-hostast:4] Dial(IAX2/hostedasterisk-12746, SIP/grandstream|30) in new stack [Sep 8 11:22:14] -- Called grandstream [Sep 8 11:22:14] -- SIP/grandstream-083d5c10 is ringing [Sep 8 11:22:22] -- SIP/grandstream-083d5c10 answered IAX2/hostedasterisk-12746 ... nothing happens when pressing *3... [Sep 8 11:22:52] == Spawn extension (from-HostAst, s, 4) exited non-zero on 'IAX2/hostedasterisk-12746' [Sep 8 11:22:52] -- Hungup 'IAX2/hostedasterisk-12746' With the automon-feature, it works well : [Sep 8 11:18:35] -- User hit '*1' to record call. filename: wav| auto-1252401515-s-IAX2-hostedasterisk-9817|m [Sep 8 11:18:46] -- User hit '*1' to stop recording call. What am I doing wrong so that my user-defined features don't work ? Even this simple Playback(tt-weasels) won't work. Jonas. On Mon, 2009-09-07 at 16:03 -0500, Anthony Messina wrote: On Monday 07 September 2009 13:40:16 jonas kellens wrote: [applicationmap] opnemencallee = #3,self/callee,Monitor,wav,/var/samba/profiles/jonaskl/recording,m FeatureName = DTMF_sequence,ActivateOn[/ActivatedBy],Application[,AppArguments[,MOH_Class]] it looks like /var/samba/profiles/jonaskl/recording is in the spot for [,MOH_Class] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Inquiry:Asterisk sound files
Please find below the error message that I am receiving on my Asterisk : -- Executing [s-noans...@macro-dialuser:4] Playback(Zap/95-1, FR1) in new stack [Sep 7 11:11:34] WARNING[7624]: format_wav.c:140 check_header: Not a wav file 6 [Sep 7 11:11:34] WARNING[7624]: file.c:316 fn_wrapper: Unable to open format wav WARNING[7624]: file.c:866 ast_streamfile: Unable to open FR1 (format 0x48 (alaw|slin)): WARNING[7624]: app_playback.c:437 playback_exec: ast_streamfile failed on Zap/95-1 for FR1 On Tue, Sep 8, 2009 at 9:34 AM, Roel Sarmiento technomage.scratchbu...@gmail.com wrote: is there an error on the asterisk cli when you're playing the sound file? On Tue, Sep 8, 2009 at 4:19 PM, hadi motamedi motamed...@gmail.comwrote: Thank you . Please find below my original and converted sound files attributes on my Asterisk : #file FR1.wav FR1.wav: RIFF (little-endian) data, WAVE audio, ITU G.711 A-law, mono 8000 Hz #file FR1.gsm FR1.gsm: data Can you please let me know what is the problem as the sox does not generate error message when converting ? On Tue, Sep 8, 2009 at 7:57 AM, Roel Sarmiento technomage.scratchbu...@gmail.com wrote: I'm not quite sure but i think if you converted the file ex: file.wav using sox it should produce something like file.ulaw, file.alaw, file.gsm. Check if its there, then check the translation if you have the codec activated, it worked for me before. On Tue, Sep 8, 2009 at 2:40 PM, hadi motamedi motamed...@gmail.comwrote: Thank you . Please be informed that the *.wav files cannot be played on my Asterisk so I had to convert to *.gsm file format .I tried to convert to *.gsm by making use of sox but the new announcement cannot be heard . On Tue, Sep 8, 2009 at 7:22 AM, Roel Sarmiento technomage.scratchbu...@gmail.com wrote: check the file formats first if .wav is listed there and if it is, then check the translation if its activated. On Tue, Sep 8, 2009 at 2:05 PM, hadi motamedi motamed...@gmail.comwrote: No . I don't receive any error message after converting from *.wav to *.gsm but the new announcements cannot be heared (when trying for playback). On Tue, Sep 8, 2009 at 6:37 AM, Roel Sarmiento technomage.scratchbu...@gmail.com wrote: Do you have an error message? On Tue, Sep 8, 2009 at 1:29 PM, hadi motamedi motamed...@gmail.com wrote: Dear All Can you please do me favor and let me know why my converted sound files are not being played and heared on my Asterisk ? Please find attached my sound files . Actually , I had them recorded as *.wav files and I tried to convert them to *.gsm as the followings : #sox FR3.wav FR3.gsm Then I put my special announcements under /var/lib/asterisk/sounds but these converted announcement files cannot be heared on my Asterisk . Can you please let me know what is the problem ? Regards H.Motamedi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Asterisk CLI commands not running !!!!!
Just a hint based on experience. Run top from de linux prompt to check if any proces causes an enormous cpu load. I once ran into the same behaviour because some asterisk related php script looped and took almost all the cpu power available. erik de wild Tripple-o Your Asterisk migration partner the Netherlands Verstuurd vanaf mijn iPhone Op 8 sep 2009 om 09:39 heeft abdelkader abdelkader2...@gmail.com het volgende geschreven:\ Hello, I am using Asterisk 1.4.22 in a debian 4.0 with a kernel version 2.6.18-6-amd64 (SMP). The processor type is Intel(R) Xeon(R) CPU E5420 @ 2.50GHz. Sometimes, I get a strange behavior from asterisk: The CLI commands does not work and Asterisk cannot receive calls. The output of every CLI command is that command is not known (no such command). Please help me resolve this problem: what can be the cause of it? is it Asterisk or my system? and what have I to do to eliminate this problem? Thks in advance. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] CallerID app for Symbian?
On 8 Sep 2009, at 10:22, Jay R. Worthington wrote: we're using a GSM-Gateway on asterisk to forward incoming calls to the cellphones, but, of course, the cellphones always display the callerid from the gateway. Does anyone know a symbian app that could (on an incoming call) connect via grps/3G to a database behind the asterisk and fetch the real callerid and do a calleridname-lookup on a number? Why not just spoof the caller ID? Allowed in most countries as in this case it isn't misleading information. S ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Using asterisk as the recording server
Again, how many calls were you able record using RAMdisk? Anywhere 300? As I stated before, this is going to be dependent on how you're manipulating the calls and the gear you're running on. The nice thing about your 'just broadcast the entire LAN to the recording solution' is that the recording service just gets to throw away everything that's not an audio channel, and it doesn't have to do squat to the call. If it COULDN'T do a lot of recordings under these circumstances it wouldn't be worth any money. I don't think I've pushed my solution past 90 simultaneous recordings of MeetMe() mixing, with more than 100 AGI channels running, with assorted ChanSpy() jobs. Bookmark my post, so when you reach your RAMDisk limit, you can join the big league. Anything I do as a scaling solution will be price versus performance. So since we're talking about a commercial solution to replace something that asterisk does, I'll have to find out what your commercial solution costs per channel, and compare that against the cost of cloning out an identical server. My solution scales to parallel servers just fine. Is OrecX really $199 per recorded channel? So that 300 channels you're talking about costs $60,000? So I can buy six $10,000 servers, each of which can run circles around my current solution, and still break even. I like my solution better. OrecX has a free version. I guess you didin't really check it out since your mind was already made up. 300+ Simultaneous calls recorded in perfect clarity for the price of an R200 or if you want higher end, a DL360 Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Intermittent metallic voice SIP-ISDN ISDN-SIP
Hi all, I'm fighting with a really strange problem that is really busting me. I have an asterisk 1.4.22 ( from a trixbox 2.6.2 ) and mISDN 1.1.7 3 extension on hardphone and 3 extension in softphone ( zoiper ) What happens is that sometimes the people on the other side of communication hear my voice as metallic and chopped. This happen either on incoming call than on outgoing call. If I keep the call up ( asking the other part to wait ) for a minute or less, then the voice get better and we can continue the call. On the other hand, if I hangup the line and call again, everithing is fine. For what I have read, it seems that the problem get triggered by some jitter, also if I can't understand why, being my asterisk in a local lan switched 100mbit. I have searched through a lot of messages and info, and tried a lot of suggested solutions, but can't get the right fix to this. Do you have any hint ? Thx Pigi The isdn is connected with an HFC-PCI card: 03:00.0 Network controller: Cologne Chip Designs GmbH ISDN network controller [HFC-PCI] (rev 02) this is my sip general part (jb enable to get the jitter buffer working): jbenable = yes jbforce = yes jbmaxsize = 200 jbresyncthreshold = 1000 jbimpl = adaptive jblog = yes This is the relevant part of the misdn-init.conf card=1,hfcpci te_ptp=1,2 poll=128 dsp_poll=128 dsp_options=0 dtmfthreshold=100 debug=0 And this is the misdn.conf [general] misdn_init=/etc/misdn-init.conf debug=0 ntdebugflags=0 ntdebugfile=/var/log/misdn-nt.log ntkeepcalls=no bridging=no l1watcher_timeout=0 stop_tone_after_first_digit=yes append_digits2exten=yes dynamic_crypt=no crypt_prefix=** crypt_keys=test,muh [default] context=misdn language=en musicclass=default senddtmf=yes far_alerting=no allowed_bearers=speech,3_1khz nationalprefix= internationalprefix=0 rxgain=0 txgain=0 te_choose_channel=no pmp_l1_check=no reject_cause=16 need_more_infos=no nttimeout=no method=standard overlapdial=yes dialplan=0 localdialplan=0 cpndialplan=0 early_bconnect=yes incoming_early_audio=no nodialtone=no presentation=-1 screen=-1 echotraining=no jitterbuffer=4000 jitterbuffer_upper_threshold=0 hdlc=no faxdetect=both faxdetect_timeout=5 max_incoming=-1 max_outgoing=-1 [isdn] ports=1,2 context=from-pstn msns=* ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] confBridge in Asterisk 1.6.2.0-rc1 doesn't stable
- Ian Wang iyu.w...@gmail.com wrote: confBridge in Asterisk 1.6.2.0-rc1 doesn't stable. It causes segment fault very often and results in asterisk crash. It would be extremely useful if you could file an issue on https://issues.asterisk.org/ with details about how you are using it including console output, number of channels in it, and as well a backtrace. Instructions for getting a backtrace are available in the doc directory under the backtrace.txt file. -- Joshua Colp Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Asterisk CLI commands not running !!!!!
Asterisk sometimes goes to sleep. (And never wakes-up). Restart it and all will be fine again. We have a watchdog which sends SIP OPTIONS packet to Asterisk and if it does not respond – restarts it. Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of abdelkader Sent: 2009 m. rugsėjo 8 d. 10:40 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk CLI commands not running ! Hello, I am using Asterisk 1.4.22 in a debian 4.0 with a kernel version 2.6.18-6-amd64 (SMP). The processor type is Intel(R) Xeon(R) CPU E5420 @ 2.50GHz. Sometimes, I get a strange behavior from asterisk: The CLI commands does not work and Asterisk cannot receive calls. The output of every CLI command is that command is not known (no such command). Please help me resolve this problem: what can be the cause of it? is it Asterisk or my system? and what have I to do to eliminate this problem? Thks in advance. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Help setting IAX variables.
On Tuesday 08 September 2009 00:14:53 Asterisk User wrote: Thanks Tilghman for your quick reply. I know that we should set variables through IAXVAR on source server to access them on Destination server. I just wanted to know the reverse case, where IAX channel variables set on destination server are accessible on Source server or not. Thanks again for your inputs. They are not. IAXVARs are only sent during the NEW, which is a one-way packet. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Using asterisk as the recording server
On Tue, Sep 8, 2009 at 5:08 AM, Erik de Wildi...@meetmecall.nl wrote: using mixmonitor might not be such a good idea. afaik the mixing of the recordings of the two channels starts after ending the call causing a high cpu load. Incorrect. The 'mix' in Mixmonitor() is that two legs of a call are mixed together as the recording happens. It is Monitor() that works in the way you describe, although 'high cpu load' is dependent on how you mix. It would be 'bursty' cpu load if you choose to mix as soon as a recording completes. Since Mixmonitor() is mixing as the recording happens, there is instantly a retrievable recording when the recording completes. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Inquiry:Problem with Call Parking
On Tue, 8 Sep 2009, hadi motamedi wrote: I sent you a message regarding my problem with Asterisk Call Parking feature and you told me that needs to check the polycom sip.cfg file . But my Asterisk doesn't have sip.cfg file . Can you please let me know how can I overcome ? sip.cfg is not an Asterisk file. sip.cfg should be in the directory the phone downloads it's configuration from. Typically, /tftpboot/ on a tftp server. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] E1 line simulation for Asterisk
I just have a T1 TE121 Card, if you want I can send you my file. What kind of card is the TE420P, I think the card is for 4 T1/E1 card, Am I right? De: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] En nombre de ABBAS SHAKEEL Enviado el: Lunes, 07 de Septiembre de 2009 11:33 p.m. Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] E1 line simulation for Asterisk Hello I have the loop back connector and TE420P card but i dont know how to configure that. Please let me know of any help. I am facing the problem in configuration of channels. i have make changes in chan_dahdi [r...@te420 etc]# dahdi_hardware pci::04:08.0 wct4xxp+ d161:0420 Wildcard TE420 (4th Gen) shows this. this means card is configured. Now i have to do configuration in chan_dahdi.conf or some other files . Please some one shed some light on it. I have asked this question in a different topic as well -- Best Regards Shakeel Abbas Teleperformance values: Integrity - Respect - Professionalism - Innovation - Commitment The information contained in this communication is privileged and confidential. The content is intended only for the use of the individual or entity named above. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify me immediately by telephone or e-mail, and delete this message from your systems. Please consider the environmental impact of needlessly printing this e-mail.___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Strange extension state changes in 1.6.0.15
8 sep 2009 kl. 15.40 skrev Benny Amorsen: I see a lot of these on an otherwise idle Asterisk 1.6.0.15: Extension Changed 773[Hints] new state Ringing for Notify User 792-00041327d17e-1. Then a little while later it changes to InUse or Idle, completely randomly. It happens for many different combinations of phones and watchers. There are no calls being made, so I can think of no reason why this happens. Obviously it wreaks havoc with BLF... It doesn't happen in 1.6.0.13. I have tried it on two different production Asterisks, and both of them exhibit this behaviour. Unfortunately I can't reproduce in my smaller test environment. Have any of you seen this? THere was a mail about someone who suddenly had all extensions marked as on hold... Open an issue in the tracker and upload as much information as possible. THanks. /O ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Asterisk remote calls with low bandwith and high latency
Hello, I have 2 sites. One(Site 1) has an asterisk PBx and the Other(site 2) has 2 remote soft phones. The latency btw both sites is btw 500ms-700ms. I know this is a shot in the dark...but are there ways of improving the voice quality for the remote calls(btw site 1 and site 2), Other than increasing bandwidth? -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke mutuku.wordpress.com Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve better customer satisfaction and sales ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Strange extension state changes in 1.6.0.15
I see a lot of these on an otherwise idle Asterisk 1.6.0.15: Extension Changed 773[Hints] new state Ringing for Notify User 792-00041327d17e-1. Then a little while later it changes to InUse or Idle, completely randomly. It happens for many different combinations of phones and watchers. There are no calls being made, so I can think of no reason why this happens. Obviously it wreaks havoc with BLF... It doesn't happen in 1.6.0.13. I have tried it on two different production Asterisks, and both of them exhibit this behaviour. Unfortunately I can't reproduce in my smaller test environment. Have any of you seen this? /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Using asterisk as the recording server
I imagine this setup will need those two communicating entities to be part of the pabx. But support extension 100 of PABX A (legacy) calls 101 on the same platform. I want asterisk connected to PABX A via E1/T1 to know about that call and start recording (tap) without bridging or being part of that conversation Hi, Asterisk won't work as a recording server if the call doesn't go through it. In the IP world it means that both media (RTP) and signalling must pass through asterisk, and in the E1/T1 digital or analog world it means that the call must be bridged through asterisk. A simple dialplan would explain it: exten = s,1,Answer() ;Asterisk receives the call, from the lecagy PBX or from the external link (this should be two different contexts) exten = s,n,MixMonitor(blah) ; Records the conversation, exten = s,n,Dial(Tech/peer/number...,30,rtTwhatever) ; and sends the call back to the legacy PBX or to an external link If you want to record 100% calls, you would have to route every call through asterisk, even internal PBX calls. Even if you want to tap your legacy PBX to a non-asterisk recording server like the ones suggested before in this thread, the calls must go through a link to make tapping possible and you should seek an alternate solution to the internal calls within your legacy PBX. The beauty of asterisk and open source IP-PBXs relies on the native recording capabilities which makes things really easy. When you see that asterisk works and that can do the recordings and much more, you would start thinking on making asterisk your main PBX solution and leaving that legacy PBX for minimal uses. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center Erik de Wild escribió: using mixmonitor might not be such a good idea. afaik the mixing of the recordings of the two channels starts after ending the call causing a high cpu load. if you have recordings going on all the time moving the 2 files that has to be mixed to a dedicated mixing server might be a good idea. after mixing it should be stored in a retrievable way. No, that was the old behavior of Monitor() with the m option that at the end of the recording it launched an underneath sox process which did the mix, causing a CPU spike on every conversation end and putting asterisk on trouble if there were many mixes at the same time. Mixmonitor took care of that, and it does the mixing while the conversation is taking place, thus generating the single file with no CPU spikes or external process calls. Your idea about the separate mixing server was what our company did about three years ago with the old first 1.2 asterisk versions, where MixMonitor used to be buggy and we were forced to implement that kind of solution. But times are a lot better now! Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Manage a E1 system
Hello Every one! I am little bit new to asterisk. I am doing research on different telecom options as well. I have question for you professionals In order to get E1 line working with Asterisk. What E1 line parameters need to be specified in Asterisk(configuration files). They vary from country to country. What is difference in one countries E1 and others E1. the most important If we want to ask a company for an E1 what information we must obtain from provider company other then the E1 line(some ie googled is signaling etc) can i get complere list :) Cheers Adam ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Fwd: Patton smartnode 463x (BRI) 25ms tail echo cancellation
Hi Is anybody using these ? Gaetan Begin forwarded message: From: Gaëtan Minet gminet...@mcit.be Date: Sat 22 Aug 2009 16:29:42 GMT+02:00 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] Patton smartnode 463x (BRI) 25ms tail echo cancellation Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Hi all We use pci/pci-e BRI cards in our installations. Due to echo problems (that was before Oslec and others), we quickly switched to cards with hardware-based EC. So we use exclusively Digium B410p cards that provide 64ms tail EC. For several reasons we'd like to switch to external BRI gateways like the Patton smartnodes (the price is getting really close to a B410p). I'm however curious about their HW EC. I see in the datasheets that it only has 25ms tail per channel (pri are 128ms, but not BRI). Are some of you using these gateway and do your experience (many) echo problems on calls ? Our other alternative is to use sangoma cards that have 128ms HW EC and seem more stable overall, but it is yet a bit more expensive. Thanks for your feedback. Regards, Gaetan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] hang up problem while calling
Hi everyone,
I have a problem at my Trixbox that is version Asterisk 1.2.26.1 svn
rev 79171 and 2.6.9-34.0.2.ELsmp kernel version. Two Digium 4fxs+4fxo
card has been installed and everything was working before made yum
update and at this server. (Centos 4.0). After update I faced with
zaptel not loading problems. I have solved these problems too but now
when I try to call with linksys spa942 or with x-lite like 9
+3549 (with 9 I could reach to pstn and with 3549 I could talk with an
extension number in company) my call is hanging up and waiting without
doing anything.. It behaves like can make call but after establish it is
waiting on line...
Did anyone face with this problem or do you have any suggestion ?
Thanks in advance
2009-09-08 12:06:22 DEBUG[7354] chan_sip.c: Checking SIP call limits for
device 5001
2009-09-08 12:06:22 DEBUG[7354] chan_sip.c: build_route: Contact hop:
sip:5...@10.10.1.109:51406
2009-09-08 12:06:22 DEBUG[19427] pbx.c: Function result is '5001'
2009-09-08 12:06:22 DEBUG[19427] db.c: Unable to find key '5001' in
family 'cidname'
2009-09-08 12:06:22 DEBUG[19427] pbx.c: Function result is '5001'
2009-09-08 12:06:22 DEBUG[19427] pbx.c: Function result is '5001'
2009-09-08 12:06:22 DEBUG[19427] pbx.c: Function result is '5001'
2009-09-08 12:06:22 DEBUG[19427] pbx.c: Function result is '5001'
2009-09-08 12:06:22 DEBUG[19427] pbx.c: Function result is '5001'
2009-09-08 12:06:22 DEBUG[19427] pbx.c: Function result is '5001'
2009-09-08 12:06:22 DEBUG[19427] pbx.c: Function result is 'Yavuzhan
Canli 5001'
2009-09-08 12:06:22 DEBUG[19427] pbx.c: Function result is '5001'
2009-09-08 12:06:22 DEBUG[19427] db.c: Unable to find key
'5001/emergency_cid' in family 'DEVICE'
2009-09-08 12:06:22 DEBUG[19427] func_db.c: DB:
DEVICE/5001/emergency_cid not found in database.
2009-09-08 12:06:22 DEBUG[19427] pbx.c: Function result is 'Yavuzhan
Canli 5001'
2009-09-08 12:06:40 DEBUG[19427] channel.c: Didn't get a frame from
channel: SIP/5001-b7904218
2009-09-08 12:06:40 DEBUG[19427] channel.c: Bridge stops bridging
channels SIP/5001-b7904218 and Zap/7-1
2009-09-08 12:06:40 DEBUG[19427] cdr_addon_mysql.c: cdr_mysql: SQL
command as follows: INSERT INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid)
VALUES ('2009-09-08 12:06:22','\ycanli\
5001','5001','93549','from-internal',
'SIP/5001-b7904218','Zap/7-1','Dial','ZAP/g0/3549|300|',18,14,'ANSWERED',3,'','1252400782.208')
2009-09-08 12:06:40 NOTICE[19427] cdr.c: CDR on channel
'SIP/5001-b7904218' not posted
2009-09-08 12:06:40 NOTICE[19427] cdr.c: CDR on channel
'SIP/5001-b7904218' lacks end
2009-09-08 12:06:40 DEBUG[19427] chan_sip.c: update_call_counter(5001) -
decrement call limit counter
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Re: [asterisk-users] E1 line simulation for Asterisk
Thanks Juan! Yeah you are exactly right. Please send me your file. thanks On Tue, Sep 8, 2009 at 7:40 PM, Juan Cardoza jcard...@tpmex.com wrote: I just have a T1 TE121 Card, if you want I can send you my file. What kind of card is the TE420P, I think the card is for 4 T1/E1 card, Am I right? *De:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *En nombre de *ABBAS SHAKEEL *Enviado el:* Lunes, 07 de Septiembre de 2009 11:33 p.m. *Para:* Asterisk Users Mailing List - Non-Commercial Discussion *Asunto:* Re: [asterisk-users] E1 line simulation for Asterisk Hello I have the loop back connector and TE420P card but i dont know how to configure that. Please let me know of any help. I am facing the problem in configuration of channels. i have make changes in chan_dahdi [r...@te420 etc]# dahdi_hardware pci::04:08.0 wct4xxp+ d161:0420 Wildcard TE420 (4th Gen) shows this. this means card is configured. Now i have to do configuration in chan_dahdi.conf or some other files . Please some one shed some light on it. I have asked this question in a different topic as well -- Best Regards Shakeel Abbas Teleperformance values: * Integrity* - *Respect* - *Professionalism* - * Innovation* – *Commitment * The information contained in this communication is privileged and confidential. The content is intended only for the use of the individual or entity named above. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify me immediately by telephone or e-mail, and delete this message from your systems. * Please consider the environmental impact of needlessly printing this e-mail. * ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] 1.2 AGI Deadlock
Every time I upgrade, I run into more issues than I previously had, so I tend to stay where I am unless I absolutely have to upgrade. I ran 1.0.3 for 2+ years with no issues. I upgraded to whatever the latest 1.2 was at the time and it crashed three times within a week. 1.2.32 and .34 seem to work fine, so I am staying on them until I absolutely have to move. Yes, I get that message with any AGI: *CLI -- Executing AGI(SIP/3211-1-081c51e0, agi-test.agi) in new stack Sep 8 11:48:43 WARNING[564]: channel.c:780 channel_find_locked: Avoided initial deadlock for '0x818edf8', 9 retries! -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-test.agi I am wondering if anybody else is seeing these same issues with the latest 1.2 line, or if it is something specific to my install. Although I can't imagine it is just me as I don't have anything out of the ordinary on that box, just two phones and a basic extensions.conf to call an AGI. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Tuesday, September 08, 2009 11:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 1.2 AGI Deadlock Peder wrote: I am running 1.2.34 (also tried on 1.2.32) and whenever I launch an AGI, I get the avoided deadlock message below. On Tue, 8 Sep 2009, Alex Balashov wrote: A deadlock? In 1.2? Really? :) Well, that was helpful. As a fellow 1.2 Luddite, I have boxes running xxx simultaneous channels, all running xx AGIs per call with no problems. First off, unless you have good reasons, you should move to a newer version just to improve your chances of getting meaningful support on this list. Do you get this deadlock message when you launch any AGI, for example, agi-test.agi? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] dahdi/DTMF problem
On Mon, Sep 7, 2009 at 7:50 PM, Greg Woods g...@gregandeva.net wrote: On Mon, 2009-09-07 at 07:20 -0600, Greg Woods wrote: incoming calls through the FXO line are dropped as soon as there is a button press. The error logged is: [Aug 23 18:15:39] WARNING[6532] chan_dahdi.c: Cannot handle frames in 2 format [Aug 23 18:15:39] WARNING[6532] file.c: Failed to write frame Which looks like this bug: https://issues.asterisk.org/view.php?id=15129 I didn't solve this, but I worked around it. I eventually gave up and installed the asterisk14 1.4.26 packages from ATrpms. This version I was able to get working with Dahdi. I'll keep my eye on the bug report to see if this ever gets fixed, then I might try to upgrade to 1.6. But I have no urgent need to do so, so I am happy to wait a while and at least I can finally retire the old system. --Greg I hope that you'll add yourself as a watcher or comment on the issue so that once somebody gets around to looking at it, you'll be notified and can assist. Jeff Peeler Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Manage a E1 system
On Tue, Sep 08, 2009 at 07:11:26PM +0500, silent sayz wrote: Hello Every one! I am little bit new to asterisk. I am doing research on different telecom options as well. I have question for you professionals In order to get E1 line working with Asterisk. What E1 line parameters need to be specified in Asterisk(configuration files). They vary from country to country. What is difference in one countries E1 and others E1. Actually E1 parameters don't differ that much by country. Do you have any specific examples? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Realtime static with Asterisk 1.6.1.6
Carlos Chavez escribió: I just upgraded from 1.6.0.14 to 1.6.1.6 and now my realtime static configuration for extensions.conf will not load. Just curious, is there any specific reason for you to upgrade from the latest 1.6.0.14 to 1.6.1? Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] 1.2 AGI Deadlock
A deadlock? In 1.2? Really? :) Peder wrote: I am running 1.2.34 (also tried on 1.2.32) and whenever I launch an AGI, I get the avoided deadlock message below. *CLI == Spawn extension (CONTEXT3, 6080, 8) exited non-zero on 'SIP/3211-1-081c40a8' -- Executing NoOp(SIP/3211-1-081c40a8, ) in new stack -- Executing AGI(SIP/3211-1-081c40a8, diallocal.agi) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/diallocal.agi Sep 8 10:29:43 WARNING[28938]: channel.c:780 channel_find_locked: Avoided initial deadlock for '0x818dcc0', 9 retries! -- AGI Script diallocal.agi completed, returning 0 This is purely a test box and it has two phones on it and one AGI, so there is no issue with utilization. Everything I read about deadlocks says this is a bad thing. I know it says avoided deadlock, but this happens every single time I use an AGI, even with nothing else happening on the box. Is this really something I should be concerned about, or is it no big deal? I am worried that if I put this into production with 200+ phones, it will cause Asterisk to die. Peder ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 -- Alex Balashov - Principal 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 -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] All hints say Hold
On Tue, 2009-09-08 at 13:03 +1200, Matt Riddell wrote: On 8/09/09 5:35 AM, Carlos Chavez wrote: Today is a strange day. My asterisk server is suddenly saying that all extensions are on hold. All my hints are like this: -= Registered Asterisk Dial Plan Hints =- 4...@hints : SIP/4101 State:HoldWatchers 0 4...@hints : SIP/4100 State:HoldWatchers 0 4...@hints : SIP/4002 State:HoldWatchers 0 4...@hints : SIP/4001 State:HoldWatchers 0 4...@hints : SIP/4000 State:HoldWatchers 0 2...@hints : SIP/2012 State:HoldWatchers 0 2...@hints : SIP/2003 State:HoldWatchers 0 2...@hints : SIP/2002 State:HoldWatchers 0 2...@hints : SIP/2001 State:HoldWatchers 0 1...@hints : SIP/1004 State:HoldWatchers 0 1...@hints : SIP/1003 State:HoldWatchers 0 1...@hints : SIP/1002 State:HoldWatchers 0 Reload SIP/Restart Phones/Make a change to SIP/Restart Asterisk :) Been there, done that, many times. I even upgraded Asterisk and got the same results. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] 1.2 AGI Deadlock
I am running 1.2.34 (also tried on 1.2.32) and whenever I launch an AGI, I get the avoided deadlock message below. *CLI == Spawn extension (CONTEXT3, 6080, 8) exited non-zero on 'SIP/3211-1-081c40a8' -- Executing NoOp(SIP/3211-1-081c40a8, ) in new stack -- Executing AGI(SIP/3211-1-081c40a8, diallocal.agi) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/diallocal.agi Sep 8 10:29:43 WARNING[28938]: channel.c:780 channel_find_locked: Avoided initial deadlock for '0x818dcc0', 9 retries! -- AGI Script diallocal.agi completed, returning 0 This is purely a test box and it has two phones on it and one AGI, so there is no issue with utilization. Everything I read about deadlocks says this is a bad thing. I know it says avoided deadlock, but this happens every single time I use an AGI, even with nothing else happening on the box. Is this really something I should be concerned about, or is it no big deal? I am worried that if I put this into production with 200+ phones, it will cause Asterisk to die. Peder ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Caller ID from POTS lines
Hi, I'm using asterisk 1.4.22-4 in Trixbox with snom 360 phones. When calls come in on our POTS lines, the caller id shows up like 555-555-1...@192.168.1.10 where 555-555-1234 is the correct phone number and 192.168.1.10 is my pbx server IP. This format does not work for redialing on outbound calls. While there may be an outbound dialing change that could be made, it seems like the correct solution would be to change the format of the caller id string sent to the phones. I verified from the snom sip trace that the caller id is always sent with @192.168.1.10 on it. What configuration change can be made in asterisk to correct this and only send the phone number as the caller id to the VOIP phone? Thanks, Jeremy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Realtime static with Asterisk 1.6.1.6
I just upgraded from 1.6.0.14 to 1.6.1.6 and now my realtime static configuration for extensions.conf will not load. All other realtime configs work (SIP, IAX2, Voicemail). I cannot find any reference or documentation about the structure of the realtime static database for 1.6.1.x but I have used the same table structure since 1.4.x. CREATE TABLE `ast_config` ( `id` int(11) NOT NULL auto_increment, `cat_metric` int(11) NOT NULL default '0', `var_metric` int(11) NOT NULL default '0', `commented` int(11) NOT NULL default '0', `filename` varchar(128) collate utf8_unicode_ci NOT NULL, `category` varchar(128) collate utf8_unicode_ci NOT NULL default 'default', `var_name` varchar(128) collate utf8_unicode_ci NOT NULL, `var_val` varchar(200) collate utf8_unicode_ci NOT NULL, PRIMARY KEY (`id`), KEY `filename_comment` (`filename`,`commented`) ) ENGINE=MyISAM DEFAULT CHARSET=utf8 COLLATE=utf8_unicode_ci; Does anyone know where I can find the table structure for 1.6.1? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] 1.2 AGI Deadlock
Peder wrote: I am running 1.2.34 (also tried on 1.2.32) and whenever I launch an AGI, I get the avoided deadlock message below. On Tue, 8 Sep 2009, Alex Balashov wrote: A deadlock? In 1.2? Really? :) Well, that was helpful. As a fellow 1.2 Luddite, I have boxes running xxx simultaneous channels, all running xx AGIs per call with no problems. First off, unless you have good reasons, you should move to a newer version just to improve your chances of getting meaningful support on this list. Do you get this deadlock message when you launch any AGI, for example, agi-test.agi? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Manage a E1 system
Thanks Tzafrir Cohen! May be i get this wrong http://www.voip-info.org/wiki/view/Asterisk+PRI#CountryVariations Any body help me what i must know about the E1 cable before asking a company to give me an E1 connection for Asterisk Digiums Card. Can i get complete list ??? Like Some one Advice for T1 as (copied from http://www.voip-info.org/wiki/view/Asterisk+config+zaptel.conf) Please Obtain the Line information from your carrier before connecting your T1 line.Questions to ask... Full T1? all 24 lines used? Line Type: Framing: Encoding: Switchtype: Cheers Adam ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Fwd: Patton smartnode 463x (BRI) 25ms tail echo cancellation
We have some installations with smartnode 4554, (same tail echo cancellation) without problems so far. Jorge Mendoza Gaëtan Minet wrote: Hi Is anybody using these ? Gaetan Begin forwarded message: *From: *Gaëtan Minet gminet...@mcit.be mailto:gminet...@mcit.be *Date: *Sat 22 Aug 2009 16:29:42 GMT+02:00 *To: *Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com *Subject: **[asterisk-users] Patton smartnode 463x (BRI) 25ms tail echo cancellation* *Reply-To: *Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com Hi all We use pci/pci-e BRI cards in our installations. Due to echo problems (that was before Oslec and others), we quickly switched to cards with hardware-based EC. So we use exclusively Digium B410p cards that provide 64ms tail EC. For several reasons we'd like to switch to external BRI gateways like the Patton smartnodes (the price is getting really close to a B410p). I'm however curious about their HW EC. I see in the datasheets that it only has 25ms tail per channel (pri are 128ms, but not BRI). Are some of you using these gateway and do your experience (many) echo problems on calls ? Our other alternative is to use sangoma cards that have 128ms HW EC and seem more stable overall, but it is yet a bit more expensive. Thanks for your feedback. Regards, Gaetan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] 1.2 AGI Deadlock
Un-top-posting... Peder wrote: I am running 1.2.34 (also tried on 1.2.32) and whenever I launch an AGI, I get the avoided deadlock message below. Sep 8 11:48:43 WARNING[564]: channel.c:780 channel_find_locked: Avoided initial deadlock for '0x818edf8', 9 retries! On Tue, 8 Sep 2009, Steve Edwards wrote: Do you get this deadlock message when you launch any AGI, for example, agi-test.agi? On Tue, 8 Sep 2009, Peder wrote: Yes, I get that message with any AGI: I bumped up the logging on a 1.2.34 box and I see that I'm getting the DEBUG message issued before your WARNING: Sep 8 10:11:48 ia02 asterisk[28290]: DEBUG[28295]: channel.c:775 in channel_find_locked: Avoiding initial deadlock for 'SIP/x.x.x.x-af850ed8' Sep 8 10:13:01 ia02 asterisk[28290]: DEBUG[21903]: channel.c:775 in channel_find_locked: Avoiding deadlock for 'SIP/x.x.x.x-afe4e658' I don't have any insight, but this box appears to be running fine -- at least no complaints from the client. If you find a solution or explanation please reply. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Realtime static with Asterisk 1.6.1.6
On Tue, 2009-09-08 at 11:54 -0500, Miguel Molina wrote: Carlos Chavez escribió: I just upgraded from 1.6.0.14 to 1.6.1.6 and now my realtime static configuration for extensions.conf will not load. Just curious, is there any specific reason for you to upgrade from the latest 1.6.0.14 to 1.6.1? Cheers, Well, yesterday my 1.6.0.14 (and .15) server went nuts. Hints were not working and several phones would not dial. Upgrading to 1.6.1 solved the problem. The only issue now is getting my dialplan to work from realtime static which works fine up to 1.6.0 -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] SIP Error
*I am getting below CLI in my asterisk :* == Manager 'sendcron' logged off from 127.0.0.1 -- Executing AGI(SIP/cc101-b7910cc0, agi://127.0.0.1:4577/call_log) in new stack -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 -- Executing Dial(SIP/cc101-b7910cc0, SIP/Sama203/119545090201||tTor) in new stack -- Called Sama203/119545090201 Sep 8 14:19:09 WARNING[2813]: chan_sip.c:9890 handle_response_invite: Forbidden - wrong password on authentication for INVITE to 'cc101 sip:xx...@203.196.128.56 sip%3axx...@203.196.128.56 ;tag=as09c56cf2' -- SIP/Sama203-09fbdaa0 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing Hangup(SIP/cc101-b7910cc0, ) in new stack == Spawn extension (default, 800119545090201, 3) exited non-zero on 'SIP/cc101-b7910cc0' -- Executing DeadAGI(SIP/cc101-b7910cc0, agi:// 127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-21-CONGESTION--) in new stack -- AGI Script agi:// 127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-21-CONGESTION--completed, returning 0 == Manager 'sendcron' logged off from 127.0.0.1 -- Executing AGI(SIP/cc101-b79017c8, agi://127.0.0.1:4577/call_log) in new stack -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 -- Executing Dial(SIP/cc101-b79017c8, SIP/Sama203/19545090201||tTor) in new stack -- Called Sama203/19545090201 Sep 8 14:19:53 WARNING[2813]: chan_sip.c:9890 handle_response_invite: Forbidden - wrong password on authentication for INVITE to 'cc101 sip:xx...@203.196.128.56 sip%3axx...@203.196.128.56 ;tag=as168401db' -- SIP/Sama203-09fbdaa0 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing Hangup(SIP/cc101-b79017c8, ) in new stack == Spawn extension (default, 80019545090201, 3) exited non-zero on 'SIP/cc101-b79017c8' -- Executing DeadAGI(SIP/cc101-b79017c8, agi:// 127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-21-CONGESTION--) in new stack -- AGI Script agi:// 127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-21-CONGESTION--completed, returning 0 My sip settings are : [Sama203] type=peer username= fromuser= authuser= secret=x host=203.xxx.xxx.56 fromdomain=203.xxx.xxx.56 nat=no canreinvite=yes insecure=very disallow=all allow=g729 context=default dtmfmode=rfc2833 It happens when I add 2 SIP in single asterisk server. 1.2.30.2 If I remove one, I dont get this error. Anyway to find out , what password asterisk recieves when I use Sama203 ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Realtime static with Asterisk 1.6.1.6
Carlos Chavez cur...@telecomabmex.com writes: On Tue, 2009-09-08 at 11:54 -0500, Miguel Molina wrote: Carlos Chavez escribió: I just upgraded from 1.6.0.14 to 1.6.1.6 and now my realtime static configuration for extensions.conf will not load. Just curious, is there any specific reason for you to upgrade from the latest 1.6.0.14 to 1.6.1? Cheers, Well, yesterday my 1.6.0.14 (and .15) server went nuts. Hints were not working and several phones would not dial. Upgrading to 1.6.1 solved the problem. The only issue now is getting my dialplan to work from realtime static which works fine up to 1.6.0 Bug 15852 perhaps, for the hints? Hopefully it isn't 15659 that is preventing you from dialing, because that one is in 1.6.1.6 as well. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] SIP Error
I have 2 sips configured : 1) register =sama:xx...@209.51.191.xxx:5060 2) register =sama:xx...@209.51.192.xxx:5060 Both are active. 5060 port will be same or different ? On Wed, Sep 9, 2009 at 12:29 AM, David @ULC ucoms2...@gmail.com wrote: *I am getting below CLI in my asterisk :* == Manager 'sendcron' logged off from 127.0.0.1 -- Executing AGI(SIP/cc101-b7910cc0, agi://127.0.0.1:4577/call_log) in new stack -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 -- Executing Dial(SIP/cc101-b7910cc0, SIP/Sama203/119545090201||tTor) in new stack -- Called Sama203/119545090201 Sep 8 14:19:09 WARNING[2813]: chan_sip.c:9890 handle_response_invite: Forbidden - wrong password on authentication for INVITE to 'cc101 sip:xx...@203.196.128.56 sip%3axx...@203.196.128.56 ;tag=as09c56cf2' -- SIP/Sama203-09fbdaa0 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing Hangup(SIP/cc101-b7910cc0, ) in new stack == Spawn extension (default, 800119545090201, 3) exited non-zero on 'SIP/cc101-b7910cc0' -- Executing DeadAGI(SIP/cc101-b7910cc0, agi:// 127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-21-CONGESTION--) in new stack -- AGI Script agi:// 127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-21-CONGESTION--completed, returning 0 == Manager 'sendcron' logged off from 127.0.0.1 -- Executing AGI(SIP/cc101-b79017c8, agi://127.0.0.1:4577/call_log) in new stack -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 -- Executing Dial(SIP/cc101-b79017c8, SIP/Sama203/19545090201||tTor) in new stack -- Called Sama203/19545090201 Sep 8 14:19:53 WARNING[2813]: chan_sip.c:9890 handle_response_invite: Forbidden - wrong password on authentication for INVITE to 'cc101 sip:xx...@203.196.128.56 sip%3axx...@203.196.128.56 ;tag=as168401db' -- SIP/Sama203-09fbdaa0 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing Hangup(SIP/cc101-b79017c8, ) in new stack == Spawn extension (default, 80019545090201, 3) exited non-zero on 'SIP/cc101-b79017c8' -- Executing DeadAGI(SIP/cc101-b79017c8, agi:// 127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-21-CONGESTION--) in new stack -- AGI Script agi:// 127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-21-CONGESTION--completed, returning 0 My sip settings are : [Sama203] type=peer username= fromuser= authuser= secret=x host=203.xxx.xxx.56 fromdomain=203.xxx.xxx.56 nat=no canreinvite=yes insecure=very disallow=all allow=g729 context=default dtmfmode=rfc2833 It happens when I add 2 SIP in single asterisk server. 1.2.30.2 If I remove one, I dont get this error. Anyway to find out , what password asterisk recieves when I use Sama203 ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Function to query ASTDB families
Hi,
Asterisk database is made of familykey records such as:
fam key1 val1
fam key2 val2
...
fam key100 val100
I'm looking for the smartest way to iterate among different keys associated
to a given family.
One way to do this is to parse database show fam response.
Is there something smarter ?
Something like ${DBKEYS(fam)} which would evaluate to key1 key2 ...
key100.
Regards
___
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Re: [asterisk-users] 1.6.1 + TDM840 FSK MWI problem
- Barry Miller asterisk-us...@notanet.net wrote: On Fri, Sep 04, 2009 at 04:10:43PM -0500, Doug Bailey wrote: - Barry Miller asterisk-us...@notanet.net wrote: Hi, Using 1.4.26.1 DAHDI 2.2.0.2, FSK VMWI devices off a TDM840 work fine. With 1.6.1.[45] same DAHDI, instead of the FSK spill I get a line polarity reversal. Stutter dialtone is generated as expected. Has anyone else seen this? Is there anything special I need to do for 1.6.1 to make FSK MWI work? [snip] The only thing I can think of that would be preventing the output would be problems in the interface chip with the On-Hook transfer mode. If you run a dahdi_monitor on the channel that should be sending the FSK spill and look at the results in a program like audacity, you can see if the MWI FSK spill is actually reaching the interface SLIC IC. Something like dahdi_monitor 1 -t spilloutput.raw (Monitors the output going to dahdi channel 1.) Hmm. With both 1.4 1.6, without touching /etc/[asterisk|dahdi], I used a butt-set to go off-hook, then back on. I got: 1.4.26.1: dahdi_monitor captured stutter dialtone, 4.5 seconds of silence, then the FSK spill. And that's what I heard. 1.6.1.6: dahdi_monitor captured stutter dialtone, 1.5 seconds of silence, then the FSK spill. Sounds good with audacity. But all I heard through the butt-in was stutter dialtone. No FSK spill at all. Here's hoping this tells you more than it does me :) Actually it does tell me a lot. The problem appears in how the interface chip is being programmed. For some reason, the interface chip is not being set to on-hook transfer mode which would allow for the mwi spill to go out on the actual fxs port lines. I am looking to see where the problem lies. (It is either in chan_dahdi or in the driver.) I hope to have more information later. Doug ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Fax For Asterisk and SendFax question
That's the general idea. The application is designed to send a TIFF over an established connection. You can detect that it is a fax or just assume so. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joan Antoni Terre Sent: Tuesday, September 08, 2009 4:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Fax For Asterisk and SendFax question Hi everybody, I've installed Free Fax For Asterisk in my Asterisk box but I don't understand how it works as when using SendFax application from dialplan, I can't find how to introduce destination fax number. How this application works? Do I have to call destination fax using Dial application, detect somehow that it's a fax and then use SendFAX application specifying FAXOPTs and the path to the fax file? Many thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Fax For Asterisk and SendFax question
Hi everybody, I've installed Free Fax For Asterisk in my Asterisk box but I don't understand how it works as when using SendFax application from dialplan, I can't find how to introduce destination fax number. How this application works? Do I have to call destination fax using Dial application, detect somehow that it's a fax and then use SendFAX application specifying FAXOPTs and the path to the fax file? Many thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Fax For Asterisk and SendFax question
thanks Danny, just another stupid question, as far as I know, when a call is answered after Dial application, it doesn't execute other dialplan priorities until it's hung up, which execute h priority, so how can I make it execute a SendFAX, or whatever else, when it's answered? thanks again 2009/9/8 Danny Nicholas da...@debsinc.com That’s the general idea. The application is designed to send a TIFF over an established connection. You can detect that it is a fax or just assume so. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Joan Antoni Terre *Sent:* Tuesday, September 08, 2009 4:52 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Fax For Asterisk and SendFax question Hi everybody, I've installed Free Fax For Asterisk in my Asterisk box but I don't understand how it works as when using SendFax application from dialplan, I can't find how to introduce destination fax number. How this application works? Do I have to call destination fax using Dial application, detect somehow that it's a fax and then use SendFAX application specifying FAXOPTs and the path to the fax file? Many thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Should digium build a 2FXO / 2FXS 4-port daughter board?
Please chime in if you've ever wished for digium to make a 4-port daughter board with a combination of 2FXO AND 2FXS ports on the same card. When using the 800 series cards, one must either choose 4-port permutations of FXS/FXO, OR one must give up 2 valuable ports. In other words, when you add ONE 100-series daughter board, you give up TWO of your physical ports. Is there a technical reason for the lack of such a card, or is it just a case of insufficient business case to justify the development? If it's the latter, I would like to state that I am ready to buy several '2x2' cards. I also assume I am not alone. I often run into situations where I am space constrained (embedded enclosure) or resource constrained (bus slots 1u rackmount) which rule out the obvious solution of a larger card (AEX/TDM2400P) and/or rule out the inclusion of an additional 4-port interface card (2x TDM400). A '2x2' 4-port 'combo' daughter board could not only allow 2-port granularity vs. 4 port granularity in the 800 and 2400 series boards, but it would also be a very attractive option as a 'beginner' interface card, allowing 2 ports of each interface, while allowing lots of flexibility in the future--4 more FXO, 4 more FXS, OR just another 2x2 card for a symmetrical 4x4 setup. In my case, I find myself most often needing a 6 x 2 setup Please chime in if you've ever wished for a product like this from Digium. Thanks! -Karl ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Fax For Asterisk and SendFax question
You can use the M parameter to run a macro after the channel picks up or the g parameter to jump to a given context. there is also a parameter to run an AGI script. Check the dial() cmd on the wiki for further details. Erik de Wild Tripple-o Your Asterisk migration partner the Netherlands Verstuurd vanaf mijn iPhone Op 9 sep 2009 om 00:43 heeft Joan Antoni Terre nebh...@gmail.com het volgende geschreven:\ thanks Danny, just another stupid question, as far as I know, when a call is answered after Dial application, it doesn't execute other dialplan priorities until it's hung up, which execute h priority, so how can I make it execute a SendFAX, or whatever else, when it's answered? thanks again 2009/9/8 Danny Nicholas da...@debsinc.com That’s the general idea. The application is designed to send a TIFF over an established connection. You can detect that it is a fax or just assume so. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com ] On Behalf Of Joan Antoni Terre Sent: Tuesday, September 08, 2009 4:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Fax For Asterisk and SendFax question Hi everybody, I've installed Free Fax For Asterisk in my Asterisk box but I don't understand how it works as when using SendFax application from dialplan, I can't find how to introduce destination fax number. How this application works? Do I have to call destination fax using Dial application, detect somehow that it's a fax and then use SendFAX application specifying FAXOPTs and the path to the fax file? Many thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Asterisk CLI commands not running !!!!!
Un-top-posting... From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of abdelkader I am using Asterisk 1.4.22 in a debian 4.0 with a kernel version 2.6.18-6-amd64 (SMP). The processor type is Intel(R) Xeon(R) CPU E5420 @ 2.50GHz. Sometimes, I get a strange behavior from asterisk: The CLI commands does not work and Asterisk cannot receive calls. The output of every CLI command is that command is not known (no such command). Just grasping at straws... How often does this happen? Does sudo lsof | grep /usr/lib/asterisk/modules/ show the modules you expect? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mindaugas Kezys Asterisk sometimes goes to sleep. (And never wakes-up). Restart it and all will be fine again. We have a watchdog which sends SIP OPTIONS packet to Asterisk and if it does not respond ? restarts it. Please share :) On Wed, 9 Sep 2009, Lee, John (Sydney) wrote: I have a cron job that restarts Asterisk every night. This is supposed to be an old Asterisk best practice for 1.2.* but I think it does not harm. Unless you're running 24x7x365. I have a 1.2.7 system (with custom hacks) that needs to be restarted every 3 or 4 months due to a memory leak. I had (until last weekend) a 1.2.2x system that had been running for over 600 days. Both systems handle about 15k calls a day. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Asterisk CLI commands not running !!!!!
I have a cron job that restarts Asterisk every night. This is supposed to be an old Asterisk best practice for 1.2.* but I think it does not harm. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mindaugas Kezys Sent: Tuesday, 8 September 2009 10:10 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Asterisk CLI commands not running ! Asterisk sometimes goes to sleep. (And never wakes-up). Restart it and all will be fine again. We have a watchdog which sends SIP OPTIONS packet to Asterisk and if it does not respond – restarts it. Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of abdelkader Sent: 2009 m. rugsėjo 8 d. 10:40 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk CLI commands not running ! Hello, I am using Asterisk 1.4.22 in a debian 4.0 with a kernel version 2.6.18-6-amd64 (SMP). The processor type is Intel(R) Xeon(R) CPU E5420 @ 2.50GHz. Sometimes, I get a strange behavior from asterisk: The CLI commands does not work and Asterisk cannot receive calls. The output of every CLI command is that command is not known (no such command). Please help me resolve this problem: what can be the cause of it? is it Asterisk or my system? and what have I to do to eliminate this problem? Thks in advance. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Should digium build a 2FXO / 2FXS 4-port daughterboard?
Definitely, 10 votes from me. For the home user, 2xFXO + 6FXS, in a single slot small profile box is ideal, but only able to offer 2xFXO + 4xFXS at the moment. SIP phones don't exactly have the appropriate WIFE factor. A standard off the shelve, no frills phone does the job. Alec Davis -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Karl Fife Sent: Wednesday, 9 September 2009 10:51 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Should digium build a 2FXO / 2FXS 4-port daughterboard? Please chime in if you've ever wished for digium to make a 4-port daughter board with a combination of 2FXO AND 2FXS ports on the same card. When using the 800 series cards, one must either choose 4-port permutations of FXS/FXO, OR one must give up 2 valuable ports. In other words, when you add ONE 100-series daughter board, you give up TWO of your physical ports. Is there a technical reason for the lack of such a card, or is it just a case of insufficient business case to justify the development? If it's the latter, I would like to state that I am ready to buy several '2x2' cards. I also assume I am not alone. I often run into situations where I am space constrained (embedded enclosure) or resource constrained (bus slots 1u rackmount) which rule out the obvious solution of a larger card (AEX/TDM2400P) and/or rule out the inclusion of an additional 4-port interface card (2x TDM400). A '2x2' 4-port 'combo' daughter board could not only allow 2-port granularity vs. 4 port granularity in the 800 and 2400 series boards, but it would also be a very attractive option as a 'beginner' interface card, allowing 2 ports of each interface, while allowing lots of flexibility in the future--4 more FXO, 4 more FXS, OR just another 2x2 card for a symmetrical 4x4 setup. In my case, I find myself most often needing a 6 x 2 setup Please chime in if you've ever wished for a product like this from Digium. Thanks! -Karl ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Inquiry:Problem with Call Parking
Thank you for your message . But I tried to find it on my server , as the followings : #find / -name sip.cfg -print But it didn't return any result . Can you please let me know where can I find it ? On Tue, Sep 8, 2009 at 2:58 PM, Steve Edwards asterisk@sedwards.comwrote: On Tue, 8 Sep 2009, hadi motamedi wrote: I sent you a message regarding my problem with Asterisk Call Parking feature and you told me that needs to check the polycom sip.cfg file . But my Asterisk doesn't have sip.cfg file . Can you please let me know how can I overcome ? sip.cfg is not an Asterisk file. sip.cfg should be in the directory the phone downloads it's configuration from. Typically, /tftpboot/ on a tftp server. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Inquiry:Problem with Call Parking
On Wed, 9 Sep 2009, hadi motamedi wrote: Thank you for your message . But I tried to find it on my server , as the followings : #find / -name sip.cfg -print But it didn't return any result . Can you please let me know where can I find it ? You probably have not setup central provisioning for your Polycom phones. I am guessing you are configuring them from their (horribly crappy) web interface. Although this kind of works, you will not be able to unleash the true power of your phones without setting up central provisioning. Worse you may be running an old version of the firmware, which may have problems. This involves getting the firmware and XML templates from Polycom, which will include the file sip.cfg. You will have to unpack these files on a TFTP or HTTP server, create XML files for each phone, and point the phone to the server to pick it up. There are numerous howtos on the web to set this up. Time for Google! j On Tue, Sep 8, 2009 at 2:58 PM, Steve Edwards asterisk@sedwards.comwrote: On Tue, 8 Sep 2009, hadi motamedi wrote: I sent you a message regarding my problem with Asterisk Call Parking feature and you told me that needs to check the polycom sip.cfg file . But my Asterisk doesn't have sip.cfg file . Can you please let me know how can I overcome ? sip.cfg is not an Asterisk file. sip.cfg should be in the directory the phone downloads it's configuration from. Typically, /tftpboot/ on a tftp server. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] RESET CDR
Hello, How can I reset CDR time , let's say after 30 seconds of answer signal, reset CDR to 0 , any idea ?? Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Should digium build a 2FXO / 2FXS 4-port daughterboard?
At 07:31 PM 9/8/2009, you wrote: For the home user, 2xFXO + 6FXS, in a single slot small profile box is ideal, but only able to offer 2xFXO + 4xFXS at the moment. Wow, I can't imagine ever using an analog phone on Asterisk. SIP phones are just so much better! Ira ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Should digium build a 2FXO / 2FXS 4-port daughterboard?
On Tue, 8 Sep 2009, Ira wrote: At 07:31 PM 9/8/2009, you wrote: For the home user, 2xFXO + 6FXS, in a single slot small profile box is ideal, but only able to offer 2xFXO + 4xFXS at the moment. Wow, I can't imagine ever using an analog phone on Asterisk. SIP phones are just so much better! Ira Actually what I thought was funny was the idea that 2 lines and 6 extensions was ideal for the home user :) Some of us should probably get out into the sun more often. j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] RESET CDR
On 9/09/09 4:34 PM, B.Masoud @ SH wrote: Hello, How can I reset CDR time , let’s say after 30 seconds of answer signal, reset CDR to 0 , any idea ?? :) Use the ResetCDR application? -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] RESET CDR
A little more help is appreciated, I know about ResetCDR() , but I want some code that resets the call data after 30 seconds! And where to put the code exactly. Thanks. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Riddell Sent: Wednesday, September 09, 2009 7:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RESET CDR On 9/09/09 4:34 PM, B.Masoud @ SH wrote: Hello, How can I reset CDR time , let's say after 30 seconds of answer signal, reset CDR to 0 , any idea ?? :) Use the ResetCDR application? -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] RESET CDR
On 9/09/09 5:14 PM, B.Masoud @ SH wrote: A little more help is appreciated, I know about ResetCDR() , but I want some code that resets the call data after 30 seconds! And where to put the code exactly. What a strange request. Why exactly are you wanting to do this? If you're wanting all your calls to look like they are 30 seconds shorter can't you just use the time-30 seconds? -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Older Aastra phones and Asterisk 1.6
On Monday 07 September 2009 16:27:30 Carlos Chavez wrote: It seems that older Aastra phones (9112i, 9133i, 480i, 480i CT) have a problem with the new SIP implementation in Asterisk 1.6.X that makes them unable to dial. They can receive calls but when you attempt to dial the phone remains silent. You can see in core show channels that the first channel is active and it is impossible to kill it without restarting Asterisk. The solution I found for this is to set session-timers=refuse in sip.conf and now I am able to send calls. I suppose this is a problem with the firmware of those phones as newer versions of Aastra phones (5Xi) work without the modification. I have several Aastra 480i CT phones on three separate Asterisk 1.6.1.6 on Fedora 11 (asterisk-1.6.1.6-1.fc11.x86_64) and do not see this problem. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
