[asterisk-users] MeetMe and usernum
hi, I am trying to get the usernum of a user when dialing in to a MeetMe conference. Is there somehow a possibility to save the usernum of a MeetMe participant into a variable? Everything should be done through the DialPlan, no manager and no *cli. Thanks for your help, Emrah -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe and usernum
Hi! Thanks a lot for your answer. The problem with the command you mentioned is... When do I call it? If two people happen to enter the conf at the sametime, I have a feeling there may be some little confusion there... Do you think I could use the agi-background option with meetme? I am using 1.6. Thanks again guys! Emrah On Mon, Mar 01, 2010 at 09:45:33AM -0500, David Backeberg dbackeb...@gmail.com wrote: On Mon, Mar 1, 2010 at 6:42 AM, Emrah e...@ekanet.net wrote: I am trying to get the usernum of a user when dialing in to a MeetMe conference. Is there somehow a possibility to save the usernum of a MeetMe participant into a variable? Everything should be done through the DialPlan, no manager and no *cli. You don't say what version you're running. I second Steve's claim. Even with 1.6, I can't think of how to do what you want without resorting to AGI. Which is technically in the dialplan, but you're going to have to do extra work elsewhere. If you're using 1.6, you will enjoy knowing about 'meetme list concise', which you can then process with awk. If you absolutely don't want to do AGI, you could always modify meetme.c, recompile, and share your work with others. I think you'll find that harder than writing an AGI. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Emrah KAVUN e...@ekanet.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Feature request: Meetme and invisible users
Hi, Sometimes during a confcall I generate calls into meetme to playback some announcements... If during that time someone joins the conference, the number of participants is announced counting my announcement call... Will it be possible to add an option to the meetme application to mark a participant unvisible and not count it in meetmecount? Basically instead of saying there are 3 other participants in this conference call, asterisk says there are 4 participants in this conference call because my announcement call is considered as a regular participant. Regards, emrah ___ -- 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] Skype for Asterisk: Public Beta available
Hi, This is the error message I get. Any idea where I may find some further debug logs? [Aug 3 08:01:23] ERROR[23831] chan_skype.c: Unable to start Skype For Asterisk library. Thanks, Emrah Tim Panton wrote: I don't know then. My understanding is that the message is caused by the wrong skypeforasterisk process running. - did you (ever) run it as a different user ? If it is a test box, you could try a full reboot. Tim. On 2 Aug 2009, at 19:35, Emrah wrote: Hi Tim, I don't have any skypeforasterisk process running. I tried to killall -9 asterisk but it did not solve my issue. Any other suggestions? Thanks for your help, Emrah Tim Panton wrote: I had that too, I cured it by kill -9 'ing the skypeforasterisk process that was left over from the previous version of the beta. Hope that helps. Tim. On 2 Aug 2009, at 11:20, Emrah wrote: I reported an issue on Mantis (#14). Waiting for an update. http://betareports.digium.com/mantis/view.php?id=14 Emrah Emrah wrote: Hi Thomas, I am experiencing the same problem, with the same error messages. Running Asterisk 1.6.1.0 on a Debian 2.6.24-etchnhalf.1-686 Regards, Emrah Thomas Kenyon wrote: Thomas Kenyon wrote: [2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x25765ca0 for 0x1390e20 [2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x25765ca0 for 0x1390e20 chef*CLI skype show users Skype Users [2009-08-02 09:39:35] ERROR[25506]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x70796b73 for 0x7f4fe0044340 [2009-08-02 09:39:35] ERROR[25506]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x70796b73 for 0x7f4fe0044340 Sorry, these are the error messages. ___ -- 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 Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk ___ -- 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 Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk ___ -- 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] Skype for Asterisk: Public Beta available
Hi Thomas, I am experiencing the same problem, with the same error messages. Running Asterisk 1.6.1.0 on a Debian 2.6.24-etchnhalf.1-686 Regards, Emrah Thomas Kenyon wrote: Thomas Kenyon wrote: [2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x25765ca0 for 0x1390e20 [2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x25765ca0 for 0x1390e20 chef*CLI skype show users Skype Users [2009-08-02 09:39:35] ERROR[25506]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x70796b73 for 0x7f4fe0044340 [2009-08-02 09:39:35] ERROR[25506]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x70796b73 for 0x7f4fe0044340 Sorry, these are the error messages. ___ -- 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] Skype for Asterisk: Public Beta available
I reported an issue on Mantis (#14). Waiting for an update. http://betareports.digium.com/mantis/view.php?id=14 Emrah Emrah wrote: Hi Thomas, I am experiencing the same problem, with the same error messages. Running Asterisk 1.6.1.0 on a Debian 2.6.24-etchnhalf.1-686 Regards, Emrah Thomas Kenyon wrote: Thomas Kenyon wrote: [2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x25765ca0 for 0x1390e20 [2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x25765ca0 for 0x1390e20 chef*CLI skype show users Skype Users [2009-08-02 09:39:35] ERROR[25506]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x70796b73 for 0x7f4fe0044340 [2009-08-02 09:39:35] ERROR[25506]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x70796b73 for 0x7f4fe0044340 Sorry, these are the error messages. ___ -- 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] Skype for Asterisk: Public Beta available
Hi Tim, I don't have any skypeforasterisk process running. I tried to killall -9 asterisk but it did not solve my issue. Any other suggestions? Thanks for your help, Emrah Tim Panton wrote: I had that too, I cured it by kill -9 'ing the skypeforasterisk process that was left over from the previous version of the beta. Hope that helps. Tim. On 2 Aug 2009, at 11:20, Emrah wrote: I reported an issue on Mantis (#14). Waiting for an update. http://betareports.digium.com/mantis/view.php?id=14 Emrah Emrah wrote: Hi Thomas, I am experiencing the same problem, with the same error messages. Running Asterisk 1.6.1.0 on a Debian 2.6.24-etchnhalf.1-686 Regards, Emrah Thomas Kenyon wrote: Thomas Kenyon wrote: [2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x25765ca0 for 0x1390e20 [2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x25765ca0 for 0x1390e20 chef*CLI skype show users Skype Users [2009-08-02 09:39:35] ERROR[25506]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x70796b73 for 0x7f4fe0044340 [2009-08-02 09:39:35] ERROR[25506]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x70796b73 for 0x7f4fe0044340 Sorry, these are the error messages. ___ -- 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 Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk ___ -- 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] Maximum number of concurrent calls
Hi, I remember reading that Asterisk allows only 100 simultaneous calls. Is that correct? If it is so, how is it possible to have a conference call with more then 100 users? I think I read here that some people managed to have 500 people in a conf room... Or, how do I increase this limit? Is it as easy as changing a value in a config.h? Thanks for your help, Emrah ___ -- 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] Voicemail feature: enable or disable the ability to leave a message
Hi, I think there is an essential option of the Voicemail application that is missing. I would like to suggest the implementation of a function to give the user the ability to either allow or disallow the recording of messages. If the ability to record a message is disabled, options u, s, and b must not be considered in order to avoid the playback of messages such as Please leave your message after the tone... the usecase is simple. A person could record a greeting that says please callback later instead of asking to leave a message. usefull also to record afterhour messages. What do you think? Regards, Emrah ___ -- 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] Voicemail feature: enable or disable the ability to leave a message
Mark, I think you did not understand my message. I am accustomed to have the option to allow or disallow the recording of a message in my voicemail, even my mobile carrier provides it. E.g.: I can record a message that says Please call back later, I am currently on the phone. without any beep tone or the possibility to leave a message. I know everything is possible to be done via the Dialplan. I could just have a Playback to achieve this or pretty easily code my own voicemail app via AGI too. I don't catch what you are trying to tell about reading the input in the Dialplan... My scenario is pretty simple, there should be an option in the voicemail to allow the user to choose whether he accepts messages or not. Cheers! Emrah Mark Michelson wrote: Emrah wrote: Hi, I think there is an essential option of the Voicemail application that is missing. I would like to suggest the implementation of a function to give the user the ability to either allow or disallow the recording of messages. If the ability to record a message is disabled, options u, s, and b must not be considered in order to avoid the playback of messages such as Please leave your message after the tone... the usecase is simple. A person could record a greeting that says please callback later instead of asking to leave a message. usefull also to record afterhour messages. What do you think? Regards, Emrah There is no reason to place this logic in the Voicemail application itself. If you wish to give users the option of leaving a voicemail, it can easily be done in the dialplan by playing prompts and reading the input of the user. Then, based on the input, you can choose whether to run the Voicemail application. Mark Michelson ___ -- 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] Voicemail feature: enable or disable the ability to leave a message
Doug, Thanks for the suggestion. I know there are plenty of workarounds there, I am not asking how to do it because I know how to do it too. What I am saying is that it could be an embedded feature in the Voicemail application, like the recent ability to flag a message as urgent. Regards, Emrah Doug Lytle wrote: Emrah wrote: Mark, I think you did not understand my message. I am accustomed to have the option to allow or disallow the recording of a message in my voicemail, even my mobile carrier provides it. E.g.: I The simplest thing to do is to allow users to set a flag, maybe using mysql or the astdb, if they want that option. And, in your dial plan, check for the existance of that flag. If it's there, then don't jump to the voice mail app, just jump to your context that would play back an audio file that the user has pre-recorded 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] Voicemail feature: enable or disable the ability to leave a message
Dom, Quoting myself from your original message: I know everything is possible to be done via the Dialplan. I could just have a Playback to achieve this or pretty easily code my own voicemail app via AGI too. I am not asking how to do this. I know everything is possible with Asterisk. In my opinion, the feature I am talking about is a very basic feature of Voicemail systems and I think it should be natively implemented in Asterisk. Thanks for your hint though. :) Emrah Don Kelly wrote: How 'bout setting up an extension that simply plays an announcement and hangs up. Then transfer calls from extensions that don't want messages to this extension. You could have a few extensions with a few different recordings to suit different situations. --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Emrah Sent: Friday, July 31, 2009 11:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Voicemail feature: enable or disable the ability to leave a message Mark, I think you did not understand my message. I am accustomed to have the option to allow or disallow the recording of a message in my voicemail, even my mobile carrier provides it. E.g.: I can record a message that says Please call back later, I am currently on the phone. without any beep tone or the possibility to leave a message. I know everything is possible to be done via the Dialplan. I could just have a Playback to achieve this or pretty easily code my own voicemail app via AGI too. I don't catch what you are trying to tell about reading the input in the Dialplan... My scenario is pretty simple, there should be an option in the voicemail to allow the user to choose whether he accepts messages or not. Cheers! Emrah Mark Michelson wrote: Emrah wrote: Hi, I think there is an essential option of the Voicemail application that is missing. I would like to suggest the implementation of a function to give the user the ability to either allow or disallow the recording of messages. If the ability to record a message is disabled, options u, s, and b must not be considered in order to avoid the playback of messages such as Please leave your message after the tone... the usecase is simple. A person could record a greeting that says please callback later instead of asking to leave a message. usefull also to record afterhour messages. What do you think? Regards, Emrah There is no reason to place this logic in the Voicemail application itself. If you wish to give users the option of leaving a voicemail, it can easily be done in the dialplan by playing prompts and reading the input of the user. Then, based on the input, you can choose whether to run the Voicemail application. Mark Michelson ___ -- 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
