Re: [asterisk-users] [Asterisk-video] (no subject)
http://secret.loynin.com Pankaj Pandey -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-video mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-video-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [Asterisk-video] (no subject)
http://fit.diggitradio.com Pankaj Pandey -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-video mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-video-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Ubuntu Asterisk 11.17.1 - segfault ERROR 4
Hi All, I am running Asterisk 11.17.1 on Ubuntu 11.10 and i am getting segfault error very frequently. Due to this my asterisk server dies and i am getting the following following error in /var/log/kern.log , Apr 22 14:21:03 pp kernel: [ 369.264497] asterisk[1267]: segfault at 986e000 ip b7689ad7 sp b47e32ac error 6 in libc-2.13.so[b760f000+17c000] Apr 22 14:21:38 pp kernel: [ 404.258595] asterisk[4136]: segfault at 69657461 ip b4623b19 sp b4a2523c error 4 in libgcc_s.so.1[b460e000+1c000] Apr 22 14:52:38 pp kernel: [ 2263.683388] asterisk[4545]: segfault at 8 ip b7638551 sp b46702f0 error 6 in libc-2.13.so[b75c5000+17c000] Any suggestions ... Thanks Regards, Pankaj Pandey +91-9990212758-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] .call file retry issue in Asterisk-10.11.1
Hi, I am working on Asterisk-10.11.1,I tried to generating outbound call through .call file and facing a issue that call retry was happening after call Answered.Is it bug in that Version or i missed some thing. Here is my call file is- Channel: DAHDI/G1/09990212758 MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: menu Extension: 1234 Priority: 4 Please suggest. Thanks Regards, Pankaj Pandey +91-9990212758-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk conferencing |MEETME or app_conference
Hi All, I am googling from few days back for a conference utility which fulfill my below scenario ,Please give your suggestion to fulfill my need. I have a scenario where leader is giving a lecture and other participants are on mute mode... At the end of conference, when QA session begins, is there a way for participants to raise hands, if they have any questions so Leader can unmute them? Is this feature already there in Meetme conference, if there then how can I implement this? Is there another utility which works in above scenario,what about app_conference? Please suggest ... Thanks Regards, Pankaj Pandey +91-9990212758-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk module app_konference
Hi all, I am looking for a complete conferencing solution over asterisk (meetme is not fulfill my needs) . I googled a lot and see a lot of stuff on appkonference. Is anybody using this module? Please suggest me and give me some feedback on it. Thanks!!! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hand Raise|Meetme Conf
Hi All, I am am looking for the below feature of asterisk MEETME. I googled a lot but did not find any help. Any body please suggest, how can we do it. Thanks!!! -- On Thu 20 Sep, 2012 3:11 PM EDT pankaj pandey wrote: Hi All, I have a scenario where leader is giving a lecture and other participants are on mute... At the end of conf , when QA session begins is there a way for participants to raise hands if they have questions so Leader can unmute them. Is this feature already there in Meetme conf ? If there then how can i implement this. please suggest... Thanks Regards, Pankaj Pandey +91-9990212758 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hand Raise|Meetme Conf
Hi All, I am am looking for the below feature of asterisk MEETME. I googled a lot but did not find any help. Any body please suggest, how can we do it. Thanks!!! -- On Thu 20 Sep, 2012 3:11 PM EDT pankaj pandey wrote: Hi All, I have a scenario where leader is giving a lecture and other participants are on mute... At the end of conf , when QA session begins is there a way for participants to raise hands if they have questions so Leader can unmute them. Is this feature already there in Meetme conf ? If there then how can i implement this. please suggest... Thanks Regards, Pankaj Pandey +91-9990212758 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-users Digest, Vol 98, Issue 38
Hi Danny, Thank you for your prompt response. The way you are suggesting is great . Infect asterisk have its own functionality that if user presses *1 during meetme conferencing asterisk automatically unmute that user and user comes in talking mode.But it is not fulfill my need. There is and issue that if 3-4 user presses *1 at the same time than how can i decide that who is asking the question and how can we manage that situation. Please suggest the another way to doing this. -Thanks !!! Message: 6 Date: Tue, 25 Sep 2012 13:15:32 -0500 From: Danny Nicholas da...@debsinc.com Subject: Re: [asterisk-users] Hand Raise|Meetme Conf To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Message-ID: 00df01cd9b49$c03b44c0$40b1ce40$@debsinc.com Content-Type: text/plain; charset=iso-8859-1 This might work: [meetme-with-handraise] Exten = s,1,meetme(1234,mX5) Exten = s,n,hangup Exten = 5,1,meetme(1234) Exten = 5,2,goto(meetme-with-handraise,s,1) Exten = I,1,playback(invalid) Exten = I,n,goto(meetme-with-handraise,s,1) According to the documentation, if the user presses 5, it should end their muted session and put them back in a talking mode. You could use X4 to return them to muted mode by pressing 4. Haven't tested it, but it's not a difficult thing to try. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of pankaj pandey Sent: Tuesday, September 25, 2012 12:56 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Hand Raise|Meetme Conf Hi All, I am am looking for the below feature of asterisk MEETME. I googled a lot but did not find any help. Any body please suggest, how can we do it. Thanks!!! -- On Thu 20 Sep, 2012 3:11 PM EDT pankaj pandey wrote: Hi All, I have a?scenario where leader is giving a lecture and other participants are on mute... At the end of conf , when QA session begins is there a way for participants to raise hands if they have questions so Leader can unmute them. Is this feature already there?in Meetme conf?? If there then how can i?implement this. please?suggest... ? Thanks Regards, Pankaj Pandey +91-9990212758 _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hand Raise|Meetme Conf
Hi All, I have a scenario where leader is giving a lecture and other participants are on mute... At the end of conf , when QA session begins is there a way for participants to raise hands if they have questions so Leader can unmute them. Is this feature already there in Meetme conf ? If there then how can i implement this. please suggest... Thanks Regards, Pankaj Pandey +91-9990212758-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hand Raise|Meetme Conf
thanks for your prompt response . I am using asterisk 1.4.32 ,please suggest which version i have to use. Thanks Regards, Pankaj Pandey +91-9990212758 From: Danny Nicholas da...@debsinc.com To: 'pankaj pandey' pankaj.n...@yahoo.com; 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Friday, 21 September 2012 12:43 AM Subject: RE: [asterisk-users] Hand Raise|Meetme Conf Which Asterisk version? From:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of pankaj pandey Sent: Thursday, September 20, 2012 2:12 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Hand Raise|Meetme Conf Hi All, I have a scenario where leader is giving a lecture and other participants are on mute... At the end of conf , when QA session begins is there a way for participants to raise hands if they have questions so Leader can unmute them. Is this feature already there in Meetme conf ? If there then how can i implement this. please suggest... Thanks Regards, Pankaj Pandey +91-9990212758-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Exceptionally long voice queue length in asterisk 1.6.2
Hi , I am using asterisk SVN-branch-1.6.2 version when i am making a call from SIP phone i found a warning of Exceptionally long voice queue length . When i search it on forum i found that This sounds like issue 15609 which has been resolved newer versions of asterisk https://issues.asterisk.org/view.php?id=15609 is it fixed in asterisk-1.6.2 SVN trunk version or not? thanks, Pankaj Thanks Regards, Pankaj Pandey +91-9990212758 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk+h324m gateway issue
Hi , i worked with h324m gateway for 3g video calling .It configured successfully . my code in extensions.conf is [from-zaptel] exten = _X.,1,h324m_gw(0@mainmenu) exten=_X.,n,Hangup [mainmenu] exten = 0,1,h324m_gw_answer() exten = 0,2,mp4play(/tmp/menu/menu.mp4,'n(1)') when i make a video call (either sip or through pri) , asterisk cli shows the following error -- Executing [123@from-zaptel:1] h324m_gw(SIP/100-b7602680, 0@mainmenu) in new stack localhost*CLI Disconnected from Asterisk server Executing last minute cleanups when i routed the call directly to [mainmenu] call stack at h324m_gw_answer() please help me ... Thanks Regards, Pankaj Pandey -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 3g call support for ISDN line
Dear All, i have a problem with 3g calling in asterisk with ISDN support . i tried it with the help of H324M gw . can any one tell that how i configure H324M gw . i fallow the bellow link http://www.voip-info.org/wiki/view/Asterisk+H324M http://sip.fontventa.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queue Call Transfer Issue
Hi all ,I am using Asterisk-1.6 and facing a problem during call transfer.when an agent transfer the call to another agent , there is no entry in the queue_log file.is this a problem in Asterisk or i am doing some wrong?? PLEASE HELP... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk appache issue
thanks for reply. how can i give the root permission to apache ? sudo. i also tried sudo . However, without careful configuration you will probably be giving root access to any process that runs as your apache user. I've never done it, but I'm guessing you could create a group, make your asterisk user and your apache user members of that group and protect resources appropriately. What are you trying to accomplish that you can't using AMI, querying a database, creating a call file or parsing a log file? Alternatively, as of 1.6.1 (or is it 1.6.2) you have CLI permissions. You can allow anybody to write to the socket, but only a limited set of commands to the user 'apache' or whatever. See /etc/asterisk/cli_permissions.conf . i am using asterisk 1.6.2 but did't find /etc/asterisk/cli_permissions.conf. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk appache issue
Hi Everyone, I installed Asterisk-1.6 by user root and its working fine. but when i tried to run any asterisk command through apache user it shows Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist? i think its a permission error. how can i give the root permission to apache ? please help !! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
