Re: [asterisk-users] Queue announcements and MOH blanking out on calls from PSTN over IAX2
On Sunday 30 Oct 2011, Raj Mathur (राज माथुर) wrote: After looking further, the problem seems to be purely in playing recorded messages over IAX2. Looking at the debug logs on the SIP server (which is playing the recorded messages) shows that it stops playing one of the messages at some point in the flow, and then never plays anything again. This seems to be very similar to: https://issues.asterisk.org/view.php?id=17232 except there is no virtualisation involved in the process -- everything is working on native hardware. It /is/ amd64 Debian Squeeze running on Intel, though. Regards, -- Raj -- Raj Mathurr...@kandalaya.org http://kandalaya.org/ GPG: 78D4 FC67 367F 40E2 0DD5 0FEF C968 D0EF CC68 D17F PsyTrance Chill: http://schizoid.in/ || It is the mind that moves -- _ -- 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] googleapps calendar
Hi Terry I managed to get it working eventually. I think that it may have been a problem with neon , as I downgraded to .25 from .29, removed all modules and make distclean, make install It started working at this point ! What would be really great would be 1) manager events for new / removed calendars 2) manager command to reload / refresh calendars 3) manager events for new / removed events 4) manager events for alarms Julian On 30 October 2011 02:31, Terry Wilson twil...@digium.com wrote: I am trying to get googleapps calendar integrated with my system. However, following all the instructions that I can find it still fails. this is my config file: [myGoogleCal] type=caldav url=https://www.google.com/calendar/dav/myemail/events/ user=myemail secret=mypassword refresh=15 timeframe=60 I just tried with: [calendar4] type = caldav url = https://www.google.com/calendar/dav/m...@mygoogleappsdomain.net/events/ user = m...@mygoogleappsdomain.net secret = mysneakypassword refresh = 15 timeframe = 60 and 'calendar show calendars' shows my calendar as free, and 'calendar show calendar calendar4' shows an upcoming event. I did have to commit a fix where if you don't have a channel set for notification, it would cause a crash. I just committed that fix a couple of seconds ago. So, everything looks to be working fine for me. when I start asterisk, and type calendar show calendars I get genesis2*CLI calendar show calendars Calendar Type Status -- myGoogleCal caldav free however, there are no events in myGoogleCal, and every 15 minutes I get the message Unknown response to CalDAV calendar pug, request REPORT to /calendar/myemail/events/: Could not read status line: connection was closed by server Sounds like a communication issue. Is there a proxy server required to access the outside? Perhaps libneon wasn't compiled with SSL support or something? You could verify that the url is reachable via a web browser (should download a .ics file) or via using a command-line tool on the Asterisk box like 'curl' to test the url, user, and password. -- _ -- 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 -- Julian Lyndon-Smith IT Director, Dot R Limited I don’t care if it works on your machine! We are not shipping your machine!” The kangaroo dances: http://www.youtube.com/watch?v=MAWl5iYOaUg -- _ -- 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] Issues with Xorcom astribanks when running Ubuntu 11.10 (oneiric) as a quemu/libvirt guest
On Sat, Oct 29, 2011 at 08:14:55PM +0200, Eric van der Vlist wrote: Hi, Xorcom astribanks get initialized straight on when using Ubuntu 11.10 packages but I am having a hard time to get the same result running in a qemu/libvirt image. qemu? qemu+kqemu (the kernel module)? kvm? I would expect plain qemu to have pretty bad performance, though I hardly tried to use it lately. Anyway, it should be more than enough for the simple firmware loading step. The first difficulty is that astribanks devices get different usb device ids during their initialisation process, requiring hot plug support. I have figured out how to solve this issue using the technique described in this post : http://www.blogs.uni-osnabrueck.de/rotapken/2011/04/11/how-to-auto-hotplug-usb-devices-to-libvirt-vms/ That doesn't seem to be enough and the initialisation fails with a status 1: Oct 28 18:58:19 asterisk-rg 'xpp_fxloader'[1006]: Trying to find what to do for product e4e4/1160/101, device /dev/bus/usb/001/004 Oct 28 18:58:19 asterisk-rg 'xpp_fxloader'[1010]: Loading firmware '/usr/share/dahdi/USB_FW.hex' into '/dev/bus/usb/001/004' That's good. Oct 28 18:58:23 asterisk-rg 'xpp_fxloader'[1024]: Trying to find what to do for product e4e4/1161/101, device /dev/bus/usb/001/005 Oct 28 18:58:34 asterisk-rg 'xpp_fxloader'[1035]: /usr/sbin/astribank_tool failed with status 1 Seeing that Xorcom requires USB 2.0 Technically the Astribank driver requires USB 2.0 as it was not worth it to adapt it to the maximal URB (messagee) size of 64 byte of USB 1.1. astribank_hexload actually was never adapted to that limitation either. While this may be considered a bug, we hardly needed to load the firmware on USB 1.1 and this gets an earlier and safer fail on most cases. Anyway, astribank_tool does not use large USB messages. I would not expect it to fail on USB 1.1 . The problem is elsewhere. What happens if you manually run: /usr/share/dahdi/xpp_fxloader load #? Or: astribank_tool -D 001/005 -Q If you have dahdi-tools 2.5, you'll need: astribank_tool -D /dev/bun/usb/001/005 -Q and that the current versions of libvirt and qemu in Ubuntu 11.10 emulate USB 1.10 in guests, I have installed Boris Derzhavets' packages: https://launchpad.net/~bderzhavets/+archive/seabios163 and updated my host definition to emulate USB 2.0 but I still have the same issue. Have I missed something? What version of dahdi-tools is it? -- 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 -- 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] Queue announcements and MOH blanking out on calls from PSTN over IAX2
hmmm so IAX channel is playing with you guys. 1- Cant you guys use SIP, does this happen with SIP trunk as well !? 2- Which version of asterisk are there on both servers. 3- See the output of the command core show file versions in your both asterisk servers. Mainly lookout for IAX channel version. Also try enabling IAX debug and paste the output on console. 2011/10/30 Raj Mathur (राज माथुर) r...@linux-delhi.org On Sunday 30 Oct 2011, Raj Mathur (राज माथुर) wrote: After looking further, the problem seems to be purely in playing recorded messages over IAX2. Looking at the debug logs on the SIP server (which is playing the recorded messages) shows that it stops playing one of the messages at some point in the flow, and then never plays anything again. This seems to be very similar to: https://issues.asterisk.org/view.php?id=17232 except there is no virtualisation involved in the process -- everything is working on native hardware. It /is/ amd64 Debian Squeeze running on Intel, though. Regards, -- Raj -- Raj Mathurr...@kandalaya.org http://kandalaya.org/ GPG: 78D4 FC67 367F 40E2 0DD5 0FEF C968 D0EF CC68 D17F PsyTrance Chill: http://schizoid.in/ || It is the mind that moves -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reporting for Asterisk Call Center
Dear; In case I need to retreive the real time data (for example, how many calls currently in the queue, and how many calls currently waiting in the queue, how many agents currently are logged in ... etc). How to get this? Is it using the AGI? From where I can get information about this? Because in the CDR, there is nothing mention or can be obtained for these informations (how many in the queue and how many is waiting .. etc), correct? Thanks for advance. Regards Bilal --- The following script will generate an asterisk database with a table named CDR that will work with asterisk 1.8. Be sure to change 'PASSWORD' with whatever password you want to use. SET SQL_MODE=NO_AUTO_VALUE_ON_ZERO; CREATE DATABASE `asterisk` DEFAULT CHARACTER SET latin1 COLLATE latin1_swedish_ci; USE `asterisk`; CREATE TABLE IF NOT EXISTS `cdr` ( `recid` mediumint(8) unsigned NOT NULL auto_increment COMMENT 'Record ID', `calldate` datetime NOT NULL default '-00-00 00:00:00', `clid` varchar(80) NOT NULL default '', `src` varchar(80) NOT NULL default '', `dst` varchar(80) NOT NULL default '', `dcontext` varchar(80) NOT NULL default '', `channel` varchar(80) NOT NULL default '', `dstchannel` varchar(80) NOT NULL default '', `lastapp` varchar(80) NOT NULL default '', `lastdata` varchar(80) NOT NULL default '', `duration` int(11) NOT NULL default '0', `billsec` int(11) NOT NULL default '0', `disposition` varchar(45) NOT NULL default '', `amaflags` int(11) NOT NULL default '0', `accountcode` varchar(20) NOT NULL default '', `uniqueid` varchar(32) NOT NULL default '', `userfield` varchar(255) NOT NULL default '', PRIMARY KEY (`recid`), KEY `calldate` (`calldate`), KEY `dst` (`dst`), KEY `accountcode` (`accountcode`), KEY `src` (`src`), KEY `disposition` (`disposition`), KEY `uniqueid` (`uniqueid`) ) ENGINE=InnoDB DEFAULT CHARSET=latin1 AUTO_INCREMENT=1 ; CREATE USER 'asterisk'@'localhost' IDENTIFIED BY 'PASSWORD'; GRANT FILE ON * . * TO 'asterisk'@'localhost' IDENTIFIED BY 'PASSWORD' WITH MAX_QUERIES_PER_HOUR 0 MAX_CONNECTIONS_PER_HOUR 0 MAX_UPDATES_PER_HOUR 0 MAX_USER_CONNECTIONS 0 ; GRANT INSERT ON `asterisk`.`cdr` TO 'asterisk'@'localhost'; If you're going to be running the mysql database on the same server as the asterisk box, the following cdr_mysql.conf should also work for 1.8: [global] hostname=localhost dbname=asterisk table=cdr password=PASSWORD user=asterisk port=3306 sock=/var/lib/mysql/mysql.sock userfield=1 loguniqueid=yes -- _ -- 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] Queue announcements and MOH blanking out on calls from PSTN over IAX2
On Sunday 30 Oct 2011, Sammy Govind wrote: hmmm so IAX channel is playing with you guys. 1- Cant you guys use SIP, does this happen with SIP trunk as well !? 2- Which version of asterisk are there on both servers. 3- See the output of the command core show file versions in your both asterisk servers. Mainly lookout for IAX channel version. Also try enabling IAX debug and paste the output on console. 1.6.2.9-2+squeeze3 on the SIP server, 1.6.2.9-2+squeeze1 on the Dial server. I doubt if we'll be able to change the architecture of an infrastructure handling up to 450 simultaneous calls for the past 6 months at this stage, so SIP is out. IAX2 has been working beautifully for our needs up to this point, and we need to find a solution that we can integrate into this architecture itself! Incidentally, if anyone's interested, the installation itself is detailed at: http://www.mail-archive.com/ilugd@lists.linux-delhi.org/msg28166.html Regards, -- Raj -- Raj Mathurr...@kandalaya.org http://kandalaya.org/ GPG: 78D4 FC67 367F 40E2 0DD5 0FEF C968 D0EF CC68 D17F PsyTrance Chill: http://schizoid.in/ || It is the mind that moves -- _ -- 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] Reporting for Asterisk Call Center
On Sunday 30 Oct 2011, bilal ghayyad wrote: In case I need to retreive the real time data (for example, how many calls currently in the queue, and how many calls currently waiting in the queue, how many agents currently are logged in ... etc). How to get this? Is it using the AGI? From where I can get information about this? Because in the CDR, there is nothing mention or can be obtained for these informations (how many in the queue and how many is waiting .. etc), correct? On the command-line you can give: queue show queue-name which will give you real-time information about the queue. Presumably you can do the same through AGI too. In addition, I believe there are some ready-made packages (both FOSS and proprietary) that will display this information nicely formatted. Regards, -- Raj -- Raj Mathurr...@kandalaya.org http://kandalaya.org/ GPG: 78D4 FC67 367F 40E2 0DD5 0FEF C968 D0EF CC68 D17F PsyTrance Chill: http://schizoid.in/ || It is the mind that moves -- _ -- 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] Queue announcements and MOH blanking out on calls from PSTN over IAX2
On Sun, Oct 30, 2011 at 12:04:09PM +0530, Raj Mathur (राज माथुर) wrote: On Sunday 30 Oct 2011, Raj Mathur (राज माथुर) wrote: After looking further, the problem seems to be purely in playing recorded messages over IAX2. Looking at the debug logs on the SIP server (which is playing the recorded messages) shows that it stops playing one of the messages at some point in the flow, and then never plays anything again. This seems to be very similar to: https://issues.asterisk.org/view.php?id=17232 except there is no virtualisation involved in the process -- everything is working on native hardware. It /is/ amd64 Debian Squeeze running on Intel, though. Do you use DAHDI timing? Try 'timing test' in the Asterisk CLI. If so, I wonder if it's possible that the redfone devices may actually have occasional hiccups as a timing source[1]. This should be easily noticable using 'dahdi_test'. Anyway, maybe try a different timing source, by disabling other res_timing*.so modules in modules.conf (and restarting asterisk). [1] Sorry, I'm not familiar with them well enough, and apologize in advance if this suggestion is silly. -- 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 -- 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] r...@linux-delhi.org
Dear Raj; Thanks a lot. Actually I need to do a dash board for reporting, so I beleive the only way is to use the AGI, correct? But where I can find documents or link that can help me to do this? About ur sentence: some ready-made packages (both FOSS and proprietary) that will display this information nicely formatted. What is the FOSS and proprietary? Any link for it? And this ready-made packages can work with asterisk 1.8? Regards Bilal - On the command-line you can give: queue show queue-name which will give you real-time information about the queue. Presumably you can do the same through AGI too. In addition, I believe there are some ready-made packages (both FOSS and proprietary) that will display this information nicely formatted. Regards, -- Raj -- Raj Mathur -- _ -- 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] r...@linux-delhi.org
On Sunday 30 Oct 2011, bilal ghayyad wrote: Actually I need to do a dash board for reporting, so I beleive the only way is to use the AGI, correct? But where I can find documents or link that can help me to do this? About ur sentence: some ready-made packages (both FOSS and proprietary) that will display this information nicely formatted. What is the FOSS and proprietary? Any link for it? And this ready-made packages can work with asterisk 1.8? FOSS is Free and Open Source Software, like Asterisk and Linux; Proprietary is software like Windows, which you cannot distribute and modify. For Queue statistics, http://www.asternic.biz/ has both FOSS and proprietary versions of its package. http://queue-tip.rubyforge.org/ is FOSS. http://sourceforge.net/projects/astacd-activity/ is FOSS. Disclaimer: I haven't used any of these packages, and there must be many more that my quick search didn't find. Regards, -- Raj -- Raj Mathurr...@kandalaya.org http://kandalaya.org/ GPG: 78D4 FC67 367F 40E2 0DD5 0FEF C968 D0EF CC68 D17F PsyTrance Chill: http://schizoid.in/ || It is the mind that moves -- _ -- 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] Issues with Xorcom astribanks when running Ubuntu 11.10 (oneiric) as a quemu/libvirt guest
Tzafrir, Le dimanche 30 octobre 2011 à 10:30 +0200, Tzafrir Cohen a écrit : On Sat, Oct 29, 2011 at 08:14:55PM +0200, Eric van der Vlist wrote: Hi, Xorcom astribanks get initialized straight on when using Ubuntu 11.10 packages but I am having a hard time to get the same result running in a qemu/libvirt image. qemu? qemu+kqemu (the kernel module)? kvm? I would expect plain qemu to have pretty bad performance, though I hardly tried to use it lately. I am using qemu-kvm right now but I am open to other open source alternatives! Anyway, it should be more than enough for the simple firmware loading step. The first difficulty is that astribanks devices get different usb device ids during their initialisation process, requiring hot plug support. I have figured out how to solve this issue using the technique described in this post : http://www.blogs.uni-osnabrueck.de/rotapken/2011/04/11/how-to-auto-hotplug-usb-devices-to-libvirt-vms/ That doesn't seem to be enough and the initialisation fails with a status 1: Oct 28 18:58:19 asterisk-rg 'xpp_fxloader'[1006]: Trying to find what to do for product e4e4/1160/101, device /dev/bus/usb/001/004 Oct 28 18:58:19 asterisk-rg 'xpp_fxloader'[1010]: Loading firmware '/usr/share/dahdi/USB_FW.hex' into '/dev/bus/usb/001/004' That's good. Oct 28 18:58:23 asterisk-rg 'xpp_fxloader'[1024]: Trying to find what to do for product e4e4/1161/101, device /dev/bus/usb/001/005 Oct 28 18:58:34 asterisk-rg 'xpp_fxloader'[1035]: /usr/sbin/astribank_tool failed with status 1 Seeing that Xorcom requires USB 2.0 Technically the Astribank driver requires USB 2.0 as it was not worth it to adapt it to the maximal URB (messagee) size of 64 byte of USB 1.1. astribank_hexload actually was never adapted to that limitation either. While this may be considered a bug, we hardly needed to load the firmware on USB 1.1 and this gets an earlier and safer fail on most cases. Anyway, astribank_tool does not use large USB messages. I would not expect it to fail on USB 1.1 . My assumption that this could be the cause of my issues was based on this blog post : http://www.parnreiter.at/xorcom-astribank.aspx. The error messages I got were very close to those mentioned in that page, especially these ones: astribank_hexload.c:99: ERROR(load_hexfile): Failed hexfile send start: -71 astribank_hexload.c:218: ERROR(main): Loading firmware to FPGA failed 'xpp_fxloader'[23554]: /usr/sbin/astribank_hexload failed with status 1 I am not in front of the server right now and can't test it again, but from memory I *think* that the error message was the same except from the value -71 which was more like -73 in my case. The problem is elsewhere. What happens if you manually run: /usr/share/dahdi/xpp_fxloader load #? Or: astribank_tool -D 001/005 -Q I'll test that as soon as I can! If you have dahdi-tools 2.5, you'll need: astribank_tool -D /dev/bun/usb/001/005 -Q and that the current versions of libvirt and qemu in Ubuntu 11.10 emulate USB 1.10 in guests, I have installed Boris Derzhavets' packages: https://launchpad.net/~bderzhavets/+archive/seabios163 and updated my host definition to emulate USB 2.0 but I still have the same issue. Have I missed something? What version of dahdi-tools is it? 2.4.1, and I see that dahdi-firmware-nonfree (that includes your firmware) is 2.2.1.1-1: vdv@lrt-rg:~$ dpkg -l *dahdi* Souhait=inconnU/Installé/suppRimé/Purgé/H=à garder | État=Non/Installé/fichier-Config/dépaqUeté/échec-conFig/H=semi-installé/W=attend-traitement-déclenchements |/ Err?=(aucune)/besoin Réinstallation (État,Err: majuscule=mauvais) ||/ Nom Version Description +++--- ii asterisk-dahdi 1:1.8.4.4~dfsg-2ubuntu1 DAHDI devices support for the Asterisk PBX ii dahdi1:2.4.1-1ubuntu1 utilities for using the DAHDI kernel modules ii dahdi-dkms 1:2.4.1+dfsg-1ubuntu2DAHDI telephony interface (dkms kernel driver) ii dahdi-firmware-nonfree 2.2.1.1-1DAHDI non-free firmware ii dahdi-linux 1:2.4.1+dfsg-1ubuntu2DAHDI telephony interface - Linux userspace parts un dahdi-source none (aucune description n'est disponible) That being said, the host (in which the firmware loads fine) has exactly the same versions installed. Thanks for your help, Eric -- _ -- 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] googleapps calendar
I managed to get it working eventually. I think that it may have been a problem with neon , as I downgraded to .25 from .29, removed all modules and make distclean, make install It started working at this point ! Good to hear. What would be really great would be 1) manager events for new / removed calendars This wouldn't be too hard to implement, just add some manager_event calls to the appropriate places in build_calendar() in res_calendar.c. 2) manager command to reload / refresh calendars I don't think existing calendars are even really refreshed if you reload via the CLI. If they were, you could just use the AMI command Command to execute the reload via AMI. Perhaps looping through all of the calendars and doing an ast_cond_signal() would kick off the refreshes. I haven't really looked at it too carefully (it's the weekend, after all). The other option is just to set the refresh time to a very small value (if you don't have lots of calendars). 3) manager events for new / removed events If someone was interested in doing it, schedule_calendar_event() in res_calendar.c would be where to add the manager_event calls for this, I think. 4) manager events for alarms This could already be done by setting up normal dialplan handling of calendar notification events and then using the UserEvent application to generate whatever AMI event you wanted. The calendaring stuff wasn't really designed to be a multi-calendar proxy for external applications (via AMI, etc.) but more just giving the PBX access to one's calendars. It would really be better if the multi-protocol calendar stuff was all in a separate app that could communicate bi-directionally with Asterisk for everything if you are looking for that kind of thing. There really weren't Asterisk APIs for everything I wanted to do with calendaring when I wrote it, so I didn't take that route. If I was re-writing it again today, I'd probably try harder to go that route. -- _ -- 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] Meetme does not return back to the dialplan
Hi everyone,
I am trying to get Meetme to return back to the context from where it
joined the meetme. For example a user uses the following context to join a
conference, once user hangs up I would like to continue executing the rest
of the dialplan. But when caller hangs up from the conference I see on CLI
that meetme exited with non-zero status but none of the rest of the
dialplan is executed. Please help. I am using asterisk 1.6.2.20
[default]
exten = _,1,MeetMe(1000,1pdMX)
exten = _,n,noop(returned from meetme) ;After user hangs up should
come here
exten = _,n,SoftHangup(${ORIG_CALLER})
exten = _,n,SoftHangup(${CONF_CALLER})
exten = _,n,Hangup
exten = h,1,noop(default-end)
exten = h,n,SoftHangup(${ORIG_CALLER})
exten = h,n,SoftHangup(${CONF_CALLER})
exten = h,n,Hangup
--
Karim Mardhani
--
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Re: [asterisk-users] Reporting for Asterisk Call Center
Thanks a lot. Really I am trying to know how to do AGI to get the information from asterisk (for example, how to talk with asterisk to know the concurrent calls, or the number of agents in the queue, ... etc)? Where I can find this? Regards Bilal Actually I need to do a dash board for reporting, so I beleive the only way is to use the AGI, correct? But where I can find documents or link that can help me to do this? About ur sentence: some ready-made packages (both FOSS and proprietary) that will display this information nicely formatted. What is the FOSS and proprietary? Any link for it? And this ready-made packages can work with asterisk 1.8? FOSS is Free and Open Source Software, like Asterisk and Linux; Proprietary is software like Windows, which you cannot distribute and modify. For Queue statistics, http://www.asternic.biz/ has both FOSS and proprietary versions of its package. http://queue-tip.rubyforge.org/ is FOSS. http://sourceforge.net/projects/astacd-activity/ is FOSS. Disclaimer: I haven't used any of these packages, and there must be many more that my quick search didn't find. Regards, -- Raj -- -- _ -- 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] Reporting for Asterisk Call Center
Un-top-posting... On Sun, 30 Oct 2011, bilal ghayyad wrote: Really I am trying to know how to do AGI to get the information from asterisk (for example, how to talk with asterisk to know the concurrent calls, or the number of agents in the queue, ... etc)? Where I can find this? An AGI is a program that is executed in the context of a channel -- meaning, during a call. Are you planning on calling into your Asterisk server to trigger the collection of statistics? I guess you could set up a 'cron job' to periodically write a call file to create a channel and execute an AGI, but I suspect you should be looking at AMI. -- 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 -- 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
