[asterisk-users] question about CDR
hello list, i need your help about cdr ,i have installed the module cdr in my asterisk 1.4 . for the inbound calls when i call my sip exten like below : exten = 506,1,Dial(SIP/223, 10) exten = 506,n,Dial(SIP/276, 10) in CDR i have just one line with SIP /276 the last line but there is no historic for the first SIP 223 recid Record ID | calldate |clid |src | dst |dcontext |channel | dstchannel |lastapp |lastdata |duration |billsec |disposition |amaflags |accountcode |uniqueid |3 | 626747 |2013-05-09 09:22:55|0661551203 |0661551203|506 |default |Zap/14-1 |SIP/276-092ac7b0 |Dial|SIP/276| 10|21 |0 |NO ANSWER any help please to have the historic for 223 and 276 thanks and regards -- _ -- 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] question about CDR
On Thu, 2013-05-09 at 08:46 +, Salaheddine Elharit wrote: hello list, i need your help about cdr ,i have installed the module cdr in my asterisk 1.4 . for the inbound calls when i call my sip exten like below : exten = 506,1,Dial(SIP/223, 10) exten = 506,n,Dial(SIP/276, 10) in CDR i have just one line with SIP /276 the last line but there is no historic for the first SIP 223 recid Record ID | calldate |clid |src | dst |dcontext |channel | dstchannel |lastapp |lastdata |duration |billsec |disposition |amaflags |accountcode |uniqueid |3 | 626747 |2013-05-09 09:22:55|0661551203 |0661551203| 506 |default |Zap/14-1 |SIP/276-092ac7b0 |Dial|SIP/276| 10|21 |0 |NO ANSWER any help please to have the historic for 223 and 276 Hi You need to look into Channel Event Logging https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=5242932 Regards Ish -- Ishfaq Malik i...@pack-net.co.uk Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- 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] question about CDR
thanks i verify but i don't understanding if can someone give me an example best regards 2013/5/9 Ishfaq Malik i...@pack-net.co.uk On Thu, 2013-05-09 at 08:46 +, Salaheddine Elharit wrote: hello list, i need your help about cdr ,i have installed the module cdr in my asterisk 1.4 . for the inbound calls when i call my sip exten like below : exten = 506,1,Dial(SIP/223, 10) exten = 506,n,Dial(SIP/276, 10) in CDR i have just one line with SIP /276 the last line but there is no historic for the first SIP 223 recid Record ID | calldate |clid |src | dst |dcontext |channel | dstchannel |lastapp |lastdata |duration |billsec |disposition |amaflags |accountcode |uniqueid |3 | 626747 |2013-05-09 09:22:55|0661551203 |0661551203| 506 |default |Zap/14-1 |SIP/276-092ac7b0 |Dial|SIP/276| 10|21 |0 |NO ANSWER any help please to have the historic for 223 and 276 Hi You need to look into Channel Event Logging https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=5242932 Regards Ish -- Ishfaq Malik i...@pack-net.co.uk Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- 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] question about CDR
hi, asterisk insert cdr when call is hangup and last dial statment, i dont understatnd why you are using 2 dial statment on same extenstion? if you you want dial to both extensions you can use 506,1,Dial(SIP/223SIP/276) if you want dial both same time or if you want to do failover the check Dial status and gotoif dialstatus = NO ANSWER or what ever you need. On Thu, May 9, 2013 at 10:46 AM, Salaheddine Elharit salah.elharit...@gmail.com wrote: hello list, i need your help about cdr ,i have installed the module cdr in my asterisk 1.4 . for the inbound calls when i call my sip exten like below : exten = 506,1,Dial(SIP/223, 10) exten = 506,n,Dial(SIP/276, 10) in CDR i have just one line with SIP /276 the last line but there is no historic for the first SIP 223 recid Record ID | calldate |clid |src | dst |dcontext |channel | dstchannel |lastapp |lastdata |duration |billsec |disposition |amaflags |accountcode |uniqueid |3 | 626747 |2013-05-09 09:22:55|0661551203 |0661551203|506 |default |Zap/14-1 |SIP/276-092ac7b0 |Dial|SIP/276| 10|21 |0 |NO ANSWER any help please to have the historic for 223 and 276 thanks and regards -- _ -- 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] Asterisk and hylafax: how to debug ...
Hi, Am Dienstag, den 07.05.2013, 21:48 +0200 schrieb Sebastian Niehaus: Am 07.05.2013 18:23, schrieb Sebastian Niehaus: For some reason, t38modem tells hylafax the line is BUSY so there is no fax send. Well, I may add the log of t38modem (sorry for the ugly formatting) Parts I consider as most important are: ModemConnection::SetUpConnection dstNum=189659 srcNum=30 srcName=root ... denied (all modems busy) [ snip ] it seems to me, that the call is routed from the modem to the modem (and not to asterisk). t38modem has some config options for call routing. Something like: route=modem:.*=sip:dn@ip:port route=sip:.*=modem:dn The first rule routes calls from the modem to a sip destination. I think in Your setup it should be route=modem:.*=sip:dn@127.0.0.1:5060. (I never used localhost in a setup like this, it should work with the IP of Your ethernet too). HTH, Karsten -- _ -- 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] Get Channel Variables in AMI Event NewExten
I believe you will have to monitor for the Newexten event, then send an AMI Getvar command. It doesn't make sense to pass all the possible channel variables along with a Newexten event. There may be a ton of extra variables that someone may not want or need on the AMI. Better to have them ask for specific variables that are not standard. Action: Getvar ActionID: ValueYouCanIdentify Channel: IAX2/X.X.X.X:4572-5011 Variable: fu_callerid This will result in a response from AMI... Response: Success ActionID: ValueYouCanIdentify Variable: fu_callerid Value: 141688xyxzz The ActionID is very important if you want to watch for an exact response to your request. Dan -- _ -- 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] Elastix vs vicidial
Hi Bilal, if you're looking for Asterisk, CRM and even Google Contacts integration, do check out Aptus FonB (www.aptus.com). I believe that's the exact solution you're looking for. Br, Maax __ Hi; I used vicidial for call center and I would like to try elastix. Can someone advise about the advantages? Does Elastix has a screen for the agent to login/logout from their PC and deal with the inbound/outbound calls and Integrated with the *CRM*? Regards Bilal -- -- _ -- 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] Get Channel Variables in AMI Event NewExten
On 05/09/2013 08:16 AM, Dan Cropp wrote: I believe you will have to monitor for the Newexten event, then send an AMI Getvar command. It doesn’t make sense to pass all the possible channel variables along with a Newexten event. There may be a ton of extra variables that someone may not want or need on the AMI. Better to have them ask for specific variables that are not standard. Action: Getvar ActionID: ValueYouCanIdentify Channel: IAX2/X.X.X.X:4572-5011 Variable: fu_callerid This will result in a response from AMI… Response: Success ActionID: ValueYouCanIdentify Variable: fu_callerid Value: 141688xyxzz The ActionID is very important if you want to watch for an exact response to your request. If you know the names of the channel variables, you can also configure manager to send them with every channel event. From manager.conf: ; ; Display certain channel variables every time a channel-oriented ; event is emitted: ; ;channelvars = var1,var2,var3 So if you want fu_callerid, set: channelvars = fu_callerid And, once that variable is set, you should get a NewExten event, you should see the following key/value pair: ChanVariable(SIP/1234-0001): fu_callerid=foobar -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- 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] chanstats console errors
Running Asterisk 10.12.2 on Debian/sparc i'm doing all sip/rtp. directmedia=yes directrtpsetup=yes I frequently see on the console: WARNING[7832]: chan_sip.c:19134 show_chanstats_cb: Could not get RTP stats What is this error trying to tell me ? 'sip show channelstats' shows all 0s (save Peer/CallID/Duration) I looked for that string in the source but i didnt learn much. -- _ -- 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] monitoring Asterisk 1.8
Hello, i'm looking for suggestions to monitor Asterisk Server? I installed Nagios but no success, I do prefer not to install any web server on the server running Asterisk. Thanks in advance. -Motty -- _ -- 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] Planned maintenance for community services on May 11, 2013
On Saturday, May 11, 2013 , the Asterisk community services listed below may have intermittent availability due to routine maintenance being performed. This maintenance will begin at approximately 12:00 AM CDT (05:00 May 11 UTC)[1] and should last no longer than five hours. The affected services are: * issues.asterisk.org * wiki.asterisk.org * code.asterisk.org * crowd.asterisk.org * bamboo.asterisk.org * signup.asterisk.org * reviewboard.asterisk.org * svn.digium.com / svn.asterisk.org / svncommunity.digium.com * svnview.digium.com * downloads.asterisk.org * downloads.digium.com * packages.asterisk.org * git.asterisk.org Thank you for your support! -- Digium's Asterisk Development Team -- _ -- 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] monitoring Asterisk 1.8
Monitor what parts exactly? Right this moment I'm in the process of installing Munin and the Asterisk plugin to monitor channel usage, SIP connections, and the like. The Munin server is running on a separate machine with just the node software on Asterisk. On Thu, May 9, 2013 at 12:23 PM, motty cruz motty.c...@gmail.com wrote: Hello, i'm looking for suggestions to monitor Asterisk Server? I installed Nagios but no success, I do prefer not to install any web server on the server running Asterisk. Thanks in advance. -Motty -- _ -- 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 -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] monitoring Asterisk 1.8
Thanks for the suggestion Carlos, do you have a HowTo? can you point me to one. I unsuccessfully follow one found using google. I'm using CentOs 6.0 Thanks, Motty On Thu, May 9, 2013 at 12:38 PM, Carlos Alvarez car...@televolve.comwrote: Monitor what parts exactly? Right this moment I'm in the process of installing Munin and the Asterisk plugin to monitor channel usage, SIP connections, and the like. The Munin server is running on a separate machine with just the node software on Asterisk. On Thu, May 9, 2013 at 12:23 PM, motty cruz motty.c...@gmail.com wrote: Hello, i'm looking for suggestions to monitor Asterisk Server? I installed Nagios but no success, I do prefer not to install any web server on the server running Asterisk. Thanks in advance. -Motty -- _ -- 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 -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] monitoring Asterisk 1.8
I'm using opennms and It's working fine. On Thu, May 9, 2013 at 3:23 PM, motty cruz motty.c...@gmail.com wrote: Hello, i'm looking for suggestions to monitor Asterisk Server? I installed Nagios but no success, I do prefer not to install any web server on the server running Asterisk. Thanks in advance. -Motty -- _ -- 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] monitoring Asterisk 1.8
http://opennms.org/wiki/Installation:Yum On Thu, May 9, 2013 at 4:03 PM, Carlos Rojas crt.ro...@gmail.com wrote: I'm using opennms and It's working fine. On Thu, May 9, 2013 at 3:23 PM, motty cruz motty.c...@gmail.com wrote: Hello, i'm looking for suggestions to monitor Asterisk Server? I installed Nagios but no success, I do prefer not to install any web server on the server running Asterisk. Thanks in advance. -Motty -- _ -- 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] monitoring Asterisk 1.8
It's not quick or simple, but there's decent documentation. I haven't been saving the links I used, so I can't just give you specific places to look, other than the best Asterisk plugin: https://github.com/munin-monitoring/contrib/blob/master/plugins/asterisk/asterisk TIP: Use chmod 755 on the plugin files after you install them. As to installing Munin itself, just start from their web site and get that running. You will then install the Asterisk plugin, create an AMI user for the plugin to connect to, and set the parameters for the plugin to the server IP and AMI account you just created. Right now I'm working on being able to monitor the servers without installing the plugin on the Asterisk box. This will give Asterisk stats only, but no server stats. Again, what specific things do you want to monitor? On Thu, May 9, 2013 at 12:53 PM, motty cruz motty.c...@gmail.com wrote: Thanks for the suggestion Carlos, do you have a HowTo? can you point me to one. I unsuccessfully follow one found using google. I'm using CentOs 6.0 Thanks, Motty On Thu, May 9, 2013 at 12:38 PM, Carlos Alvarez car...@televolve.comwrote: Monitor what parts exactly? Right this moment I'm in the process of installing Munin and the Asterisk plugin to monitor channel usage, SIP connections, and the like. The Munin server is running on a separate machine with just the node software on Asterisk. On Thu, May 9, 2013 at 12:23 PM, motty cruz motty.c...@gmail.com wrote: Hello, i'm looking for suggestions to monitor Asterisk Server? I installed Nagios but no success, I do prefer not to install any web server on the server running Asterisk. Thanks in advance. -Motty -- _ -- 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 -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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 -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] monitoring Asterisk 1.8
Thanks for your help; I just want to monitor the queue, calls on hold average time, incoming out going call, I only want to monitor Asterisk, not the server Asterisk in running on. thanks, -Motty On Thu, May 9, 2013 at 1:06 PM, Carlos Rojas crt.ro...@gmail.com wrote: http://opennms.org/wiki/Installation:Yum On Thu, May 9, 2013 at 4:03 PM, Carlos Rojas crt.ro...@gmail.com wrote: I'm using opennms and It's working fine. On Thu, May 9, 2013 at 3:23 PM, motty cruz motty.c...@gmail.com wrote: Hello, i'm looking for suggestions to monitor Asterisk Server? I installed Nagios but no success, I do prefer not to install any web server on the server running Asterisk. Thanks in advance. -Motty -- _ -- 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 -- _ -- 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] monitoring Asterisk 1.8
Then you want a queue manager and reporting tool. Usually when people say monitor Asterisk is has to do with the state of the system itself. You should look at http://www.asternic.net and similar products. Munin will tell you channels in use, but not the other stuff you want. On Thu, May 9, 2013 at 1:12 PM, motty cruz motty.c...@gmail.com wrote: Thanks for your help; I just want to monitor the queue, calls on hold average time, incoming out going call, I only want to monitor Asterisk, not the server Asterisk in running on. thanks, -Motty On Thu, May 9, 2013 at 1:06 PM, Carlos Rojas crt.ro...@gmail.com wrote: http://opennms.org/wiki/Installation:Yum On Thu, May 9, 2013 at 4:03 PM, Carlos Rojas crt.ro...@gmail.com wrote: I'm using opennms and It's working fine. On Thu, May 9, 2013 at 3:23 PM, motty cruz motty.c...@gmail.com wrote: Hello, i'm looking for suggestions to monitor Asterisk Server? I installed Nagios but no success, I do prefer not to install any web server on the server running Asterisk. Thanks in advance. -Motty -- _ -- 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 -- _ -- 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 -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] monitoring Asterisk 1.8
You can use queue-stats http://www.asternic.org/stats/demo/ they has a free version On Thu, May 9, 2013 at 4:12 PM, motty cruz motty.c...@gmail.com wrote: Thanks for your help; I just want to monitor the queue, calls on hold average time, incoming out going call, I only want to monitor Asterisk, not the server Asterisk in running on. thanks, -Motty On Thu, May 9, 2013 at 1:06 PM, Carlos Rojas crt.ro...@gmail.com wrote: http://opennms.org/wiki/Installation:Yum On Thu, May 9, 2013 at 4:03 PM, Carlos Rojas crt.ro...@gmail.com wrote: I'm using opennms and It's working fine. On Thu, May 9, 2013 at 3:23 PM, motty cruz motty.c...@gmail.com wrote: Hello, i'm looking for suggestions to monitor Asterisk Server? I installed Nagios but no success, I do prefer not to install any web server on the server running Asterisk. Thanks in advance. -Motty -- _ -- 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 -- _ -- 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
[asterisk-users] DID providers
Howdy, Looking to port numbers that we currently own in the US Virgin Islands to *any* carrier that can do it in the states. XO says it is out of their scope. Our normal DID carrier (IP Comms) apparently uses XO... Looking for recommendations? Thanks, j -- _ -- 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] monitoring Asterisk 1.8
Queuemetrics works well for this also, and can be installed on a separate machine/VM. www.queuemetrics.com Bruce From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of motty cruz Sent: Thursday, May 09, 2013 3:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] monitoring Asterisk 1.8 Thanks for your help; I just want to monitor the queue, calls on hold average time, incoming out going call, I only want to monitor Asterisk, not the server Asterisk in running on. thanks, -Motty On Thu, May 9, 2013 at 1:06 PM, Carlos Rojas crt.ro...@gmail.commailto:crt.ro...@gmail.com wrote: http://opennms.org/wiki/Installation:Yum On Thu, May 9, 2013 at 4:03 PM, Carlos Rojas crt.ro...@gmail.commailto:crt.ro...@gmail.com wrote: I'm using opennms and It's working fine. On Thu, May 9, 2013 at 3:23 PM, motty cruz motty.c...@gmail.commailto:motty.c...@gmail.com wrote: Hello, i'm looking for suggestions to monitor Asterisk Server? I installed Nagios but no success, I do prefer not to install any web server on the server running Asterisk. Thanks in advance. -Motty -- _ -- 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 -- _ -- 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] monitoring Asterisk 1.8
There is nagios plugin check_asterisk_channels Examples: Check channels/calls, with no concern about limits. check_asterisk_channels Check channels/calls. Issue a warning if there are more than 10 active channels, and a critical if there are more than 15 active channels. check_asterisk_channels -w 10 -c 15 Caveats: This plugin calls the asterisk executable directly, so make sure that the user executing this script has appropriate permissions! Usually the asterisk binary can only be run by the asterisk user or root. To grant the nagios user permissions to execute the script, try something like the following in your /etc/sudoers file: nagios ALL=(ALL) NOPASSWD: /path/to/plugins/directory/check_asterisk_channels You can easily edit this to add more monitoring Jai Rangi On Thu, May 9, 2013 at 12:23 PM, motty cruz motty.c...@gmail.com wrote: Hello, i'm looking for suggestions to monitor Asterisk Server? I installed Nagios but no success, I do prefer not to install any web server on the server running Asterisk. Thanks in advance. -Motty -- _ -- 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] Elastix vs vicidial
thank you for your question because i research some topics about call center with asterisk can you please give a tutorial and the method that you use to implement this call center with vividial . 2013/5/9 Maaz Bin Mahmood pointed@gmail.com Hi Bilal, if you're looking for Asterisk, CRM and even Google Contacts integration, do check out Aptus FonB (www.aptus.com). I believe that's the exact solution you're looking for. Br, Maax __ Hi; I used vicidial for call center and I would like to try elastix. Can someone advise about the advantages? Does Elastix has a screen for the agent to login/logout from their PC and deal with the inbound/outbound calls and Integrated with the *CRM*? Regards Bilal -- -- _ -- 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 -- * **Élève Ingénieur INE2 à l'Institut National des Postes et Télécommunications * *INPT - Rabat - Maro*c * * * * *Responsable de la cellule Asterisk au **Club Electronique et Systemes Embarqués de l'INPT* *Membre du projet ilearn, SIFE INPT* * * * Tel : +212642398782 * -- _ -- 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] qualify=yes: OPTIONS: How to Change?: `From: asterisk`
My Google-Fu skills have failed me, I have not been able to find a solution to the problem I am facing. asterisk + from + asterisk + options + qualify != what I am looking for -- When qualify is enabled on a trunk, the From line shows asterisk. See the SIP message below. I would like to keep qualify enabled without sending the other end any reference to asterisk. Can anyone point me to a setting that will change or remove `²asterisk²` from `FROM:` in the OPTIONS message? Thanks, Brian LaVallee -- /etc/asterisk/sip.conf (Asterisk 1.8.15-cert1) [general] ; - Truncated [TRUNK] ; - Truncated qualify=yes ; ; end -- IP 4.4.4.4.sip 3.3.3.3.sip: UDP, length 573 OPTIONS sip:server.carrier.tld SIP/2.0 Via: SIP/2.0/UDP 4.4.4.4:5060;branch=aBcDeFgHiJkLmNo;rport Max-Forwards: 70 From: asterisk sip:accountid@4.4.4.4;tag=as1832334c To: sip:server.carrier.tld Contact: sip:accountid@4.4.4.4:5060 Call-ID: f80a4ad87fee7c9fdc19b7769495fdb5@4.4.4.4:5060 CSeq: 102 OPTIONS Date: Thu, 09 May 2013 07:22:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 -- _ -- 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] qualify=yes: OPTIONS: How to Change?: `From: asterisk`
On 5/9/13 8:21 PM, Brian LaVallee wrote: When qualify is enabled on a trunk, the From line shows asterisk. See the SIP message below. I had the same annoyance/issue. fixed it in https://issues.asterisk.org/jira/browse/ASTERISK-17616 the patch was included in 1.8.9 rc1. -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] qualify=yes: OPTIONS: How to Change?: `From: asterisk`
On Thursday, May 09, 2013 8:23 PM, Jeremy Kister wrote: On 5/9/13 8:21 PM, Brian LaVallee wrote: When qualify is enabled on a trunk, the From line shows asterisk. See the SIP message below. I had the same annoyance/issue. fixed it in https://issues.asterisk.org/jira/browse/ASTERISK-17616 the patch was included in 1.8.9 rc1. Interesting. I hadn't noticed this bug or its inclusion into 1.8.x. IIRC, pretty sure I worked around this myself in the past by setting a global callerid= value in sip.conf, so if you have a good (!) reason not to upgrade, the OP might give that a shot. -- Nathan Anderson First Step Internet, LLC nath...@fsr.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] Thanks! qualify=yes: OPTIONS: How to Change?: `From: asterisk`
Thanks Jeremy! On 5/9/13 8:21 PM, Brian LaVallee wrote: When qualify is enabled on a trunk, the From line shows asterisk. See the SIP message below. I had the same annoyance/issue. fixed it in https://issues.asterisk.org/jira/browse/ASTERISK-17616 That's looks like the problem I was seeing. the patch was included in 1.8.9 rc1. I've been trying to stick to the current AsteriskNow as my base standard, hopefully the fix was applied to the latest version. Thanks again! Brian LaVallee -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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
