Re: [asterisk-users] question about CDR
thanks asghar for your help and support and thanks ishfaq 2013/5/9 Asghar Mohammad asghar...@gmail.com 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 -- _ -- 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
Thanks! Matthew and Dan. On Thu, May 9, 2013 at 10:18 PM, Matthew Jordan mjor...@digium.com wrote: 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 -- _ -- 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
We use Zabbix as monitoring tool and SNMP to get statistics and other info from Asterisk. for this you will have to make sure the snmp module for asterisk gets compiled and the Asterisk MIB is used. Regards, Michel. On 09-05-13 21:23, motty cruz 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
Try with http://www.observium.org (Observium). You can customize script to report into Observium's dashboard. Regards, On Fri, May 10, 2013 at 7:55 AM, Michel Verbraak mic...@verbraak.orgwrote: We use Zabbix as monitoring tool and SNMP to get statistics and other info from Asterisk. for this you will have to make sure the snmp module for asterisk gets compiled and the Asterisk MIB is used. Regards, Michel. On 09-05-13 21:23, motty cruz 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 -- - Luis Morales Consultor de Tecnologia Cel: +58(0412)2352745 OpenID: http://lmorales.myopenid.com/ Twitter: @magnadata Linux User ID : 470650 - Empieza por hacer lo necesario, luego lo que es posible... y de pronto estarás haciendo lo imposible Leonardo Da'Vinci - -- _ -- 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] Voicemail send to e-mail
Hi, I want to configurate asterisknow 3.00 to send voicemail to email. I do not know how to setup server to send email.Please help me Regard Boris Rouliane Date: Fri, 10 May 2013 08:10:09 -0430 From: faston...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] monitoring Asterisk 1.8 Try with http://www.observium.org (Observium). You can customize script to report into Observium's dashboard. Regards, On Fri, May 10, 2013 at 7:55 AM, Michel Verbraak mic...@verbraak.org wrote: We use Zabbix as monitoring tool and SNMP to get statistics and other info from Asterisk. for this you will have to make sure the snmp module for asterisk gets compiled and the Asterisk MIB is used. Regards, Michel. On 09-05-13 21:23, motty cruz 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 -- - Luis Morales Consultor de Tecnologia Cel: +58(0412)2352745 OpenID: http://lmorales.myopenid.com/ Twitter: @magnadata Linux User ID : 470650 - Empieza por hacer lo necesario, luego lo que es posible... y de pronto estarás haciendo lo imposible Leonardo Da'Vinci - -- _ -- 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] Asterisk 12 and OPUS Codec
I was wondering if anyone knows if Asterisk 12 will be supporting the OPUS codec, which is part of the WebRTC standard as the default codec. Thank you, -- James Mortensen Project Manager, VoiceCurve, Inc. 866-707-4590 james.morten...@voicecurve.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
Re: [asterisk-users] Asterisk 12 and OPUS Codec
On 13-05-10 02:45 PM, James Mortensen wrote: I was wondering if anyone knows if Asterisk 12 will be supporting the OPUS codec, which is part of the WebRTC standard as the default codec. Doubt it. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] Job Posting
Ntegrated Solutions in Dallas, TX is still looking for voice guy. This position is for US hire only, will not sponsor H1B work visa. http://www.ntegrated.net/careers/ Thanks. JR -- JR Richardson Engineering for the Masses -- _ -- 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] ISP trunk session ID?
Hi folks, What I trying to do here is exactly this: http://www.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/I_sect14_tt599.html My provider given me a Huawei modem which have 2 phone jacks on it, but instead of using it I rather redirect my POTS number to my PBX. I ran into couple of bumps on the road but now it's half-working. I extracted the SIP user, pass, server info from the modem and even managed to put my PBX into the same VLAN they use, on the exact same IP address like the modem but there is 1 problem: It seems this modem also sends some session ID to the ISP's sip server, something what Asterisk doesn't by default. So if I do this: 1, Let the modem register at the sip service (the phone number can be called and ringing out) 2, Disconnect the modem 3, Let the PBX connect to the SIP server 4, PBX accepts the calls 5, About 5-10 minutes later it stops doing it, when I call the number it shows busy (beep, beep, beep), no matter if I restart Asterisk or not it won't work anymore just if I do the same trick again I'm sure the remote SIP server breaks the voip channel or something, it does NOT drop me out tho, my PBX can register any time without problem but no packets will ever come forward me anymore. It's kind of hard to solve this from 1 side. There must be some solution for this. Please help! Thank You, Sergej -- _ -- 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] ISP trunk session ID?
hi, you can try to change sip user agent and sdp session s , owner in sip config same as your phone,s (modem). asterisk by default send user agent = asterisk version , s= asterisk , o= asterisk. some providers are not happy if they see asterisk word :) On Sat, May 11, 2013 at 12:27 AM, Sergej Petrovsky sergej5...@yandex.comwrote: Hi folks, What I trying to do here is exactly this: http://www.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/I_sect14_tt599.html My provider given me a Huawei modem which have 2 phone jacks on it, but instead of using it I rather redirect my POTS number to my PBX. I ran into couple of bumps on the road but now it's half-working. I extracted the SIP user, pass, server info from the modem and even managed to put my PBX into the same VLAN they use, on the exact same IP address like the modem but there is 1 problem: It seems this modem also sends some session ID to the ISP's sip server, something what Asterisk doesn't by default. So if I do this: 1, Let the modem register at the sip service (the phone number can be called and ringing out) 2, Disconnect the modem 3, Let the PBX connect to the SIP server 4, PBX accepts the calls 5, About 5-10 minutes later it stops doing it, when I call the number it shows busy (beep, beep, beep), no matter if I restart Asterisk or not it won't work anymore just if I do the same trick again I'm sure the remote SIP server breaks the voip channel or something, it does NOT drop me out tho, my PBX can register any time without problem but no packets will ever come forward me anymore. It's kind of hard to solve this from 1 side. There must be some solution for this. Please help! Thank You, Sergej -- _ -- 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] Tier 1 Service Providers (ATT, Level 3)
Anyone here using Level 3 or ATT wholesale sip terminations services? I would like to know on any minimums they would require? Also, an idea of how competitive the rates are. I am not asking to disclose your custom rate deck, just a what to expect. Finally, if you guys can PM me contact info to someone from the wholesale department, I would really appreciate it. Kind Regards, Nick. -- _ -- 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] 11.4: no incoming gv/xmpp
I've set up google voice to chat with me: Forwards calls to: me@gmail.com and xmpp: [general] debug=no; Enable debugging (disabled by default). autoprune=yes ; Auto remove users from buddy list. Depending on your ; setup (ie, using your personal Gtalk account for a test) ; you might lose your contacts list. Default is 'no'. autoregister=no ; Auto register users from buddy list. ;collection_nodes=yes ; Enable support for XEP-0248 for use with ; distributed device state. Default is 'no'. ;pubsub_autocreate=yes ; Whether or not the PubSub server supports/is using ; auto-create for nodes. If it is, we have to ; explicitly pre-create nodes before publishing them. ; Default is 'no'. ;auth_policy=accept ; Auto accept users' subscription requests (default). ; Set to deny for auto denial. [gv](!) type=client serverhost=talk.google.com secret=password priority=25 port=5222 usetls=yes usesasl=yes status=available statusmessage=Not available timeout=5 [google1](gv) username=me@gmail.com and xmpp show connections: asterisk*CLI xmpp show connections Jabber Users and their status: [google1] me@gmail.com - Connected But when I call me, nothing rings through to asterisk. i see the call on gv, Any help appreciated. sean -- _ -- 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] ISP trunk session ID?
Sorry to chime in here, is it possible to change the Server: Asterisk , s=Asterisk, and o= within sip.conf? What are the directives exactly please? Thanks in Advance, Nick. On 5/10/13, Asghar Mohammad asghar...@gmail.com wrote: hi, you can try to change sip user agent and sdp session s , owner in sip config same as your phone,s (modem). asterisk by default send user agent = asterisk version , s= asterisk , o= asterisk. some providers are not happy if they see asterisk word :) On Sat, May 11, 2013 at 12:27 AM, Sergej Petrovsky sergej5...@yandex.comwrote: Hi folks, What I trying to do here is exactly this: http://www.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/I_sect14_tt599.html My provider given me a Huawei modem which have 2 phone jacks on it, but instead of using it I rather redirect my POTS number to my PBX. I ran into couple of bumps on the road but now it's half-working. I extracted the SIP user, pass, server info from the modem and even managed to put my PBX into the same VLAN they use, on the exact same IP address like the modem but there is 1 problem: It seems this modem also sends some session ID to the ISP's sip server, something what Asterisk doesn't by default. So if I do this: 1, Let the modem register at the sip service (the phone number can be called and ringing out) 2, Disconnect the modem 3, Let the PBX connect to the SIP server 4, PBX accepts the calls 5, About 5-10 minutes later it stops doing it, when I call the number it shows busy (beep, beep, beep), no matter if I restart Asterisk or not it won't work anymore just if I do the same trick again I'm sure the remote SIP server breaks the voip channel or something, it does NOT drop me out tho, my PBX can register any time without problem but no packets will ever come forward me anymore. It's kind of hard to solve this from 1 side. There must be some solution for this. Please help! Thank You, Sergej -- _ -- 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
