[asterisk-users] Call Parking
Hey, I am configuring call parking on asterisk however I have the following issue, - Caller A calls caller B - Caller B receives the call and transfers it to the parking extension 799, so that caller C can receive the call -*The extension to which the call has been parked is read out to caller A (who did not park the call) instead of caller B (who parked the call) : *this is the problem.* How can I solve that? I have tried all the options of T, t and r however in vain. *Here is my config in extensions.conf for call parking* ;Parking calls for office A exten = 799,1,Set(CHANNEL(parkinglot)=parkinglot_main) exten = 799,n,Set(_PARKINGLOT=parkinglot_main) exten = 799,n,Answer() exten = 799,n,Park() exten = 799,n,Hangup() [macro-officeA] exten = s,1,Dial(${ARG1},10,Ttr) exten = s,n,GotoIf($[${DIALSTATUS} = BUSY]?busy:unavail) exten = s,n(unavail),Voicemail(${MACRO_EXTEN}@officea-vmail,u) exten = s,n,Hangup() exten = s,n(busy),VoiceMail(${MACRO_EXTEN}@officea-vmail,b) exten = s,n,Hangup() exten = 1234,1,MeetMe(1234,i) *Features.conf has this:* [parkinglot_main] context = officeA parkext = 799 parkpos = 800-850 findslot = next Thanks Richard Zulu Twitter www.twitter.com/richardzulu http://www.linkedin.com/in/richardzulu Skype: zulu.richard -- _ -- 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] Custom Dialplan
Hallo Barry, extensions_additional.conf is supposed to be edited by FreePBX. Gopal, on using extensions_custom, the SIP phones work however the details are not captured in the reporting mechanism of FreePBX, which is what I need most. Richard Zulu Twitter www.twitter.com/richardzulu http://www.linkedin.com/in/richardzulu Skype: zulu.richard * * *There is no place like 127.0.0.1* On Mon, Aug 8, 2011 at 2:03 PM, Gopal krishnan gopalakrishnan...@gmail.comwrote: Use extensions_custom.conf file to update your custom configurations. On Sun, Aug 7, 2011 at 3:59 AM, Barry L. Kline blkl...@attglobal.netwrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 08/05/2011 04:32 AM, Richard Zulu wrote: I would like to import my dialplan into freepbx+asterisk since I am switching to that...how can I create my own custom dialplan in freepbx? I'm not sure why you'd want to... freepbx is anathema to custom dialplans. That said, I believe you end up naming your extensions.conf file to extensions_additional.conf and freepbx will pick it up when it starts. It's been a long, long time since I've dealt with freepbx -- in fact I went the other way: from freepbx+asterisk to pure asterisk. When I was using freepbx that was the solution you seek. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFOPcAxCFu3bIiwtTARAkjKAKCPCgcoaRyPNs7BXhge7xxcy7C2qQCdF6hx 2Bwz/YEUSbKFsfzD9V0xX6Q= =W2Dn -END PGP SIGNATURE- -- _ -- 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] Asterisk reporting
Hallo, I have a production asterisk server running on Ubuntu however all my configs where done using the CLI. I would like to implement a reporting element into the server so I can know the number of calls made, for what duration, on what dates. What tool can I use that can fit within any already laid out dialplan? Thanks Richard Zulu Twitter www.twitter.com/richardzulu http://www.linkedin.com/in/richardzulu Skype: zulu.richard * * *There is no place like 127.0.0.1* On Thu, Aug 11, 2011 at 4:12 AM, neo haux neo.h...@gmx.com wrote: Hi I want to change my old answering phone machine and two wireless phones with asterisk box + degium TDM400p (3 fxs+1FXO)+ one LAN phone(Aastra Nortel 9133i) + Wifi/SIP phone I am wondering if I´ll lost actual functionalities that are present in my old answering machine: 1) is it possible to show the caller number (coming from PSTN/FXO) in both SIP phones (wifi/SIP and LAN phone) ? Does SIP protocol take in charge this functionality 2) Most important question is : can I see on those internal phones (Wifi/SIP phone and LAN phone) that I´ve some recoded messages on asterisk. Indeed, I have this fucntionality with my old answering machine where I can see the number of new messages recorded in a big LCD screen. Thx -- _ -- 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] Custom Dialplan
Hey, I have been using asterisk on slackware and had thus come up with my own dialplan. I would like to import my dialplan into freepbx+asterisk since I am switching to that...how can I create my own custom dialplan in freepbx? Thanks Richard Zulu Twitter www.twitter.com/richardzulu Skype: zulu.richard * * *There is no place like 127.0.0.1* -- _ -- 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] Segmentation Fault
Hey, I have installed Asterisk 1.8 on slackware 13.1, php, mysql and apache. I am later to install freepbx to help with reporting on VOIP activity. However, after installing asterisk, I am getting a segment fault. My log file shows this: darkstar kernel: asterisk[2660]: segfault at 81c4f ip 77514810 sp 7fffcd48 error 4 in libc-2.11.1.so[77492000+16b000] I have used gdb so that I can perform a backtrace however the program executes and exits without a stack thus not helpful. Any help is appreciated! Richard Zulu Twitter www.twitter.com/richardzulu Skype: zulu.richard * * *There is no place like 127.0.0.1* -- _ -- 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] Monitor Asterisk and Ast-gui
Hey, I have installed asterisk 1.8 on Slackware 13.1 from source and it is working well. I have 300 ip phones in a natted environment and my asterisk server has a public IP I would love to monitor my SIP activity on my VOIP Server, statistics like amount of sip traffic, who made what call and to whom, how many calls were made in a month, how many ip phones are up and running, which sip phone has made most calls among others. How best can I do that? On the other hand, I have also tried installing ast-gui onto asterisk 1.8, it has installed well but it however keeps looping whenever i try to login in, it says checking permissions on gui folder and loops. Haven't found much help on other mailing lists, any direction given in welcome. Thanks Richard Zulu Twitter www.twitter.com/richardzulu Skype: zulu.richard * * *There is no place like 127.0.0.1* -- _ -- 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] Monitor asterisk
Hallo Keane, I truly have a nagios server, up and running 24/7 -- Richard Zulu Managing Director Time Information Company P.O Box 31842 Clock Tower Kampala, Uganda www.time.co.ug Mobile :+256752624006 Skype: zulu.richard -- _ -- 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] Monitor asterisk
Thanks Nasri, I don't want to only be able to use the CLI because I need the Helpdesk and application support Unit to be able to monitor, and they are not all the techy with CLI and stuff.. On Sun, Aug 8, 2010 at 5:00 AM, Nasir Iqbal na...@ictinnovations.comwrote: Hi following asterisk cli commands can help show channels, show uptime and show sysinfo here is an example asterisk -x core show sysinfo On Sun, Aug 8, 2010 at 12:25 AM, Richard Zulu richard.z...@time.co.ugwrote: Hey guys, I have my asterisk box running without a gui. I now need to monitor usage, calls, traffic of voice calls on this asterisk server. I cannot now install a gui because the configs will be wiped out, how can i go about monitoring all the above? -- Richard Zulu Managing Director Time Information Company P.O Box 31842 Clock Tower Kampala, Uganda www.time.co.ug Mobile :+256752624006 Skype: zulu.richard -- _ -- 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 -- Nasir Iqbal ICT Innovations http://www.ictinnovations.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 -- Richard Zulu Managing Director Time Information Company P.O Box 31842 Clock Tower Kampala, Uganda www.time.co.ug Mobile :+256752624006 Skype: zulu.richard -- _ -- 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] Monitor asterisk
Hey guys, I have my asterisk box running without a gui. I now need to monitor usage, calls, traffic of voice calls on this asterisk server. I cannot now install a gui because the configs will be wiped out, how can i go about monitoring all the above? -- Richard Zulu Managing Director Time Information Company P.O Box 31842 Clock Tower Kampala, Uganda www.time.co.ug Mobile :+256752624006 Skype: zulu.richard -- _ -- 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