[Asterisk-Users] call monitoring and indications / beeps
Hi All, Is it possible to configure asterisk to play a beep at a regular interval when a conversation is being recorded / monitored? There are a number of ways indicating to a user that a conversation is being recorded, one is to play an announcement, another accepted way is to play these beeps at a regular interval (15 / 30 seconds or similar) however i cannot seem to find a way to get them to play when monitoring a call - any ideas? Cheers, Ben ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Monitoring / Call Takeover with Asterisk
Well, the Flash Operator Panel supports barge in too, with the option to barge muted so the people involved in the conversation won't notice the interruption. And then the supervisor can drop one of the channels or mute/unumute them. But it uses meetme, as well as all the other manager applications that supports this. VICIDIAL supports this when using VICIDIAL for inbound and/or outbound calling. Blind monitoring, barging in on the call and hijacking the customer from the agent. I have been doing some work with the Asterisk Management API and there is a commadn where you can transfer a call. This is what you may be looking for Not sure, trying to be as helpful as I can -Original Message- From: Cory Andrews [mailto:[EMAIL PROTECTED] Sent: Tue 3/28/2006 9:59 PM To: asterisk-users@lists.digium.com Cc: Subject: [Asterisk-Users] Call Monitoring / Call Takeover with Asterisk Does Asterisk support, in a call center type environment, the ability for a supervisor to monitor a call between a system user and a 3rd party, and allow them to physically take over the call. For instance if a call center supervisor is randomlay monitoring agent calls, and for some reason need to intervene on a call without first having been conferenced into the call? -- Nicolás Gudiño Buenos Aires - Argentina ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Monitoring / Call Takeover with Asterisk
Does Asterisk support, in a call center type environment, the ability for a supervisor to monitor a call between a system user and a 3rd party, and allow them to physically take over the call. For instance if a call center supervisor is randomlay monitoring agent calls, and for some reason need to intervene on a call without first having been conferenced into the call? Cory J AndrewsVOIPSupply.com454 Sonwil DriveBuffalo, NY 14225++voice - 716.630.1555 X22email - [EMAIL PROTECTED]AIM - B2CORY ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call Monitoring / Call Takeover with Asterisk
I do not think so but it would be a great feature. -Original Message- From: Cory Andrews [mailto:[EMAIL PROTECTED] Sent: Tue 3/28/2006 9:59 PM To: asterisk-users@lists.digium.com Cc: Subject: [Asterisk-Users] Call Monitoring / Call Takeover with Asterisk Does Asterisk support, in a call center type environment, the ability for a supervisor to monitor a call between a system user and a 3rd party, and allow them to physically take over the call. For instance if a call center supervisor is randomlay monitoring agent calls, and for some reason need to intervene on a call without first having been conferenced into the call? Cory J Andrews VOIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 ++ voice - 716.630.1555 X22 email - [EMAIL PROTECTED] AIM - B2CORY winmail.dat___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Monitoring / Call Takeover with Asterisk
http://www.voip-info.org/wiki-Asterisk+manager+API I have been doing some work with the Asterisk Management API and there is a commadn where you can transfer a call. This is what you may be looking for Not sure, trying to be as helpful as I can On 3/29/06, Steve Totaro [EMAIL PROTECTED] wrote: I do not think so but it would be a great feature. -Original Message- From: Cory Andrews [mailto:[EMAIL PROTECTED] Sent: Tue 3/28/2006 9:59 PM To: asterisk-users@lists.digium.com Cc: Subject: [Asterisk-Users] Call Monitoring / Call Takeover with Asterisk Does Asterisk support, in a call center type environment, the ability for a supervisor to monitor a call between a system user and a 3rd party, and allow them to physically take over the call. For instance if a call center supervisor is randomlay monitoring agent calls, and for some reason need to intervene on a call without first having been conferenced into the call? Cory J Andrews VOIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 ++ voice - 716.630.1555 X22 email - [EMAIL PROTECTED] AIM - B2CORY ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Devraj Mukherjee Eternity Technologies Pty Limited ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Monitoring / Call Takeover with Asterisk
Hello, VICIDIAL supports this when using VICIDIAL for inbound and/or outbound calling. Blind monitoring, barging in on the call and hijacking the customer from the agent. MATT--- On 3/28/06, Devraj Mukherjee [EMAIL PROTECTED] wrote: http://www.voip-info.org/wiki-Asterisk+manager+API I have been doing some work with the Asterisk Management API and there is a commadn where you can transfer a call. This is what you may be looking for Not sure, trying to be as helpful as I can On 3/29/06, Steve Totaro [EMAIL PROTECTED] wrote: I do not think so but it would be a great feature. -Original Message- From: Cory Andrews [mailto:[EMAIL PROTECTED] Sent: Tue 3/28/2006 9:59 PM To: asterisk-users@lists.digium.com Cc: Subject: [Asterisk-Users] Call Monitoring / Call Takeover with Asterisk Does Asterisk support, in a call center type environment, the ability for a supervisor to monitor a call between a system user and a 3rd party, and allow them to physically take over the call. For instance if a call center supervisor is randomlay monitoring agent calls, and for some reason need to intervene on a call without first having been conferenced into the call? Cory J Andrews VOIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 ++ voice - 716.630.1555 X22 email - [EMAIL PROTECTED] AIM - B2CORY ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Devraj Mukherjee Eternity Technologies Pty Limited ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Monitoring?
You could use contexts for this. By default put everyone into the 'internal' context. Managers would go into the 'managers' context, which would include the 'internal' context. The manager context specifically would have the exten's to monitor or barge into calls. By including the internal context, they'd have the same dialplan otherwise. You determine which context a user gets by default in sip.conf (if you're using sip phones..). On 3/23/06, Charles Marcus [EMAIL PROTECTED] wrote: 1. Is Asterisk capable of allowing for setting up Groups so that only one extension in a Group can selectively monitor one of the other extensions in the Group (but none of the others can initiate it)? This would be for Managers to listen to Sales Calls of other members of their Team, to provide feedback to the Rep for training purposes. 2. Alternatively, can a Group be defined that will allow multiple extensions to listen in on another call in progress? Again, we want to use this kind of functionality to do some Sales Technique Training calls. -- Best regards, Charles ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Monitoring?
1. Is Asterisk capable of allowing for setting up Groups so that only one extension in a Group can selectively monitor one of the other extensions in the Group (but none of the others can initiate it)? This would be for Managers to listen to Sales Calls of other members of their Team, to provide feedback to the Rep for training purposes. 2. Alternatively, can a Group be defined that will allow multiple extensions to listen in on another call in progress? Again, we want to use this kind of functionality to do some Sales Technique Training calls. -- Best regards, Charles ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call monitoring from 3th phone
I can't say for sure that it's 10.. but it's somewhere between 8 and 13 as I hit * to cycle.. when I get up in that range... it will stop spying.. and asterisk will stop taking calls until I do a restart. On 1/5/06, Tom Vile [EMAIL PROTECTED] wrote: I have not had that issue. Are you saying 10 concurrent channels being spied on or after the 10th it starts to crash? On 1/5/06, Matt [EMAIL PROTECTED] wrote: I've found that chanspy crashes asterisk after about 10 channel spys.. asterisk just stops responding, and I have to restart it. On 1/4/06, Tom Vile [EMAIL PROTECTED] wrote: correct it only works with bridged calls. On 1/4/06, Leo Ann Boon [EMAIL PROTECTED] wrote: Tom Vile wrote: use chanspy or zapbarge That slipped my mind :). Had always been using the conf method since pre 1.0. Does app_chanspy work with reinvite=yes? I understand it only works with bridged calls. On 1/4/06, Leo Ann Boon [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] wrote: is it possible only monitoring call between phone A and B from phone C? I think you want to do service observation? You can do the following: a. Use a 'stealth' meetme conference room say 1234 that doesn't need PIN to log in and also doesn't play a tone on entry/exit (may not be legal in your country). b. Use manager API to redirect 'A' and 'B' to the conference room. c. 'C' joins the conference room with the mute option. d. C will now be able to hear what A and B are saying. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call monitoring from 3th phone
I've found that chanspy crashes asterisk after about 10 channel spys.. asterisk just stops responding, and I have to restart it. On 1/4/06, Tom Vile [EMAIL PROTECTED] wrote: correct it only works with bridged calls. On 1/4/06, Leo Ann Boon [EMAIL PROTECTED] wrote: Tom Vile wrote: use chanspy or zapbarge That slipped my mind :). Had always been using the conf method since pre 1.0. Does app_chanspy work with reinvite=yes? I understand it only works with bridged calls. On 1/4/06, Leo Ann Boon [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] wrote: is it possible only monitoring call between phone A and B from phone C? I think you want to do service observation? You can do the following: a. Use a 'stealth' meetme conference room say 1234 that doesn't need PIN to log in and also doesn't play a tone on entry/exit (may not be legal in your country). b. Use manager API to redirect 'A' and 'B' to the conference room. c. 'C' joins the conference room with the mute option. d. C will now be able to hear what A and B are saying. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call monitoring from 3th phone
I have not had that issue. Are you saying 10 concurrent channels being spied on or after the 10th it starts to crash? On 1/5/06, Matt [EMAIL PROTECTED] wrote: I've found that chanspy crashes asterisk after about 10 channel spys.. asterisk just stops responding, and I have to restart it. On 1/4/06, Tom Vile [EMAIL PROTECTED] wrote: correct it only works with bridged calls. On 1/4/06, Leo Ann Boon [EMAIL PROTECTED] wrote: Tom Vile wrote: use chanspy or zapbarge That slipped my mind :). Had always been using the conf method since pre 1.0. Does app_chanspy work with reinvite=yes? I understand it only works with bridged calls. On 1/4/06, Leo Ann Boon [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] wrote: is it possible only monitoring call between phone A and B from phone C? I think you want to do service observation? You can do the following: a. Use a 'stealth' meetme conference room say 1234 that doesn't need PIN to log in and also doesn't play a tone on entry/exit (may not be legal in your country). b. Use manager API to redirect 'A' and 'B' to the conference room. c. 'C' joins the conference room with the mute option. d. C will now be able to hear what A and B are saying. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call monitoring from 3th phone
is it possible only monitoring call between phone A and B from phone C? -- [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call monitoring from 3th phone
[EMAIL PROTECTED] wrote: is it possible only monitoring call between phone A and B from phone C? I think you want to do service observation? You can do the following: a. Use a 'stealth' meetme conference room say 1234 that doesn't need PIN to log in and also doesn't play a tone on entry/exit (may not be legal in your country). b. Use manager API to redirect 'A' and 'B' to the conference room. c. 'C' joins the conference room with the mute option. d. C will now be able to hear what A and B are saying. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call monitoring from 3th phone
use chanspy or zapbarge On 1/4/06, Leo Ann Boon [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] wrote: is it possible only monitoring call between phone A and B from phone C? I think you want to do service observation? You can do the following: a. Use a 'stealth' meetme conference room say 1234 that doesn't need PIN to log in and also doesn't play a tone on entry/exit (may not be legal in your country). b. Use manager API to redirect 'A' and 'B' to the conference room. c. 'C' joins the conference room with the mute option. d. C will now be able to hear what A and B are saying. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call monitoring from 3th phone
Tom Vile wrote: use chanspy or zapbarge That slipped my mind :). Had always been using the conf method since pre 1.0. Does app_chanspy work with reinvite=yes? I understand it only works with bridged calls. On 1/4/06, Leo Ann Boon [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] wrote: is it possible only monitoring call between phone A and B from phone C? I think you want to do service observation? You can do the following: a. Use a 'stealth' meetme conference room say 1234 that doesn't need PIN to log in and also doesn't play a tone on entry/exit (may not be legal in your country). b. Use manager API to redirect 'A' and 'B' to the conference room. c. 'C' joins the conference room with the mute option. d. C will now be able to hear what A and B are saying. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call monitoring from 3th phone
correct it only works with bridged calls. On 1/4/06, Leo Ann Boon [EMAIL PROTECTED] wrote: Tom Vile wrote: use chanspy or zapbarge That slipped my mind :). Had always been using the conf method since pre 1.0. Does app_chanspy work with reinvite=yes? I understand it only works with bridged calls. On 1/4/06, Leo Ann Boon [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] wrote: is it possible only monitoring call between phone A and B from phone C? I think you want to do service observation? You can do the following: a. Use a 'stealth' meetme conference room say 1234 that doesn't need PIN to log in and also doesn't play a tone on entry/exit (may not be legal in your country). b. Use manager API to redirect 'A' and 'B' to the conference room. c. 'C' joins the conference room with the mute option. d. C will now be able to hear what A and B are saying. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Monitoring / Ext to Ext with Sipura-841
Hi All, I am using Asterisk 1.2 with 10 Sipura-841 phones. Outgoing and incoming calls sound great. However, extension to extension calls are really loud with a lot of background noise picked up. Also, the same issue exists when using 888 to barge in and monitor calls. I've been through the configuration for the 841s but I can't seem to find a setting that addresses the problem. Is anyone else experiencing this and if so do you have any pointers on what I might try to correct? TIA, Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call monitoring in external application (newbie)
Hi all, I'm newbie in asterisk (just first install) I'm looking some ideas to send info about incoming call to another process (my app) I have this problem asterisk is actually installed syde by side with the legacy pbx, one my program talk with the pbx and offers some custom services on the lan, I need to inform my program with at least caller id when a call is incoming on asterisk, I can get data via tcp ip or other way ? unfortunately my program run on win :-/ thanks for any suggestion... ciao. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Monitoring
Can anyone help me how to open recorded converstations in asterisk? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Monitoring
Hi, if the file format is a problem, try Wavepad, it could help you. Giorgio Ian Bert Tusil wrote: Can anyone help me how to open recorded converstations in asterisk? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- GIORGIO INCANTALUPO Tel. +39 02 9350 4780 (104) FGA Software 20017 Rho - Via Puccini, 8 E-Mail : [EMAIL PROTECTED] Internet: http://www.fgasoftware.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Monitoring
On 7/27/05, Giorgio Incantalupo [EMAIL PROTECTED] wrote: Hi, if the file format is a problem, try Wavepad, it could help you. Giorgio Ian Bert Tusil wrote: Can anyone help me how to open recorded converstations in asterisk? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- GIORGIO INCANTALUPO Tel. +39 02 9350 4780 (104) FGA Software 20017 Rho - Via Puccini, 8 E-Mail : [EMAIL PROTECTED] Internet: http://www.fgasoftware.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I built a web interface named ARI (Asterisk Recording Interface). Download it here: http://www.littlejohnconsulting.com/?q=ari Place it in /var/www/html/recordings. AMP is including it in their distribution and I will make updates there and on my website. Regards; Dan Littlejohn (512) 791-0137 www.littlejohnconsulting.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] * Call Monitoring
I've got a nagios plugin making sure the * box is up, but I would like to do more than that. I need to make sure the PRIs connected to my box stay up and I need to make sure calls are not failing for any reason. Are there any * monitoring packages like this? -Daniel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * Call Monitoring
Daniel Corbe wrote: I've got a nagios plugin making sure the * box is up, but I would like to do more than that. I need to make sure the PRIs connected to my box stay up and I need to make sure calls are not failing for any reason. Are there any * monitoring packages like this? -Daniel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://www.voip-info.org/wiki-Example+Argus+Config ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * Call Monitoring
On Mon, 21 Feb 2005 08:00:40 -0500, Daniel Corbe [EMAIL PROTECTED] wrote: I need to make sure the PRIs connected to my box stay up and I need to make sure calls are not failing for any reason. Are there any * monitoring packages like this? There aren't any specific tools that do exactly what you want afaik. It wouldn't take much to taylor a few things yourself though. As for the PRI processing calls. You could always drop a call file in from the cron every 10 minutes that makes a call out and back in. Then you you can run a script that looks over your CDR to verify that the call was received. Have it call a specific context or application to look for. As for calls failing this could be a challange. What do you consider failing? You could use something like my-swatch to tail the log file looking for certain patterns. PRI alarms would be an obvious. Might take you a day or so to get these things going, but it would be well worth your time and piece of mind. -Chuji ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * Call Monitoring
Okay
here's a quick and dirty little perl script to monitor the PRI Status
and mimic nagios plugin output.
-Daniel
On Mon, 21 Feb 2005 07:50:45 -0600, Brian Roy [EMAIL PROTECTED] wrote:
On Mon, 21 Feb 2005 08:00:40 -0500, Daniel Corbe
[EMAIL PROTECTED] wrote:
I need to make sure the PRIs connected to my box stay up and I need to
make sure calls are not failing for any reason. Are there any *
monitoring packages like this?
There aren't any specific tools that do exactly what you want afaik.
It wouldn't take much to taylor a few things yourself though.
As for the PRI processing calls. You could always drop a call file in
from the cron every 10 minutes that makes a call out and back in. Then
you you can run a script that looks over your CDR to verify that the
call was received. Have it call a specific context or application to
look for.
As for calls failing this could be a challange. What do you consider
failing? You could use something like my-swatch to tail the log file
looking for certain patterns. PRI alarms would be an obvious.
Might take you a day or so to get these things going, but it would be
well worth your time and piece of mind.
-Chuji
#!/usr/bin/perl
###
# Michael Jastremski
# Monitor Asterisk PBX via Manager Interface
# http://megaglobal.net/docs/
###
# Based upon:
#
# TACI - Trivial Asterisk Call Interface v.02
# Last update 3/30/2004
# Tony Wasson [EMAIL PROTECTED]
#
#
# Modified by Daniel Corbe to monitor PRI spans
# [EMAIL PROTECTED]
#
# -Daniel
#
$ENV{'PATH'}='';
$ENV{'BASH_ENV'}='';
$ENV{'ENV'}='';
$| = 1;
use Net::Telnet ();
use File::Basename;
use lib /usr/local/nagios/libexec;
use utils qw(%ERRORS);
my $mgr_user = nagios;
my $mgr_secret = XyXyXyXyXy;
my $failed = 0;
my $reason = undef;
my $server_ip = 127.0.0.1;
my $prispan = $ARGV[0];
$tn = new Net::Telnet (Port = 5038,
Prompt = '/.*[\$%#] $/',
Output_record_separator = '',
Errmode= 'return'
);
$tn-open($server_ip);
$tn-waitfor('/0\n$/');
$tn-print(Action: Login\nUsername: $mgr_user\nSecret: $mgr_secret\n\n);
unless($tn-waitfor('/Authentication accept*/'))
{
$failed = 1;
$reason = Failed Connect;
}
else
{
$tn-print(Action: Command\n);
$tn-print(Command: pri show span $prispan\n\n);
#Response: Follows
#Primary D-channel: 24
#Status: Provisioned, Up, Active
unless($tn-waitfor('/Response: Follows\nPrimary D-channel: (.*)?\nStatus:
Provisioned, Up, Active/'))
{
$failed = 1;
$reason = PRI Span # . $prispan . is down;
}
else
{
$tn-print(Action: Logoff\n\n);
}
}
print PRI Span #$prispan is up\n unless $failed;
print $reason\n if $failed;
exit $ERRORS{'CRITICAL'} if $failed;
exit $ERRORS{'OK'};
exit 0;
__END__
TODO:
-- Maybe check other variables?
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * Call Monitoring
On Mon, 2005-02-21 at 08:00 -0500, Daniel Corbe wrote: I've got a nagios plugin making sure the * box is up, but I would like to do more than that. I need to make sure the PRIs connected to my box stay up and I need to make sure calls are not failing for any reason. Are there any * monitoring packages like this? Interesting you should ask this today... I got to work this morning and was wondering why some of my calls were still diverting to my mobile. Eventually I realised that they were diverting on no answer. A restart of asterisk, reload of modules etc made no differences, I couldn't do anything with the line. Eventually I worked out it was a telco problem (no dialtone/etc) so I logged the fault. I looked at zttool and it showed a red alarm... In around 10-20 minutes I hacked zttool.c and converted it into a very basic cli version (which doesn't need newt) and would just dump the current status of all the spans. Similar to what you see on screen when you first start zttool. Then, I threw together some simple shell scripting to analyse/send the report to BigBrother (www.bb4.org). So far it is working nicely, by tomorrow night (yes, 27 hours after reporting it) hopefully my line should come back, and the alarm should change to OK... I'll put the package etc onto www.deadcat.net (BB addons website) and drop a post here when it is done. Will also put it onto www.websitemanagers.com.au/asterisk/ BTW, I did need to suid the zttool-cli command to root, as the normal BB user doesn't have the needed permissions. I haven't looked into this, but if anyone has a suggestion on a better way to do this, feel free to let me know. Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 9345 4395[EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * Call Monitoring
Yeah, I'd be interested in porting your work so it runs under nagios. Please post your results when you're finished. -Daniel On Tue, 22 Feb 2005 02:54:22 +1100, Adam Goryachev [EMAIL PROTECTED] wrote: On Mon, 2005-02-21 at 08:00 -0500, Daniel Corbe wrote: I've got a nagios plugin making sure the * box is up, but I would like to do more than that. I need to make sure the PRIs connected to my box stay up and I need to make sure calls are not failing for any reason. Are there any * monitoring packages like this? Interesting you should ask this today... I got to work this morning and was wondering why some of my calls were still diverting to my mobile. Eventually I realised that they were diverting on no answer. A restart of asterisk, reload of modules etc made no differences, I couldn't do anything with the line. Eventually I worked out it was a telco problem (no dialtone/etc) so I logged the fault. I looked at zttool and it showed a red alarm... In around 10-20 minutes I hacked zttool.c and converted it into a very basic cli version (which doesn't need newt) and would just dump the current status of all the spans. Similar to what you see on screen when you first start zttool. Then, I threw together some simple shell scripting to analyse/send the report to BigBrother (www.bb4.org). So far it is working nicely, by tomorrow night (yes, 27 hours after reporting it) hopefully my line should come back, and the alarm should change to OK... I'll put the package etc onto www.deadcat.net (BB addons website) and drop a post here when it is done. Will also put it onto www.websitemanagers.com.au/asterisk/ BTW, I did need to suid the zttool-cli command to root, as the normal BB user doesn't have the needed permissions. I haven't looked into this, but if anyone has a suggestion on a better way to do this, feel free to let me know. Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 9345 4395[EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * Call Monitoring
BTW, I did need to suid the zttool-cli command to root, as the normal BB user doesn't have the needed permissions. I haven't looked into this, but if anyone has a suggestion on a better way to do this, feel free to let me know. It's called sudo jens ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Monitoring on IAX Channels - ChanSpy
Hi I was trying to use ChanSpy command, but it seems like it is not implemented, or is not included in the standard asterisk distribution. Can someone tell me how to obtain this. I am trying to monitor IAX channels, and I know now that ZapMonitor cmd doesnt work on this,so can anyone tell me how to monitor IAX channels. Thanks in advance Do you Yahoo!? Yahoo! Search presents - Jib Jab's 'Second Term'___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call monitoring
I start cal monitoring with:
exten = _1XX,1,Answer
exten = _1XX,2,Monitor,wav
exten = _1XX,3,Dial(SIP/${EXTEN}|30|tr)
I can record the call, that is correctly forwarded to SIP destination,
but i cannot ear the ringing tone.
If i put
exten = _1XX,3,Dial(SIP/${EXTEN}|30|mr)
i can ear instead music on hold.
without Monitor r dial options work...
Any tips ???
Thanks,
Igor
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
