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Hi all, I would like to know if there is any possibility to send an event when a call is monitored? For both start and stop monitor. There is no event sent on asterisk 1.2 for that monitor case. I did not find any changes regrding that on 1.4. Am I wrong? Is it even possible to send an event when a monitor starts or stop ? Or is this a bad idea. Regards, Daniel

Hello, Is there any way of removing this line from showing on the console? I have a script that logs in every few seconds to manager and it makes the CLI output very hard to follow because of the " == Parsing '/etc/asterisk/manager.conf': Found". (Yes, Found! manager.conf was there 3 seconds ago, guess what it's still there.) There is a very old feature request about this

...d don't see a benefit in adding a proxy to that picture. =Stefan -- reuter network consulting Neusser Str. 110 50760 Koeln Germany Telefon: +49 221 1305699-0 Telefax: +49 221 1305699-90 E-Mail: stefan.reuter@reucon.com Jabber: stefan.reuter@reucon.com Steuernummern 215/5140/1791 USt-IdNr. DE220701760 -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 252 bytes Desc: OpenPGP digital signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20070305/bc19ec20/signature.pgp

Hi - Am I allowed to have multiple managers logged in with the same manager username at the same time? I'm referring to the id names in manager.conf. I expect so, but just want to check to help in troubleshooting a problem. thanks -todd-

...tact me off-list and I'll have a look if I still missed anything. -- reuter network consulting Neusser Str. 110 50760 Koeln Germany Telefon: +49 221 1305699-0 Telefax: +49 221 1305699-90 E-Mail: stefan.reuter@reucon.com Jabber: stefan.reuter@reucon.com Steuernummern 215/5140/1791 USt-IdNr. DE220701760 -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 252 bytes Desc: OpenPGP digital signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20070417/e7a692b2/signature.pgp

Hi, I'm wondering whether or not I should go for ODBC or IMAP voicemail storage. Before diving into details, I would be very pleased to get input form others. 1. With IMAP, is it necessary to save a copy of voicemails in /var/log files so that a user can still listen to his (or her) own voicemails with his own hardphone ? 2. How then, can you make sure to skip non-voice mails stored in the

...velopers? Best regards, Stefan -- reuter network consulting Neusser Str. 110 50670 Koeln Germany Telefon: +49 221 1305699-0 Telefax: +49 221 1305699-90 E-Mail: stefan.reuter at reucon.com Jabber: stefan.reuter at reucon.com WWW: http://www.reucon.com Steuernummern 215/5140/1791 USt-IdNr. DE220701760 -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 260 bytes Desc: OpenPGP digital signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20090513/bfcc97aa/attachment.pgp

..._1_0_0_m3_released.html =Stefan -- reuter network consulting Neusser Str. 110 50670 Koeln Germany Telefon: +49 221 1305699-0 Telefax: +49 221 1305699-90 E-Mail: stefan.reuter at reucon.com Jabber: stefan.reuter at reucon.com WWW: http://www.reucon.com Steuernummern 215/5140/1791 USt-IdNr. DE220701760 -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 260 bytes Desc: OpenPGP digital signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20091019/715fd2a6/attachment.pgp

Hello all, A client of us, needs a queue monitoring system. In realtime he needs to now the PRI status, the agents logged in and logged out, the number of received calls by agent, ....,etc. I am not a call center specialist and i want to find a call center software to offer to my client that fits his needs. I need a monitoring solution for incomming and outgoing calls and a queue management

Dear all, I've noticed that when I have a phone registered in Asterisk, and then I register another phone with the same user, the "sip show peers" in the CLI shows that Asterisk replaced the IP of the first phone by the IP of the last one registered for that user. Consequently, if someone calls that user, only the last phone rings!! How may I configure Asterisk to be able to