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...16.03.04 [Aug 11 15:53:50] Organization: Doubango Telecom RTP debug : RTP Debugging Enabled for address: 178.119.146.190:0 [Aug 11 16:01:20] Got RTP packet from 178.119.146.190:58814 (type 08, seq 014114, ts 3292374327, len 000160) [Aug 11 16:01:20] Sent RTP packet to 178.119.146.190:50760 (via ICE) (type 08, seq 033787, ts 3292374320, len 000160) [Aug 11 16:01:20] Got RTP packet from 178.119.146.190:58814 (type 08, seq 014115, ts 3292374487, len 000160) [Aug 11 16:01:20] Sent RTP packet to 178.119.146.190:50760 (via ICE) (type 08, seq 033788, ts 3292374480, len 000160) [...

Hello thank you for your answer. I don't understand how there are many tutorials and examples on the web where every time the outcome is a working setup. Very strange I feel now after my personal experience with Asterisk 11 and webRTC. You also say Asterisk 13. How about Asterisk 12 then ?? Kind regards. On 10-08-16 21:53, Matt Fredrickson wrote: > I don't see an ice-ufrag or

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

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Hello, I look at the mobicents project. Somebody has experience within both projects ? Regards Harry ___________________________________________________________________________ Faites de Yahoo! votre page d'accueil sur le web pour retrouver directement vos services pr?f?r?s : v?rifiez vos nouveaux mails, lancez vos recherches et suivez l'actualit? en temps r?el.

...ty and ExtraPriority properties. A patch for manager.c is available at http://www.reucon.net/~srt/bristuff_redirect.patch as a result to a bug filed against Asterisk-Java at http://jira.reucon.org/browse/AJ-34 I've sent a notice to kpj. =Stefan -- reuter network consulting Neusser Str. 110 50760 Koeln Germany Telefon: +49 221 1305699-0 Telefax: +49 221 1305699-90 E-Mail: srt@reucon.net Jabber: srt@jabber.reucon.net -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 254 bytes Desc: OpenPGP digital signat...

Hi all, I want to be able to see the status of my Agents on a web interface. I have no idea how to do so. I have found a few sample script to communicate with queues manager to view queues.But I couldn't find any on viewing the agent status. Could anybody give me a clue? Regards, Pim

Our windows users are looking for a simple application to permit dialling and transfer from Windows desktop (or web page). I've looked at everything mentioned in the WIKI, and most are either not appropriate, or are not maintained any longer. I've used Flash Operator Panel, and quite like it, but I don't believe there is a way to have a per-user view (so people can only manage

These are common functions. Why "Asterisk Manager" doesn't provide commands to hold and retrive an active channel? If it must be implemented by AGI, could anyone give a direction or steps? Thanks in advance, James -------------- next part -------------- An HTML attachment was scrubbed... URL:

...t;Example to hangup all active channels: >for (AsteriskChannel channel : server.getChannels()) >{ >channel.hangup(); >} >http://asterisk-java.org >I am sure other libraries provide similar abstraction. >=Stefan >-- >reuter network consulting >Neusser Str. 110 >50760 Koeln >Germany >Telefon: +49 221 1305699-0 >Telefax: +49 221 1305699-90 >E-Mail: stefan.reuter >Jabber: stefan.reuter -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070216/ff6dfd63/attachm...

...sure is the EventListener objects you pass to these connections synchronize access to shared data (if there are such accesses). I think this approach is rather simple for the user and 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: applic...

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-

...e article include a nice diagram on how using Local channels and Originate relates to the events you see on the Manager API. =Stefan P.S. If you still encounter problems please contact 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: applic...

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

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

Hi, I'd like to use the convenience of apt packaging, but debian sarge's default asterisk is something like 1.0.7. Are there any apt repositories which provide newer versions of the software? Thanks! -- Jean-Michel Hiver - http://ykoz.net/ D?couvrez la R?union des Technologies IP & Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE

i've seen that opening a socket on the asterisk server i can originate a call from one extension to another in a specific context. Is it possible to transfer an existing call from the extension ... SIP/xxx to another extension in a specific context? thanks

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

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