search for: reucon

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

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

...y ExtraUnqiueId is used, the Priority property is used to redirect the extra channel (instead of ExtraPriority) 2. If the property ExtraChannel is used, 0 is used to redirect the extra channel regardless of the Priority 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@r...

...ws you to move your AGI scripts to a dedicated server and increases performance by eleminating the need to start the language interpreter for each request. Our current snapshot release includes an AGI demo in Groovy, JavaScript and PHP to get you up and running quickly. Check it out at http://blogs.reucon.com/asterisk-java/2009/05/13/scripting_support_for_fastagi.html and provide feedback. Will this also be useful for non-Java developers? 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.r...

Hi, We've just released milestone 3 of Asterisk-Java 1.0.0. Next to a few bug fixes this new milestone makes Asterisk-Java OSGi compliant and adds support for the modern SLF4J logging framework. Have a look at http://blogs.reucon.com/asterisk-java/2009/10/19/asterisk_java_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....

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

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

...channel was hang up. Only successful OriginateResponse will contain the unique id of the established channel. Is there any way that my AMI application can get the original SIP response of the failed Originate action? I'm using Asterisk 1.8.22 and slightly tweaked asterisk-java ( https://blogs.reucon.com/asterisk-java/) 1.0.0. -- ????? ?????? ???? ???? ???? ?????? ??????? ?????????! ???? NOW! Moshiach is coming very soon, prepare yourself! ??? ?????? ?????? ?????? ??? ????? ????? ???! -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/...

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:

...(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: application/pgp-signature Size: 252 bytes Desc: OpenPGP digital signature Url : http://lists.digium.com/p...

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-

...ou 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: application/pgp-signature Size: 252 bytes Desc: OpenPGP digital signature Url : http://lists.digium.com/p...

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

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

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.

Hi, I need a hack to query current calls coming in and going out an Asterisk 1.6.1 system and send this list back as a custom UserEvent to other listening endpoints. For various reasons, I might need to write this hack in PHP though I'm more experienced with Asterisk Java. What is your opinion regarding PHP AMI API and Asterisk 1.6.1 ? I'm referring here to

Hi, In our office, we're slowly migrating from a cisco call manager set up to asterisk. Problem is management doesn't want to buy any other hardware ?as they had already invested a lot in cisco. The main cause of this is asterisk's added features like unique FAX number for everyone in the company (which will be the same as phone DID), Voice mail, Auto Answer etc yet we need thousands