similar to: Queue is not working

First of all I would like to thank Mark for getting roundrobin to go roundrobin. Good job. Now we have some options here for leastrecent and fewestcalls strategy. It needs some work on the logic and Mark recommend that I ask the list and get some input before he makes any changes to it. fewestcalls from what I have seen would always ring the agent with the fewestcalls first then go into

hello all, i have two asterisk server for call transfer and one more asterisk server for agent login(server_X) where agent take the call. server_A and server_B server_A is connected with pri and configure with 60 channel for call transfer into server_X server_B is connected with pri and configure with 30 channel for call transfer into server_X my query is that some time two call originate same

Hello, i use doevecot 2.2.18 current. My Problem is with email-extension and case sensitiv folders. Example: user+extenstion will be delivered to the user and subfolder extenstion so this is okay. but user+extenstion will not be delivert to the user and exiting subfolder Extenstion so that is not okay. The mail will be also delivered in the subfolder extension. The exiting Folder Extension will

Hi, How do you all handle the situation when a centrale fastagi server process(es) are down? AGI(..) prints "Unable to locate host" and the dailplan jumps to extension h. I'd like to handle the return value and keeping the caller in the dailplan and not to the hangup extension. Any tips about how to handle a AGI(..) returns -1 condition? thx Arjan Kroon Mobillion BV

Am 30.07.2015 um 08:52 schrieb Jost Krieger: > On Wed Jul 29 22:42:32 2015, Sascha wrote: > >> i use doevecot 2.2.18 current. My Problem is with email-extension and >> case sensitiv folders. >> >> Example: >> user+extenstion will be delivered to the user and subfolder extenstion >> so this is okay. >> but user+extenstion will not be delivert to the

Hi all, I would like to connect in sip mode an Eyebeam client to a messenger via Asterisk. I want to use video. Nat is not an issue as vpn connections will be used. Is this a difficult tasks, can someone give me some pointers to get started... Have a good week-end, Francois Random Thought: --------------- Wanna buy a duck?

Hello I'm using eyeBeam, and Asterisk keeps sending my clients a second call, when they are still in one call (because eyeBeam has lots of channels). I was using X-Lite (with 3 channels) and Asterisk never sent the client a second call. How can I force Asterisk (or eyeBeam) just to send one call at each time. Is this a configuration I need to do in eyeBeam or Asterisk? Thanks Regards Joao

My suggestion would be the one-line eyeBeam phone under development. Check out support.xten.com. //Jan -----Ursprungligt meddelande----- Fr?n: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] F?r afoc@interconnessioni.it Skickat: den 18 oktober 2005 14:48 Till: Asterisk Users Mailing List - Non-Commercial Discussion ?mne: Re: SV: [Asterisk-Users] Queues

hi, hs anyone able to make video to work on asterisk? i tried following this: http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+xten+eyeBeam i can see that eyebeam is trying to broadcast a video but the other eyebeam is not receiving it. i tested the same setup but this time using ser with rtpproxy and eyebeam video works fine. any ideas? where do you think should i start

Is it possible to get a few 7960's and asterisk to allow all of the 7960 phones to use one extentsion and can only be used by one person at a time, have it indicate on the other 7960's when one of the others has the line engaged. Basicly so like I can setup a rule when an incoming call comes from IAX to divert to this extension, it will ring the extension (thus all phones), and allow me to

I want to add things to the prompts like: "mark urgent" "add to message" "pause while recording message" Any examples of how to do this? I'd also like to switch around prompts, not simply edit the sound files. Is it an agi, special dailplan, patching the app_voicemail.c file? All three? Any input/examples are welcome. -thanks

Hi folks, I'm using a fewestcall queue here, and I'm having the follow problem: I have 3 static agents in my default queue: 2001 2002 2003 User 2001 and 2002 are logged in, but 2003 are logged out. When someone call to my default queue, the queue try to ring 2003 (that isn't logged). There is some way to the queue only ring logged users? Here is my show queue: zeus*CLI> show

I posted this over in asterisk-dev, realized I probably should have put it here. Hi there, We?ve been having a strange issue with a customer?s queues where a queued call will ring an available agent, agent answers, then a second or two later the agent is offered a second call which they cannot answer, since they?re already speaking with a client. This in turn causes a few issues: - Agent stats

Hi all, I upgraded one of our servers running asterisk 1.6.X to 1.8.2.3. I compiled it with SRTP support. Everything seems to work OK but I am having a weird issue. I cannot disable SRTP. I tried the /encryption=no/ in /sip.conf /and the /_SIPSRTP_CRYPTO=disable/ on my dailplan and it keeps trying to use the SRTP. Well, right now I have to have/ noload=res_srtp.so/ on my /modules.conf /otherwise

5/16/01 12:45 PM kevin@kevinsearle.com Kevin: >> The reason for asking is that everything else had "hackme" as a pw >> and "prettyplease" raised a question because we thought it may be >> a part of something else (ie. ices) and curious if we missed something >> in the setting up process. > >There are three different passwords - one for the admin,

Hello Everyone I just bough my Xten Eyebeam but i don figure out how to make the video works i only see a black screen where de remote video suposse to appear, Any help regarding this matter will be very preciated Thank You

I am looking for a SIP Soft Video phone, which I can use with Asterisk. If you have one installed (regardless if free or purchased) please tell me which one, the settings in Asterisk and your experience with it. bye Ronald

I am trying to connect two Xten eyeBeam Video Phone No problems in voice connecting. I tryed to modify my sip.conf [general] language=it videosupport=yes ; enable Asterisk video support port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) disallow=all allow=h263 allow=gsm allow=ulaw allow=alaw ; H.263 is our video codec ;

Guys. Im writing this because Ive checked the wiki, Xten website and read a lot of docs and still cant figure out a way around the NAT issues. Maybe somebody else can give me some ideas from a fresh perpective. My test setup is this: Asterisk -> 2wire homeportal Firewall -> internet Computer with Xten eyebeam The asterisk box and the computer with xten beam are behind the same

Lets presume that my both software are open. Xlute and Eyebeam But I want my calls from Asterisk to land only on Eyebeam and Not on xlite. How to set it ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090129/4011be6c/attachment.htm