similar to: queue caveats

Regarding this article (1) I have one question to make. What can I do to record the call if the agent makes a transfer using the "flash" button instead of "transfer button" or using blindxfer or atxfer defined in features. conf If the agent makes the transfer with "flash", the comunication between the person who is calling and is already in the queue and the target

When agent tries to transfer a phone call (*2 - att transfer) he hangs up. Why? When a phone call isn't from queue then att transfer works fine. In features conf I have *1 for recording, *2 for att transfer and #1 for blind. In queue blind transfer works. For disconnect I have #0. I guess that * is somewhere defined as for hang-up the call, but where? I can't find it anywhere. Any help

Hello, when a call comes in and is answered by colleague A, this colleague A sees the CallerID of the external calling number. When colleague A transfers the call to colleague B, attended or unattended, then colleague B sees the number of colleague A on his screen while talking to the external calling number. I expect here that colleague B would see the external calling number on the screen

I all. I have configured a pair of * servers, sip connected each other Mi problem is the following If on the first * i configure a queue containing phone number of the second * (i.e with a round robin strategy) I have non problem as far as all phones are online. If one of the remote phone number is unavailable, when the round-robin strategy touch that phone the call is answered by the voicemail

Hello list, I am looking for a way to have multiple remote Windows users download a package and get connected to *. My idea would be that they run a simple app, it connects without any setting to an * box (maybe via IAX) and then people press a button to talk. It would be okay if they had to enter a username and password, but not more than that. Looking for such software, I keep finding

Hello list, I was looking for a way to implement non-blind call transfers with *, i.e. the usual behaviour of most PBXs when pressing the flash button: - A and B are talking - A pushes flash - A is free to compose a new number - B hears music on hold - A's call is answered by C - A hangs up - B and C are in conversation As much as I can understand, * only supports blind transfers, where if

Hi, What do I need to do to put an agent into two queues? The idea being that the agent will get the call no matter which queue it comes into? ~ Matt

Release 0.66 of IPSwitchBoard is now available for FREE download at: http://www.voip-info.org/tiki-index.php?page=IPSwitchBoard+BETA Enhancements: Support for Call Parking and retrieve/forward them again. Last Call on the Queues Page now displays a date-time in human readable format. Added CallerID on the Queue Members listing on the Queue page. New page with Agent information. Minor bug

Hello list, I am having trouble listening to GSM files created by Asterisk using a browser. I am noticing that some of my users succeed in listening to them and some others don't. I guess it is something of a codec problem that does not seem to be installed on all machines (though they are all WinXP). Anybody knows what one should do to listen to GSM files? I send files through the

Hello list, I put together a quick note about how to "see" oh323 calls while they are handled by your * box. http://www.oinko.net/astrecipes/index.php?n=89 The article is just a draft with usage examples; I'd love to hear your comments and updates if there is something I got wrong. Thanks l. -- Assum est, versa et manduca.

Hello using Asterisk 1.8.32.3. What is the best way of knowing a call is being transfered (attended and unattended) ? And also knowing whereto (sip user) the call is being transfered and who is the transferer ? So I can log this information. Kind regards. J. -------------- next part -------------- An HTML attachment was scrubbed... URL:

I know I must be being daft, but is there a way to set which context the queuing system uses when it dials the operators/agents? By default it appears to use the default context. I've looked through voip-info.org and can't find anything, someone please put me out of my misery.

PBX: Asterisk 1.4 Phones: PSTN phone connected to TDM400 X-Ten Lite Polycom 430 Scenario Polycom 430 = User1 User2 calls User1(Polycom 430) asks to be transfered to User3 User1 does an attended transfer using the trnsfr button on the polycom User2 is placed in music-on-hold User3s phone rings. (So far so good Right?) User3 picks up the phone to answer User2 only to

Ok, I've seen discussion before on doing transfers (attended and unattended), there seems to be much confusion over it. As things sit, I've been trying (unsuccessfully) to do transfers with a zap channel (analog phone attached to TDM400). I have no idea what I'm missing. My current understanding is that I need to have transfer=yes in zapata.conf, do a flash hook, dial the 2nd

Hi, How do I read (make sense of) the timestamp in the queue_log? I'm probably just slow but I don't understand it. Thanks! Regards, Jan

Anyone know how to get app_queue to send logs to MySQL or any other sql server. I found info for cdr's and even configs but nothing on queue_log. If sql is not supported in the current app_queue I will be willing to pay someone to add it. John Bittner Simlab.net

Hello all, I am running into some problems with a snom 105 phone trying to do a attended transfer . Snom phones are connected to Asterisk. This does not work, it will only do a unattended transfer. I have downloaded the manual from snom and followed the instructions. Has anyone experienced the same problem ? any ideas how to solve the problem. thanks, Arne.

Does anyone know a softphone that can accept URLs during a call and open that page in the default browser when the call is answered? I Know DIAX and the IDEFISK, only pro version.I need another ones. It can be using the cmd SetURL Regards. -- Bruno de Assump??o Loureiro msn: loureiro_bruno@hotmail.com

I have a challenge that is ending up quite interesting. I need to identify which SIP phone "touched a call last", that is, which phone did the last transfer or dialed the original call if no transfers were done. It is easy in the case of a regular, non-transfered call. Just put something in callerid= in sip.conf, and that will show up in ${CALLERID}. The same with an attended transfer,

Hello, We have an application where operators will sometimes take an incoming call from a queue, then contact an outside line, do a consultation, and finally do a SIP attended transfer to join the two parties together. We'd like to record the incoming caller's conversation with the operator and the attended part of the outgoing call, but not the unattended part, after the transfer has