search for: markq

...queues will do it all for you. http://www.voip-info.org/tiki-index.php?page=Asterisk+config+queues.conf Use this part to play the sound to the callers: ........ ;queue-youarenext = "queue-youarenext" ; ("You are now first in line.") ........ and use: ........ ;announce = queue-markq ........ To announce what you want to the callee Hope this helps _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/ma...

I think I solved the errors I was getting with my patch, sort of anyway. Brief over view: Tell all the callers their position in the queue. When they move, tell them their new position. I was receiving Thread xxx already blocked by xxx. I found that if I only tell caller 4 and above (Which becomes caller 3) that their position changed, I do not receive the errors.

...be specified which is played for the member as ; soon as they answer a call, typically to indicate to them which queue ; this call should be answered as, so that agents or members who are ; listening to more than one queue can differentiated how they should ; engage the customer ; ;announce = queue-markq This allows you to have one announcement per queue, You could use a script to change the file, but it does not lock the file so if you change it while it is being read the other person (agent) hears the change Do you want a dynamic message played to each agent before he picks up?? That can also...

Hi, I'm testing a Polycom IP600. With firmware version 1.1 the phone reboots at any time. With firmware version 1.2, the first reboot was an endless reboot. Then I moved the phone to another lan port, then it worked fine. Then I installed again in the initial lan port and the phone works well. However after some time of inactivity (1 hour?), the IP600 stops to send and receive calls.

...4] WARNING[4997]: channel.c:3106 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x4 (ulaw) [Dec 29 17:27:14] WARNING[4997]: file.c:147 ast_stopstream: Unable to restore format back to 4 -- Executing [s at sky:2] Queue("Skype/rexesbposolutions-084159e8", "markq|t|||900") in new stack -- Started music on hold, class 'default', on Skype/rexesbposolutions-084159e8 [Dec 29 17:27:14] WARNING[4997]: channel.c:3106 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x40 (slin) [Dec 29 17:27:14] WARNING[4997]: res_musiconhol...

Hi, I've spotted weird crash of Asterisk cvs Stable. I have defined queue in queues.conf : [prodaja] music = default announce = queue-markq strategy = ringall context = from-pstn timeout = 15 retry = 5 maxlen = 0 announce-holdtime = no announce-frequency = 30 announce-holdtime = yes monitor-format = gsm|wav|wav49 monitor-join = yes eventwhencalled = yes member => Agent/1000 ------------------------------------------- then I have in...

Hi All I would like to be able to have an announcement played to an operator advising them of the queue the call came from before the call is pasted over to them, so they know how to greet the customer. Does anyone have any ideas or can point me to some resource which details this? Many Thanks in Advance. SP

We all know you can monitor multiple devices in one hint, and it shows busy if any device in the group is busy. This is good for a user with multiple devices, but not useful for teams where any person could take a call, like a customer service group. Does anyone know if it's possible to have a hint with multiple devices which only shows busy if every device is busy? -- Carlos Alvarez

yes, robert, but how do i "join" the two legs inside a call file or in the dialplan? i have experienced that call files can do a call to a channel and if this call is answered it can either be connected to an extension inside a context or call an application with parameters. roland -----Original Message----- From: asterisk-users-bounces@lists.digium.com

Hi... I just had a real quick and simple question... I have a asterisk implementation setup w/ real time off of a mySQL database for SIP peers and queues, voicemail, agents etc... I after the upgrade to asterisk 1.4.3 there are some new configuration features i would like to use. I was wondering if i could just add to the database table a column for the new config option? if this will work or

...time=5000 recordagentcalls=yes recordformat=wav createlink=yes group=1 ackcall=yes agent => 1001,1234,Agent1 agent => 1002,1234,Agent2 queues.conf ----------- [general] [default] [callcenter] monitor-format = wav announce-holdtime = yes wrapuptime=15 musiconhold = default announce = queue-markq strategy = roundrobin announce-frequency = 60 context = queue-out member => Agent/1001 member => Agent/1002 sip.conf -------- [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) context = from-sip [mysjphone...