similar to: addqueuemember recording and reporting

I am having a difficult time with the transition from agentcallback login... Here are a few of the isssues, I am logging in using chan_ local ie:local/8000 as the extension Call Detail records no longer show agent/xxxx as the dstchannel show agents no longer shows the channels state show queues does not show the member Can anybody help? I have a ton of time invested in applications I developed

I have had a lot of complaints about the time it takes to setup a call. I have timed it and it is almost five seconds before it even starts ringing. The SIP device sends the request almost instantly but the channel is taking a long time to pickup and dial. It wouldn't be so bad but they hear nothing. I would like to provide ringback before the zaptel actually starts ringing the channel. Has

I have compiled zaptel on Mandrake following everything I have always done on Fedora. It is 2.6 udev so... I had to modify the 01-devfs.rules Make linux26 Make Make install... Everything appears to compile correctly but it says module not found when doing "modprobe zaptel" Is this a rights issue? Jordan Novak -------------- next part -------------- An HTML attachment was

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Good Day again list, Encountered another problem in the ACD queue... If I use the ADDQueueMember to dynamically add members as foolows, exten => 403,1,AddQueueMember(techsupport|SIP/${CALLERIDNUM}) lets assume I called extension 403 from my extension 2204. then a caller (extension 2203) enters into the techsupport queue I am able to receive the support call on my phone (extension 2204

We are using an Asterisk box to do conferencing right now. I have had about sixteen active lines in conference and the quality was acceptable. We now have a need for 50 people to conference at one time. Does anyone have enough experience doing this to give me some pointers. Will it even be reasonable to try this? Is the mixing done on the the hardware, I plan on using a quad span t-1 card from

I am looking at some directions on how to install and it is asking me to edit defines.php, it states that the file should be located in the source directory, but I can't seem to find it anywhere on my machine. Anyone been thru this? Jordan Novak Communications Technician -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have installed asterisk on numerous servers. Every install was done on Fedora and (White box Linux). I now have zap channels in one of the boxes (T-1). No matter what type of channel I call on (sip or zap) I get some really strange artifacts in the sound, almost like a skip in the playback. It seems to always be in about the same place in the recording. Usually in the beginning of playback. For

Hi i have problem with AddQueueMember logic. I need login Agent(Member) in asterisk. use this option: for example: AddQueueMember(queuetest,SIP/ekiga,10,,Agent/13) and now i want to call to this Agent: exten => _1XX,1,Dial(Agent/${EXTEN:1}) call to 113 and asterisk should call to Agent => 13 on interface SIP/ekiga. This doesn't work, How can i do this on Asterisk 1.4(not

I am developing an ACD front end using Asterisk 1.2.14. I heard that AgentCallBackLogin will be deprecated in future version of *. Is this true? If it is, how can I use AddQueueMember to replace AgentCallBackLogin? I mean to login an agent in multiple queues at once. I have multiple queues and a lot of agents defined in queues.conf and agents.conf. Each agent may login more than one queue. It

I am using Asterisk to send Voicemail out as Email. I am running into a problem I believe to be caused by the exchange server requiring SMTP authentication. I cannot get the sys admin's to turn it off. Does anyone know enough about sendmail to help me. I am assuming that the default mail client is sendmail. It will also send to other non-SMTP authenticated servers. Your help is much

Hi, I have callcenter running with v 1.2 with AgentCallbackLogin and now trying to move to 1.4 using the example doc, doc/queues-with-callback-members.txt. From what I understand the basic idea in the example is to 1. Authenticate a caller with VMAuthenticate 2. Get his SIP Channel number 3. Use

I have three polycom 501 that are all hearing echo. The other party sounds fine but you can hear yourself rather well. The volume does help if lowered but that also makes the other party extremely quiet. Is there any way to control the gain of the mic or stop the microphone from picking up so much from the handset. It only happens while you are on the handset. -------------- next part

I am looking for completly wireless desktop phones. Until I realized we needed wireless i was going to use polycom soundpoint 501's. Any suggestions on a comparable wireless phone? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070328/88e22671/attachment.htm

Hi, I'm currently migrating to 1.4 and have problems changing deprecated AgentCallbackLogin to AddQueueMember. I have dynamic queues and dynamic agents (MySQL Realtime), and pseudo-dynamic agents.conf (with huge amount of possible agent numbers). Agent login is done trough manager API: * AgentCallbackLogin * QueueAdd In 1.4 seems AddQueueMember can do all the same, but there is no such

Following on from a posting yesterday from Kevin, I have the following in the dialplan: exten => 709,1,AddQueueMember(SomeQueue|Local/706@AgentQ) I am on extension 706. From the CLI: SomeQueue has 0 calls (max unlimited) in 'rrmemory' strategy (0s holdtime), W:0, C:0, A:3, SL:0.0% within 60s No Members No Callers I call 709, get a console message

Hello, I add a member to a queue with AddQueueMember, but the Queue still indicates "joinempty" : Add member to queue : /-- Executing [queueadd at sub-GetParams:2] AddQueueMember("SIP/sip17-00005c1e", "myqueue11,member3") in new stack -- Executing [queueadd at sub-GetParams:3] NoOp("SIP/sip17-00005c1e", "AQMSTATUS = ADDED") in new stack/ ...

hello, anyone has worknig ISDN hfc-pci card in DDI (DID) point2point mode? what kernel ? and second question mISDN driver .. anyone has working solution with mISDN and maybe fritz card? what you suggest for DDI -> point2point mode (card,kernel,chan_..., ...) ? thank you, Tomaz

Hi, I'm having hard time to emulate agencallbacklogin. Agent can logon and receive call without any problem using addqueuemember. The problem comes when I try to evaluate their performance using queuemetrics. Here is an exemple of my log script: ;Agent Login exten => _60XXX,1,Macro(agentLogin) [macro-agentlogin] exten => standard,1,AddQueueMember(queue1) exten =>

Is there an app or softphone for meetings that displays the hosts screen like webex or intercall. Jordan Novak -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060321/df90d527/attachment.htm