similar to: Wrong extension indicated when logging in as agent

Hi, This problem may be related to a configuration problem but I believe it is a bug in the FOP since restarting the FOP server clears the problem. Here is the scenario: Using AgentCallBackLogin and have four agents logged in a call is made to one of the agents directly from an internal phone. Okay so far. Call is hung up and the same extension is used to call another agent okay again, no

Hi, I have an odd problem when doing a blind transfer. The transfer is intiated and the transferred caller hears nothing until the timeout. I have tried setting the 'r' and the 'm' variables in the dial command. Nothing happens when I use the 'r' variable when I use the 'm' variable I briefly hear music on hold and then it stops until the timeout for no answer

Peter Zotov писал 09.01.2013 19:59: > Hello. > > I've managed to create a bitcode file (attached; also available at > [1]) which produces > a series of identical errors when verified: > > | Global variable initializer type does not match global variable > type! > | %i.NilClass* @nil > > When ran through llvm-dis and recompiled, through, it verifies >

Tzafrir, How did you set sip:tzafrir@local.xorcom.com I use ser----asterisk look at my sip.conf and extensions.conf Regards Harry //////////////////////////////////////////////////// [general] context=sip realm=nxs.yi.org bindport=5050 bindaddr=nxs.yi.org srvlookup=yes tos=lowdelay maxexpirey=3600 defaultexpirey=1000 allow=all musicclass=default language=fr insecure=very allowguest=yes

Tzafrir, How did you set sip:tzafrir@local.xorcom.com I use ser----asterisk look at my sip.conf and extensions.conf Regards Harry //////////////////////////////////////////////////// [general] context=sip realm=nxs.yi.org bindport=5050 bindaddr=nxs.yi.org srvlookup=yes tos=lowdelay maxexpirey=3600 defaultexpirey=1000 allow=all musicclass=default language=fr insecure=very allowguest=yes

Hi, I have a small call center with 4 Zap lines and 4 agents. Agents login using sip phones with AgentCallbackLogin. I occasionally gets a complaint that when customers call the call center, after the initial greeting is over the call gets cut after playing the thank you message. I started investigating and found that that happens when the call gets transferred to an agent who is making an

Hi, I am setting up a small call center using *. I have ZAP setup for incoming calls and IAX setup for agents. Agents login using AgentCallbackLogin. When customers call, it's getting picked up and when queue is trying to call back the agents, I am getting error. I am using CVS HEAD, and updated just now. The error is: -- Executing Answer("Zap/1-1", "") in new

Here is some dialog from the Console: -- Starting simple switch on 'Zap/13-1' Feb 10 07:22:36 NOTICE[21105]: chan_zap.c:6063 ss_thread: Got event 18 (Ring Begin)... -- Executing Goto("Zap/13-1", "mainmenu|s|1") in new stack -- Goto (mainmenu,s,1) -- Executing BackGround("Zap/13-1", "thank-you-for-calling-poker -support") in new stack

Hi, I have the same problem with the queue configuration When I receive 2 calls only 1 phone ring even if more agent's phone are free. The second call will go to an other agent only if the first call is pickup. Somebody have a solution ? This is my config file : Queue.conf [general] ; ; Global settings for call queues ; ; Persistent Members ; Store each dynamic agent in each queue

Hi all! My problem is that calls being placed in the queue, and are waiting while the agents are busy, when an agents is then free they gets connected to the agent but there is silence (no voice). If a caller has not to wait in the queue, there is no problem. My agents have an iax2 client, and imcoming calls are over SIP. queue.conf: persistentmembers=yes autofill=yes ringinuse=no

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi there, I'm now the proud owner of an X100P and am struggling to set up a CVS-compiled Asterisk to do my bidding. I checked zaptel/zapata/asterisk out today and pretty much did a straight make install on all packages. So far the only consistent trick I can make it perform is calling from one SIP phone to another. Could I get a bit of

Hello. I've managed to create a bitcode file (attached; also available at [1]) which produces a series of identical errors when verified: | Global variable initializer type does not match global variable type! | %i.NilClass* @nil When ran through llvm-dis and recompiled, through, it verifies successfully. If I disassemble it one more time, the result is identical to the first

Hi all, I have a question with the "getelemptr" instruction. E.g.: I have some GEP instructions in my program. Some look like: <INST_GEP op0=26 op1=64 op2=429/> . <INST_GEP op0=341 op1=64 op2=101 op3=499 op4=0/> The first instruction above in assembly file: %tmp60 = getelementptr [512 x i32]* @weights, i32 0, i32 %k.3.ph Ok, we see it all: Index of @weights in value

I tried to add a isel pattern [(set I32Regs:$dest, (select (seteq I32Regs:$op1, I32Regs:$op1), I32Regs:$op3, I32Regs:$op4))] for my instruction. I get an error message "Could not infer all types in a pattern!" Above the error message it prints error message line containing the name of my op and (set I32Regs:i32:$dest, (select:i32 (setcc:{i1:i8:i16:i32:i64} I32Regs:i32:$op1,

I am trying to setup an interactive menu where a caller hits the main menu and can then dial an extension. As far as I can tell the "Waitexten" app is failing to read 3 digits and just reading the first and then announcing that it is invalid since all extensions are 3 digits. How do I make Waitexten wait for 3 digits? I have setup the extension "100" for users to reach the

Hi list I encountered problem in using Background command. Below is my extensions.conf. [mainmenu] exten => 4,1,Wait(1) exten => 4,2,Background(thank-you-for-calling) exten => 4,3,Goto(n01|s|1) [n01] exten => s,1,NoOp(${CONTEXT}) exten => s,2,Background(thank-you-cooperation|m) exten => s,3,WaitExten() exten => s,4,Playback(digits/pound) exten => 1,1,Playback(digits/1)

Hi I can't for the life of me work out why this is not working. When in the campon contect if you hit a DTMF key 2 you get moved to the exten => 2 defined in the mainmenu context not the exten => 2 defined in the campon context. What is wrong? The same happens if you hit key 1. [campon] exten => _*1XXX,1,Answer exten => _*1XXX,2,SetCallerID(${CALLERIDNUM}) exten =>

Just another bit of info which might help solve this: Looking at the Asterisk log messages - I notice when I start up Asterisk, I see the error: pbx_config.c: Can't use 'next' priority on the first entry! Could I be right that its something got to do with priorities? I changed the incomingpstn context to the following eliminating the 'n' field and still the same errors were

Guys I am having some serious issues with my call queue and Management is breathing down my neck pretty bad, and I am running out of ideas. I have a single queue for my tech support department. I originally was using the AgentCallbackLogin for them and it tested out great on our testing weekends, but it hasn't worked out since. It would only let one of them take calls at a time, no matter

I'm using ver. 1.0.7 here are a couple of lines from my extensions.conf file: exten => x,1,AgentCallbackLogin(${CALLERIDNUM}|${CALLERIDNUM}@sip) exten => x,2,Hangup I'm looking for a way to capture the Agent ID after login, to keep track which agent is associated in a certain call.