similar to: cmd pickup Problem

I'm Japanese. Sorry,English is not so understood,Please let me question by items. In Asterisk-1.2.1 and 1.2.2,I cannnot understand the operation of Authenticate application's 'j' option. exten => 123,1,Answer() exten => 123,2,Authenticate(789,j) exten => 123,3,Playback(pin-number-accepted) exten => 123,4,SayDigits(111) exten => 123,103,SayDigits(999) In this

Hi all, how can I pickup a call from a queue? Which context parameter do I have to specify? The context that calls the queues application is ext-queues. This is what I tried so long (777 is the extension of the queue I want to pickup from): exten => _**ZXX,1,Noop(Attempt to Pickup ${EXTEN:2} by ${CALLERID(num)}) exten => _**ZXX,n,Pickup(${EXTEN:2}@from-did-direct) exten =>

Greetings, I have a dialplan that calls the dictate application, but I want to do some post-processing on the RAW file created. The post processing is working fine as long as the dictation application exits gracefully, but fails when the user simply hangs up. How can I make sure the system() command is run regardless? Example: [test-dictation] exten => 123,1,Dictate(/tmp/dictate) exten

Hello list, I try to pick up a ringing extension but nothing works. To be clear, I'm trying to pick up extension 10. [Jun 14 17:37:34] -- Executing [**10 at from-TESTCORP:4] Pickup("SIP/testcorp3-00000041", "10 at 123456") in new stack [Jun 14 17:37:34] NOTICE[16555]: app_directed_pickup.c:159 pickup_exec: No target channel found for 10. [Jun 14 17:37:34] --

Hi, I am trying to set up a dial plan und I have a few problems to realise some functions. The dial plan should look like this: 123,1,Answer() 123,2,Queue(1stlevel,t) 123,3,Queue(2ndlevel,t) 123,4,Queue(3rdlevel,t) 123,5,Hangup() If a member of the 1stlevel-Queue can answer the call it should be hanged up after finishing. If not, the current member answering the call should be able to transfer

Hi everyone I am having problems with my call queue We currently run a customer care call center which has attendants login during the daytime. Customers who call the 'customer care line (a specific number) always get routed to the cutomer care queue (called 124). After hours, staffs of the Network operating center provide customer care services for customers who call in after the last

Good day, does anyone have AstTAPI running for incoming calls, and would like to show some examples. My setting right now looks like this: sip.conf -------- [22] type=friend dtmfmode=info username=22 mailbox=22 secret=privat host=dynamic context=privat canreinvite=yes callgroup=1 incominglimit=2 extension.conf -------------- exten => 123,1,noop ;Hint(SIP/22) exten =>

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi! I am having problems with the PlayTones application and VoIP softphones. I have the following in my extensions.conf: exten => 123,1,Answer exten => 123,2,PlayTones(Busy) exten => 123,3,Hangup But when I connect with gnophone(IAX) or kphone(SIP) and dial 123 the call just hangs up immediately. I get the following on the console: --

Hi all, I have a GXP2000 with BLF configured. I follow the configuration guide to enable the pickup cmd as follow and include it under corresponding content. [BLF_group_pickup] exten => _**1XX,1,Pickup(${EXTEN:2}) exten => _**1XX,n,Hangup The I press the single key to pickup the call to extension 100 when there is a call to it. From CLI, I can see it issue **100 to asterisk but failed

Hi, I'm using an Aastra 57i together with Asterisk 1.4.13. The 57i is configured for call pickup as recommended by Aastra. When the LED flashes, I press the corresponding button and the display tells me "Call not possible". In the CLI is see the follwoing output. Why isn't the call transfered to user2 and what does "No target channel found for 31 mean"? User2 can

Hello, We have a situation where a call comes in, users are notified via an external process (curl request to web service), and we can't answer the call until a callee can call in and pickup the call. How can we implement this functionality? We tried using : [caller-inbound-leg] ; code to send the CALL_UUID information to users. exten =>

I had a compile problem with the CVS I downloaded on 21 June. I have a Debian box with 2.4.18 kernel (version needed for support of Conexant ADSL). There is a difficulty with Zaptel build regarding HDLC detection. It tries to build it in and then results in unresolved kernel symbols and fails to load. I have had to comment out the entire HDLC defines in zconfig.h to get a driver to install at

Hi, I have problems while trying to dial from simple analog phone that attached to my TDM400P card. No matter which number i press i immediately get a congestion tone. when calling from outside (e.g cellphone )to the line on port 4 and pressing extension #123 everything works fine and i manage to make a connection. I've plugged on port(Zap) 4 the analog line and on port 1 the phone.

I am using AddQueueMember to add a local channel to a queue. My (simplified) dial plan is [AddMember] exten => 789,1,AddQueueMember(SomeQ|Local/456@Agent) [Queue] exten => 123,1,Queue(SomeQ|nt|||120) exten => 123,2,Hangup() exten => h,1,NoOp(InQ) [Agent] exten => 456,1,Dial(SIP/456) exten => 456,2,Set(SomeVar=SomeValue) exten => 456,3,Hangup exten => h,1,NoOp(InAgent)

I have installed asterisk in a 4.11 RELEASE FreeBSD, and we are using two Zoom X5v with SIP and works fine, we can call each other and this is OK ----------extensions.conf---------- [general] static=yes writeprotect=no [sip] exten => 123,1,Dial(SIP/123,20) exten => 123,2,Voicemail(u${EXTEN}) exten => 123,3,Hangup exten => 123,103,Voicemail(b${EXTEN}) exten => 123,104,Hangup

Hello all, We have a customer who upgraded from Asterisk 1.6 to 1.8, and pickup groups which previously worked fine have stopped working. Can anyone advise if there has been a change in how pickups work? Here is an example where 1000101 is trying to pick up a call to 1000103: <SIP/product-local-00000005>AGI Rx << EXEC Dial "Local/1000103 at product-pickup

Hi all, There is a problem with my dialplan. here is the dialplan: exten=> 123,1,Dial(SIP/U1,,Ttg) exten=> 123,2,Hangup exten=> h,1,AGI(onhangup.pl) The problem is whenever U1 is called or calls someone, if U1 hangsup the call then the h extension is NOT executed. but if the other person hangsup the call, then the h extension is executed (assuming that the other person is calling

Hello, I'm just starting out with asterisk and I'm playing around with the system. Currently I have a Digium TE210P connected to a PRI on the Asterisk server. I have a SIP soft phone on my laptop for testing that is working fine. When I try to place a call from my soft phone I get this from Asterisk: May 1 09:11:41 NOTICE[20098]: app_dial.c:1029 dial_exec_full: Unable to create

Hi, I am using asterisk1.2 In this, my dialplan is going large , so i need to configure this small pieces for this, i did in my extensions.conf when I dial the 123 its not going , means that file is not reading. is there any parameters to add any where ? please tell me this #include is not working ... extensions.conf [general] [global] trunk=zap/g0 #include exten-internal.conf [default] exten

I have a TDM400 in the server. I want to press **1XX to pickup a call. It is ok if I pickup a call dialled from 1XX to 1YY (internal network call). However, it is failed to pick up a call from PSTN thro' TDM400 card. It seems I can't guess the correct context of it. How can I know the context of the call using CLI? The default context of the TDM400 is from-pstn but pickup still