similar to: Blind transfer from Queue in AGI script failuire

Hi I have this in my queue extension and I see this in asterisk when I call to the queue, but no file is created in the directory any ideas? exten => s,1,Set(MONITOR_FILENAME=/var/spool/asterisk/queuecalls/QSAMPLE-${UNIQUEID}) -- Executing [s at kundservice:1] Set("SIP/0850001175-b7942770", "MONITOR_FILENAME=/var/spool/asterisk/queuecalls/QSAMPLE-1228382046.12") Regards

Can't we use pattern matching inside a macro? Because when I am trying to do so call is terminating even for a very simple dummy dialplan. [demo3] exten=>98,1,NoOp() exten=>98,2,Macro(testme) exten=>h,1,NoOp(terminating call); [macro-testme] exten=>s,1,Playback(Digits/2) exten=>s,2,WaitExten(15) exten=>s,3,NoOp() exten=>_X,1,NoOp(${EXTEN}) exten=>_X,2,Goto(s,3)

So I have a problem. A customer of mine wants a PBX, owns an office building. I want to sell him on asterisk. He has 4 tenants. I am using my asterisk box to simulate it. My asterisk box has a TDM400P card, not a PRI card. Don't know if it makes any difference. Anyway, I want to route incoming phone calls to different contexts based on the phone number being called. Here is my

I've had good luck using the llvm-gcc & llvm-g++ on small projects, but I just discovered that it's apparently missing some of the GCC intrinsic functions -- specifically, when I try and compile VXL (http://vxl.sourceforge.net) it dies when it encounters __builtin_bswap32 . This is on OS X with the llvm-gcc-4.2 & llvm_g++-42 that are part of the XCode 3.2.3 I don't know if

Hi Kent, I suggest you open a bug report with a preprocessed testcase. Best wishes, Duncan. > I've had good luck using the llvm-gcc& llvm-g++ on small projects, > but I just discovered that it's apparently missing some of the GCC > intrinsic functions -- specifically, when I try and compile VXL > (http://vxl.sourceforge.net) it dies when it encounters >

Hi, Running Redhat 7.3 with SAMBA. I am trying to have a shared folder that everyone in the office can use an write to. I have one problem, as users open a file, it changes the user flag and only that user can open it in the future. All users are in the same group = Users Please help Regards Michael Crocombe Hi, If I can be of any further assistance, please don't hesitate to contact

Hi! Maybe someone could help me out? When a call is routed via a2billing AGI and user does a transfer, the call is dropped. If the trunk is called directly everyhing works. Here's a direct scenario (working fine): [pbx000001] exten => 101,1,Set(__TRANSFER_CONTEXT=pbx000001) exten => 101,n,Dial(SIP/pozitel/37129238254,45,t) exten => 102,1,Dial(SIP/12345,60) so, when user calls ext

Hi Our sip provider has two servers that sends calls to our asterisk 1.6. When server 1 sends call everything is working, but when server 2 sends call I get [Jan 13 14:56:23] NOTICE[16680]: chan_sip.c:16869 handle_request_invite: Call from '' to extension '0840303390' rejected because extension not found. And the provider get an "404 not found" error on their side. What

Hi How do i set up so that everyone can dial, for example *21* to forward all calls to a cellphone or another extension and how do I enable so that cals can be transferd between extentions. I use asterisk 1.6 and have my phones in unistim.conf and my extensions in extensions.conf. Regards /ralf ________________________________________________ Ralf Tr?skman, IT AdLibris AB, Odengatan 106, 113

Hi When i try to transfer calls from my ip2002 phone in asterisk 1.6, I can make the transfer and it rings on the extension I transfer to, but when I accept the call, asterisk dumps. How can I get it to work? And how do I save the dump error? Regards /ralf ________________________________________________ Ralf Tr?skman, IT AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden Dir:

Hi I want to be able to use one provider if I dial 0 before the number and another if I dial 1 before, how can I do that in asterisk 1.6? /ralf Ralf Tr?skman, IT AdLibris AB, Box 3667, 103 59 Stockholm. Bes?ksadress: Sveav?gen 56C, 111 34, Stockholm - Obs ny address! Dir: +46-(0)8-5460 60 91, mob: +46-(0)70-7548074, vxl: +46-(0)8-5460 60 00, fax: +46-(0)8-5460 60 99 ralf at

Hi I have asterisk 1.6 and running queues with realtime mysql. I am trying to set another musiconhold then "default" but I cant get it to work, I have an musiconhold entry in my queue_table, but don't know what to put in there and where to put the file. Regards /ralf ________________________________________________ Ralf Tr?skman, IT AdLibris AB, Odengatan 106, 113 22 Stockholm,

Hi We runs asterisk 1.6 on a ubuntu 8.04 server. How can I get asterisk to start at boot? I have created an file named asterisk in /etc/event.d and put in this # This service maintains Asterisk from the point the system is # started until it is shut down again. description "Asterisk daemon" start on runlevel-2 stop on shutdown respawn exec

Hi I have an asterisk 1.6 running, and our provider have an openser on their end. When I get an incoming call I get this on my end [Jan 8 14:51:56] NOTICE[16680]: chan_sip.c:16869 handle_request_invite: Call from '' to extension '0840303395' rejected because extension not found. If I wait approx a minute and try again, the call will go trough. We don't use REGISTER or

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, We just released Opus 1.1-beta and Monty wrote a nice demo showing off all the new features, including many improvements over the alpha release: http://people.xiph.org/~xiphmont/demo/opus/demo3.shtml As usual, the code can be downloaded from: http://opus-codec.org/downloads/ In addition to 1.1-beta, we also released 1.0.3, which includes a

After more than two years of development, we have released Opus 1.1. This includes: * new analysis code and tuning that significantly improves encoding quality, especially for variable-bitrate (VBR), * automatic detection of speech or music to decide which encoding mode to use, * surround with good quality at 128 kbps for 5.1 and usable down to 48 kbps, and * speed improvements on all

I executed rsync --progress --verbose --stats --recursive test@100.100.100.100::test /home/test2 --stats but each time it showed me files are being transfer but actually the files is not changed or updated any. for example, 12 100% 0.00kB/s 0:00:00 abcdef 44 100% 0.00kB/s 0:00:00 demo2 0 100% 0.00kB/s 157:26:58 demo3 0 100% 0.00kB/s

We have 40 nortel ip2002 phones connected to asterisk 1.6, the problem I have is that the phone looses the connections with the server and then drops calls, we can reconnect but the customers don't like it. Anyone has the same problem? /ralf ________________________________________________ Ralf Tr?skman, IT AdLibris AB, Sveav?gen 56C, 111 34 Stockholm, Sweden Dir: +46-(0)707548074, vxl:

Hi Im using a sip phone SPA921, and the one that calls me can hear me but I cant hear them, when I make the call I can hear them. Im running asterisk 1.6 behind a firewall, I have port 10000-20000 for rtp and 5060 for sip forward to my asterisk. Any tips? Regards /ralf ________________________________________________ Ralf Tr?skman, IT AdLibris AB, Sveav?gen 56C, 111 34 Stockholm, Sweden Dir:

Hi Is there a user client that a group, like customer service can use? We have today an avaya IP-office with phonemanager pro and I want something equal to phonemanager pro, where you can logon to ques and see how many calls is in that queue and so on. Regards /ralf ________________________________________________ Ralf Tr?skman, IT AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden Dir: