similar to: Forwarding calls and trasfer calls

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 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 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 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

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 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 To dial an outside line i have to dial 0. I want to have that when we dial local numbers, that is we are in the 08 area, I don't want to have to dial 08, how to set this up in asterisk 1.6? Regards /ralf ________________________________________________ Ralf Tr?skman, IT AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460

Hi I have a grandstream gxp-2000 and trying it on an asterisk 1.6. When I call internally between extensions I can hear the other person in the gxp2000, but when I call externally from the gxp I can't hear the person on the other end, but he can hear me. How do you configure the grandstream 2000 to work on asterisk 1.6? Regards /ralf ________________________________________________ Ralf

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:

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

Hi [Feb 17 07:59:45] WARNING[21539]: channel.c:3477 ast_request: No channel type registered for 'USTM' [Feb 17 07:59:45] WARNING[21539]: app_dial.c:1502 dial_exec_full: Unable to create channel of type 'USTM' (cause 66 - Channel not implemented) == Everyone is busy/congested at this time (1:0/0/1) I get this after I restart my asterisk 1.6, it all worked yesterday. I have the

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:

Hello, I am wondering if anyone is using the Nortel 2001 2002 or 2004 phones on their asterisk implementation. My local dealer says they are not compatible with any open source implementations. Is there a phone compatibility list somewhere? Cheers Cian

Hello, I am trying to configure new opus codec in asterisk 14, but unable to find any examples of codecs.conf settings for this codec. All I am trying to do - setup peer with using opus in narrow band mode (8kHz sampling rate). Does anybody know how to configure chan_opus? -- Regards, Igor Goncharovsky Unistim Dev: http://unistim.igorg.ru -------------- next part -------------- An HTML

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

Hello, I use a Suse 11.0 as a Samba 3.2 PDC. The clients run XP SP3. I have upgraded a few weeks ago from Suse 10.3 and now all files tranfer that I do - for example, downloading a file using a web browser - it leaves a trash file named "transferd-file:Zone.Identifier" or "tranferd-file:encryptable". The odd thing is that from Windows I can`t read/write/delete these files.

Hi i've read this post http://lists.digium.com/pipermail/asterisk-dev/2007-May/027666.html I just want to know if there are some upgrades... on 1.4 or 1.2. I'd like to store two records in the CDR instead of one, when a call is transferd. Is it possibile now? Thanks to all -- /*************/ nik600 https://sourceforge.net/projects/ccmanager

Guys. This idea has been banging my headfor days now and I feel the need to share with you. Imagine this scenario: all calls come in thru a receptionist, asterisk records all incoming calls, the receptionist's work is to transfer the calls to internal people but some of them are bosses and you know how bosses are, they don't want their calls to be recorded, so, I have been trying to

Hello, Cedric Hans has released an UNISTIM channel driver for asterisk (stable). You can download it at : http://mlkj.net/asterisk/chan_unistim-0.9.2.tar.bz2 Copy of README : This is a channel driver for Unistim protocol. You can use at least Nortel i2004 phones with it. Only few features are supported : Send/Receive CallerID, Redial, SoftKeys, SendText(), Music On Hold, Message Waiting