similar to: Asterisk user client for customer service

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 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 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 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 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 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 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 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

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 [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

Buenas noches; Además del proyecto que comenté antes y con el que sigo discutiendo, también me estoy peleando con otro... con el que también tropiezo. Necesito reordenar los datos presentados en dos columnas a filas y columnas. Los datos ideales serían algo así: Ques <- c(rep("Q1",3),rep("Q2",3),rep("Q3",3)) Info <- rep(c("aca", "ahi",

Hello list we are installing 2 new servers (to run asterisk) shortly, for a "stand alone" service. Ignoring our current naming convention, we'd like to name them something.. but we are not sure what. a consideration is that on the screens of the phones it shows extension@hostname (eg 3001@telephony) (all extensions are numeric) so the users will see it everyday i'm not

hello i was uploading drivers and renamed a printer by error with the name of an other existing one. now, when i rename one of them or click-right and ask for properties, it's always the first printer that is renamed or that i see the properties of. so i cant fix my mistake. has anybody the same problem ? how should i fix that ? thanks ELH -- ?ric LE H?NAFF ?cole normale sup?rieure -

Actually when i try to access a question using serel gem(which is a ruby wrapper for stackoverflow api), i get an object with no body parameter included in it. questions = Serel::Question.find(id) see http://api.stackoverflow.com/1.1/usage/methods/questions-by-ids. According to this, we need to turn the body parameter true. How to turn the parameter "body" true. -- You received

hello i cant manage groups with usrmgr and the configuration below debian sarge, samba 3.0.22, openldap 2.2.23, smbldap-tools 0.8.7. usrmgr reports he cant find the groups. is it solved by samba 3.0.23a ? does anybody report enhancement after upgrading to 3.0.23a ? Regards ELH -- ?ric LE H?NAFF ?cole normale sup?rieure - Centre de ressources informatiques Informaticien, Ing?nieur d?veloppements

Summary: If you use ZFS, do not downgrade from build 35 or later to build 34 or earlier. This putback (into Solaris Nevada build 35) introduced a backwards- compatable change to the ZFS on-disk format. Old pools will be seamlessly accessed by the new code; you do not need to do anything special. However, do *not* downgrade from build 35 or later to build 34 or earlier. If you do so, some of

Dear list, I have the following data set id 1 2 3 4 5 6 7 8 9 10 disease a b c d e f g h i j age 23 40 32 34 25 32 22 35 29 21 city NY LD NY SG NY LD VG SA LD SG sex 1 1 2 2 2 2 1 1 1 2 treat_a y y y y treat_b n n n n n n ques1_1 2 4 5 6 8 3 1 2 4 5 ques1_2 6 4 5 12 10 9 8 4 5 7