SV: [Asterisk-Users] GotoIf problem
-Oprindelig meddelelse- Fra: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] På vegne af kurt x Sendt: 9. marts 2005 20:57 Til: Chris Wade Cc: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [Asterisk-Users] GotoIf problem I,ve gotten the GotoIf statement working now. I hard coded the value 10 in place of the ${DIGITS} varible. Worked like a charm. Thanks to everyone who helped. Kurt Hi Kurt, You are writing the ${DIGITS} variable wrong, you are missing a { eg.: you are writing $DIGITS} and it should be ${DIGITS} Thorben ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: [Asterisk-Users] Snom phone hint exten question
I haven't used it in a while, but I had to put subscribecontext=sip for the phone's (in your case the snom) sip entry. This seems like it has been removed from the wiki. Has it changed or is this incorrect? Hi James, I have just found out that all you need to do is make the hint in the context where the phone registers. That means that all you need to do is put '690,hint,SIP/bt-karen' in your [sip] context, nothing else and it should work. Remember to take the power from the phone for a short while after you have configured this, otherwise it won't work. thorben ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: [Asterisk-Users] Snom phone hint exten question
Ok your example confused me a little. You put 690,hint,SIP/bt-karen From this section in my extensions from your example I should have exten = 690,hint,SIP/bt-karen exten = 691,hint,SIP/snom-james So set hint on the opposite extensions? [sip] exten = 690,hint,SIP/snom-james exten = 690,1,SetMusicOnHold(random) exten = 690,2,Dial(SIP/snom-james,30,Ttr) exten = 690,3,Voicemail2,u690 exten = 690,103,Voicemail2,b690 exten = 691,hint,SIP/bt-karen exten = 691,1,SetMusicOnHold(random) exten = 691,2,Dial(SIP/bt-karen,30,Ttr) exten = 691,3,Voicemail,u691 exten = 691,103,Voicemail,b691 Hi James I am sorry I made a typo. You need to set [sip] like this: [sip] exten = 690,hint,SIP/snom-james exten = 691,hint,SIP/bt-karen exten = 690,1,SetMusicOnHold(random) exten = 690,2,Dial(SIP/snom-james,30,Ttr) exten = 690,3,Voicemail2,u690 exten = 690,103,Voicemail2,b690 exten = 691,1,SetMusicOnHold(random) exten = 691,2,Dial(SIP/bt-karen,30,Ttr) exten = 691,3,Voicemail,u691 exten = 691,103,Voicemail,b691 That should work thorben ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Snom phone hint exten question
I am going to now sit in a corner and go quietly insane while playing the banyo with no strings. Still doesn't work, I dialed in an outside line and picked up the receive on extension 691, yet the light on the snom phone did not come on. I dialed out of extension 691 to an outside line, yet still the light did not come on. Snom190 has firmware 3.56m the button is set to Destination 691 Hi James, I am using the latest CSV-HEAD of *, I do not think it works with * stable. Thorben ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: [Asterisk-Users] Snom phone hint exten question
-Oprindelig meddelelse- Fra: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] På vegne af James Bean Sendt: 19. februar 2005 08:14 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: [Asterisk-Users] Snom phone hint exten question Hi, I am sorry to be asking this but the wiki is down and has been for a couple of days and I need to get this working before Monday to get my live system setup. Trying to get the Snom 190's and soon to arrive 3com 3102's to use the function keys and for the life of me I can't work it out from the conversations on the archive what I am going exactly wrong here? The snom 190 with function keys is extension 690, the other extension (691) is just a BT102 so it doesn't have any function keys to program. When extension 691 is dialing out, or receives a call I want it to just tell the snom190 on ext 690 so the light shows up. (Soon as I got it going here I have a live system I will be setting it up on). Thank you to anyone in advance for the help. This is my extensions.conf --- [pstn] exten = s,1,SetMusicOnHold(random) exten = s,2,Dial(SIP/snom-jamesSIP/bt-karen,45,t) exten = s,3,Hangup ;exten = s,10,VoiceMail(u100);Whatever box you want. [internal] exten = i,1,Playback(invalid) exten = i,2,Hangup exten = t,1,Hangup exten = 098,1,WaitMusicOnHold(5) exten = 099,1,Echo ;simple echo test when you dial 099 on your phone include = outgoing include = sip [outgoing] exten = _9X.,1,Dial(Zap/g1/${EXTEN:1}) exten = _9X.,2,Congestion() exten = _9X.,3,Hangup [sip] exten = 690,2,SetMusicOnHold(random) exten = 690,3,Dial(SIP/snom-james,30,tr) exten = 690,4,voicemail2,u690 exten = 690,102,voicemail2,b690 exten = 691,hint,SIP/bt-karen exten = 691,1,Macro(stdexten,SIP/bt-karen) exten = 691,2,SetMusicOnHold(random) exten = 691,3,Dial(SIP/bt-karen,30,tr) exten = 691,4,voicemail2,u691 exten = 691,102,voicemail,b691 include = internal include = outgoing Hi, You need to 'hint' SIP/bt-karen in the pstn context: [pstn] exten = s,hint,SIP/bt-karen exten = s,1,SetMusicOnHold(random) exten = s,2,Dial(SIP/snom-jamesSIP/bt-karen,45,t) exten = s,3,Hangup Thorben ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Snom phone hint exten question
Unfortunately that did not work, I hard rebooted the snom phone, the bt102 and the asterisk server, the light just stays off, and I tested the LED on the button as well just to make sure its working I also added a hint to the outgoing context so when they make an outgoing call, still no luck. My extensions.conf is now [pstn] exten = s,hint,SIP/bt-karen exten = s,1,SetMusicOnHold(random) exten = s,2,Dial(SIP/snom-jamesSIP/bt-karen,45,t) exten = s,3,Hangup ;exten = s,5,VoiceMail(u100);Whatever box you want. [internal] exten = i,1,Playback(invalid) exten = i,2,Hangup exten = t,1,Hangup exten = 098,1,WaitMusicOnHold(5) exten = 099,1,Echo ;simple echo test when you dial 099 on your phone Include = outgoing include = sip [outgoing] exten = _9X.,hint,SIP/bt-karen exten = _9X.,1,Dial(Zap/g1/${EXTEN:1}) exten = _9X.,2,Congestion() exten = _9X.,3,Hangup [sip] exten = 690,1,SetMusicOnHold(random) exten = 690,2,Dial(SIP/snom-james,30,tr) exten = 690,102,voicemail2,u690 exten = 691,hint,SIP/bt-karen exten = 691,1,SetMusicOnHold(random) exten = 691,2,Dial(SIP/bt-karen,30,tr) exten = 691,102,voicemail,u691 include = internal include = outgoing Have you set the function key on the SNOM to 'Destination' and typed '691' in the number? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Snom phone hint exten question
Have you set the function key on the SNOM to 'Destination' and typed '691' in the number? I am sorry, I meant that you have to type 'bt-karen' in the number. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: [Asterisk-Users] Snom phone hint exten question
Also instead of putting a whole bunch of hints in, how might I go about putting a cluster of SIP extensions in the hint off the PSTN situation? Could you also maybe throw me a couple of hints what the exten = 691,1,Macro(stdexten,SIP/bt-karen) Macro portion I have seen in some examples but I am not sure what it does. James Hi James, I think you have to put a whole bunch of hints in, as variables are not supported with hint AFAIK. Macro(stdexten,SIP/bt-karen) is the way to call a macro called 'stdexten' in your dial plan. Thorben ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] wiki down?
Yes - no go from here -Oprindelig meddelelse- Fra: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] På vegne af Roy Sigurd Karlsbakk Sendt: 19. februar 2005 19:14 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: [Asterisk-Users] wiki down? hi is the wiki down again? roy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Setting a Forward to an external number onyourphone
[EMAIL PROTECTED] wrote: Hi! Maybe I have just been looking on the wrong pages but there is a question that is very important for me. I already studied some Demo-Dialplans and made some basic experiences with Asterisk. But what I need to find out is how I can handle this. I am leaving my office and I want to tell asterisk to forward calls now to my mobile phone by just hitting a key (on my IP-Phone) or by using a special key-sequence. How can this be handled because I need to change the dialplan based on some information coming from a device attached to a channel. When back in my office I hit the key again and the calls are now routed to my IP-Phone (or ISDN-Phone on zap-channel) again. With IP-Phones I can imagine just unregistering the phone and having a dialplan with a fallback-option or something like that. But what if I want to tell asterisk to forward calls from now on to a number I want to manually add just for today (hitting a key, entering the new target number and that's it). where can I find some information on how to make this feature available. This is the way we do it, we have it call the local extension as well as a mobile number, and then you choose where to pick up: [applications] exten = _*22*.,1,Answer() exten = _*22*.,2,DBput(DUAL/${CALLERIDNUM}=${EXTEN:4}) exten = _*22*.,3,PlayBack(call-fwd-parallel) exten = _*22*.,4,PlayBack(activated) exten = _*22*.,5,Hangup exten = **22,1,Answer() exten = **22,2,DBdel(DUAL/${CALLERIDNUM}) exten = **22,3,PlayBack(call-fwd-parallel) exten = **22,4,PlayBack(de-activated) exten = **22,5,Hangup [macro-dial] exten = s,1,DBget(DUAL=DUAL/${ARG2}) exten = s,2,Dial(${ARG1}${OUTGOING}/${DUAL},20,wW) exten = s,3, VoiceMail(su${ARG2}) exten = s,102,Dial(${ARG1},20,Ww) exten = s,103,VoiceMail(su${ARG2}) exten = s,203,VoiceMail(sb${ARG2}) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TOUCH_MONITOR
TOUCH_MONITOR is the variable to set if I need to specify my own options for One Touch Record (filetype|filename|m). I cannot get it to work. Can you help? Thanks in advance ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: [Asterisk-Users] Newbie question
Hi Tim, You could put he call back into the queue when the dial times out. Check for the length of the CALLERID, if it's equal to the length of your internal numbers then goto voicemail otherwise goto the queue. thorben -Oprindelig meddelelse- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Tim De Lange Sendt: 18. februar 2005 14:25 Til: asterisk-users@lists.digium.com Emne: [Asterisk-Users] Newbie question Hello! When the oprator transfers calls to internal extensions to unavailable or busy extensions, how can I prevent these calls from going to voicemail, and route them back to the oprator? But other calls, ie internal between extensions, and calls coming in via DID should get voicemail if extensions are busy / unavailable? Any help be appreciated. TIA! Tim ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: [Asterisk-Users] Flash Pane - Monitor Parked Calls?
Need help with how to configure for parked calls in the Flash Operator Panel's op_buttons.cfg file ... I've looked on the wiki, google and asternic's site and can't seem to find how to setup op_buttons.cfg to monitor parked calls. For example, if someone parks in 701, I'd like to see that represented on the panel. I've tried a number of things ... this is what I have now and it does not work ... [701] Position=12 Label=Park 701 Extension=701 Context=parkedcalls Icon=1 Any help would be great! Thanks, Bruce [EMAIL PROTECTED] Hi Bruce, Try this; I took this from the sample configuration: [PARK701] Position=17 Icon=3 Extension=700 Label=Park 701 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users