[Asterisk-Users] Dynamic features on call waiting
Hello, I have problems using dynamic features while an other person is doing call waiting in a call. I define a dynamic application mapping in features.conf as the following: testfeature = *9,caller,Playback,tt-monkeys I also set DYNAMIC_FEATURES = testfeature. The mapping is working well. But during a third person is calling I'm hearing just the call waiting tone and none of my mapped features are working for this time. How can I change this behaviour? (I'm using Asterisk 1.2.4-BRIstuffed-0.3.0-PRE-1l) Thank you in advance, Henry ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Configuring behaviour of flash hook
Hi all, I have some problems configuring the right behaviour of Zaptel devices in asterisk. Especially with three way call, call waiting and call transfers. Having one party on hold and the other on line I would like to have - Flash Hook + 3 to activate three way call - Flash Hook + 2 to swap between user on hold and user on line. On call waiting I would like to have: - Flash Hook + 2 to accept waiting call - Flash Hook + 0 to reject it Having one party on hold while talking with an other I would also like to have: - Flash Hook + 1 to disconnect from the current call. I know that some of these features are configured in features.conf, but always totally without the use of flash hook. Is there any way to configure this behaviour in asterisk? Thanks in advance :) Henry ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Setting MSN for outgoing ISDN calls
Hi all, I have a problem setting the MSN for outgoing calls. I'm using a HFC PCI card together with zaptel and bristuff. All my outgoing calls are using the same (first|default|main) MSN. In my zapata.conf I tried different values for pridialplan, prilocaldialplan, nationalprefix, etc but without any success. In my extension.conf I'm setting the MSN with: exten = _X.,1,SetCIDNum(MSN) exten = _X.,2,Dial(Zap/g2/Number,60, T) What are the right values for pridialplan for Germany? Is setting the MSN with SetCIDNum the right way? Would be fine if someone could provide me a working zapata.conf. Thanks in advance, Henry -- Hi! I'm a .signature virus! Copy me into your ~/.signature to help me spread! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Setting MSN for outgoing ISDN calls
Answering my own question. It worked with prilocaldialplan=local, pridialplan=unknown and running CallerPres before every Dial command. Henry -- Hi! I'm a .signature virus! Copy me into your ~/.signature to help me spread! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Outbound Caller ID number on E1
Hi,
I just had the same problem (see post Setting MSN for outgoing ISDN
calls).
It was very helpful to enable pri debug (pri debug span X). Just try
different values for pridialplan, prilocaldialplan. And also try to do
CallerPres right before the Dial command.
How do you set your CallerID or MSN? I just do:
exten = _X.,1,CallingPres(0)
exten = _X.,2,SetCIDNum(123456)
exten = _X.,3,Dial(Zap/g1/${EXTEN},60, T)
exten = _X.,4,Busy()
123456 is my number without area code. (prilocaldialplan=local).
Hope that helps,
Henry
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Re: [Asterisk-Users] Outgoing FXO and CDR
On Fr, 2006-01-27 at 14:06 +0100, Matt Riddell (IT) wrote: If you are the USA, you can try to use callprogress=yes in zapata.conf, but the warnings above the entry still stand. I will try callprogress but I thought it is just there for checking if the other side hung up? Also I'm not in USA does that mean callprogress will not work for me in Germany/Europe at all? Henry -- Hi! I'm a .signature virus! Copy me into your ~/.signature to help me spread! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Outgoing FXO and CDR
Hi all, When I do outgoing calls via my FXO card (TDM400, analog line), they get always marked ANSWERED in my CDR. I guess it is not that easy for fxo to determine if there is actually a call or just ringing. But anyway, is there a way to get this working right? Thanks in advance, Henry ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Three Way Calling with HFC PCI Card
Hello, On Di, 2006-01-03 at 16:31 +0100, Giovanni Miano wrote: Use meetme app Unfortunately meetme is no solution for me. If nobody can help me, is there at least anybody who has the same problem? As far as I can see there are lots of people using the HFC PCI card, is nobody using Three-Way-Calling? It would be really helpful to know if the problem is with zaptel +asterisk or just with my setup. Thanks in advance :) Henry -- Hi! I'm a .signature virus! Copy me into your ~/.signature to help me spread! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Three Way Calling with HFC PCI Card
Hello, I'm using a TDM 400 together with two HFC PCI Cards in my Box. Three Way Calling with a SIP or analog Phone is working perfectly. But if I try to do Three Way Calling with my ISDN Phone I get an error message: Facility Name requested on channel 0/2 not in use on span 1 I use bristuff with my HFC card and don't know why I get this message? I'm using still asterisk 1.0 and can not update to the newest version at the moment. Is there a simple trick to make it work or is this problem already solved in asterisk 1.2? Thanks in advance, Henry -- Hi! I'm a .signature virus! Copy me into your ~/.signature to help me spread! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
