RE: [asterisk-users] command like break ore exit in the dialpan
I don't know if I understand you correctly but you could place a Goto or a Hangup there: exten = 99,1,Gotoif(?2:4) exten = 99,2,Meetme(100) exten = 99,3,Goto or Hangup exten = 99,4,Meetme(100|options) -Original Message- From: nik600 [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 16, 2007 1:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] command like break ore exit in the dialpan Hi i have a similar dialplan: exten = 99,1,Gotoif(?2:3) exten = 99,2,Meetme(100) exten = 99,3,Meetme(100|options) i'd like to do something like: exten = 99,1,Gotoif(?2:4) exten = 99,2,Meetme(100) exten = 99,4, ... exit ... exten = 99,3,Meetme(100|options) How can i exit the dialplan? I won't use meetme twice! Thanks nik ___ --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 ___ --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] AEL: CID match and pattern in switch statement
Hi all,
I am using Asterisk 1.2.10 on Debian Sarge and currently I am rewriting my
extensions.conf with ael.
The replacement of the following part makes me mad:
[set-language]
exten = _X./_0031.,1,Set(incoming_call=1|lang=nl)
exten = _X./_0031.,2,Goto(incoming,${EXTEN},1)
exten = _X./_0049.,1,Set(incoming_call=1|lang=de)
exten = _X./_0049.,2,Goto(incoming,${EXTEN},1)
exten = _X.,1,Set(incoming_call=1|lang=en)
exten = _X.,2,Goto(incoming,${EXTEN},1)
First I tried it this way:
context set-language {
_X./_0031. = {
Set(incoming_call=1|lang=nl);
jump [EMAIL PROTECTED];
};
_X./_0049. = {
Set(incoming_call=1|lang=de);
jump [EMAIL PROTECTED];
};
_X. = {
Set(incoming_call=1|lang=en);
jump [EMAIL PROTECTED];
};
};
The CID match did not seem to work so I tried to solve it with a switch
statement:
context set-language {
_X. = {
Set(original_extension=${EXTEN});
switch (${CALLERID(num)}) {
pattern 0031.:
Set(incoming_call=1|lang=nl);
jump [EMAIL PROTECTED];
pattern 0049.:
Set(incoming_call=1|lang=de);
jump [EMAIL PROTECTED];
default:
Set(incoming_call=1|lang=en);
jump [EMAIL PROTECTED];
};
};
};
Unfortunately pattern does not work at all. I would be very happy if someone
can enlighten me what I am doing wrong here.
Are there any other solutions for replacing that part of my extensions.conf?
Is there any AEL documentation available in addition to the information in
the wiki and doc/README.ael?
Regards, Jens
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RE: [Asterisk-Users] Siemens Gigaset SL75
No, the Gigaset is the only WLAN phone I tested so long, so I can not compare it to the other phones you mentioned. -Original Message- From: Olivier [mailto:[EMAIL PROTECTED] Sent: Friday, November 24, 2006 10:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Siemens Gigaset SL75 Have you ever compared it to Linksys WIP 330 or Zyxel 2000 ? Those 2 seem to get average reviews from users (short range, autonomy, ...). On paper, it seems to me a decent WiFi phone is still lacking today. Maybe this Gigaset SL75 could fill the void. ___ --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] Siemens Gigaset SL75
I bought the phone in Germany. Except another wlan phone from Siemens which was not available any more, I did not find any alternatives to it. -Original Message- From: Olivier [mailto:[EMAIL PROTECTED] Sent: Wednesday, November 22, 2006 8:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Siemens Gigaset SL75 This phone seems attractive but is not distributed in France. I wondered the reasons behind that. Just for curiosity, in which country did you buy it ? How would you compare it to alternatives ? Cheers 2006/11/21, [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] : Hi, yes I tested this one week ago and it worked without problems. It is a nice wlan-phone with some (in my opinion) unnecessary features. Regards, Jens -Original Message- From: Olivier [mailto: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ] Sent: Friday, November 17, 2006 10:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Siemens Gigaset SL75 Hi, Has anyone tested Siemens Gigaset SL75 with Asterisk ? How would you rate its performances ? Cheers ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Extension Matching with Match As You Go Dialing
Hi all, when calling from the PSTN with Match As You Go Dialing (lift the handset before start to dial) over a zap channel Asterisk simply takes the first extension digit and tries to match it. Because no valid one-digit-extension exists in my dialplan, matching fails and Asterisk says that a invaild extension was dialed. On voip-info.org I read that zap channels have a fixed timeout period of 3 seconds. But in my case Asterisk does not wait even 100 ms before taking the invalid extension. How can I tell Asterisk to wait for more extension digits, especially when the first digit of the dialed extension is no vaild extension number? Thanks in advance. Regards, Jens ___ --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] Asterisk stopps matching extensions after first digit
Hi all,
I have problems receiving calls from PSTN with an Wildcard T207P.
All internal SIP devices have a 3 digit extension, e.g. 873.
When I call the extension from the PSTN this way everything works fine:
1. enter the number on the phone
2. lift off the handset
But when I call it that way Asterisk stopps matching the extension after the
first extension digit (8 in that case):
1. lift off the handset
2. enter the number on the phone
Asterisk then says that the extension does not exist and that the call is
rejected.
The context for receiving incoming calls in extensions.conf looks like this
[zap-in]
exten = _999[8-9]XX,1,Goto(internal,${EXTEN:3},1)
Is it possible to tell asterisk that it should match only 3 digits
extensions and where can this be configured? (extensions.conf, zapata.conf
or anywhere else)
Thanks in advance
Regards, Jens
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[asterisk-users] Got SUBSCRIBE for extensions without hint
Hi all,
for the user with extension 191 I get the following error message:
ERROR[10131]: chan_sip.c:10831 handle_request_subscribe: Got SUBSCRIBE for
extensions without hint. Please add hint to 191 in context internal
I don't know where to add something because the entries in sip.conf look the
same for every user and in extensions.conf a pattern for all internal users
is used:
[internal]
exten = _[1-7]ZX,1,Dial(SIP/${EXTEN},5,Ttr)
exten = _[1-7]ZX,2,Set(original_extension=${EXTEN})
exten = _[1-7]ZX,3,Goto(status-${DIALSTATUS},1)
exten = status-BUSY,1,VoiceMail(b${original_extension})
exten = status-BUSY,2,Playback(vm-goodbye)
exten = status-BUSY,3,Hangup()
exten = status-NOANSWER,1,VoiceMail(u${original_extension})
exten = status-NOANSWER,2,Playback(vm-goodbye)
exten = status-NOANSWER,3,Hangup()
Why do I get this message for the user with extension 191?
Thanks for your help.
Regards, Jens
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[Asterisk-Users] Hardware to connect analog and ISDN fax devices
Hi all, I've read a lot of problems with faxing over asterisk. Most of them referred to Fax over Internet, but I want to connect analog and ISDN fax devices to asterisk to send and receive faxes over PRI: +-+ +--+++ | | | || ISDN Fax | | PRI |-| Asterisk |++ | | | || Analog Fax | +-+ +--+++ Can this be done without problems and can I use Digium cards to accomplish this? Or do I need other hardware devices? Regards, Jens ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Transfer does not work
Hi !
I am trying to transfer calls between internal SIP softclients, but it does
not work. Every time I press a key on the softclient, the CLI shows the
following output:
Attempting native bridge of SIP/456-9ee0 and SIP/173-f586
This is my extensions.conf:
[macro-voicemail]
exten = s,1,Dial(${ARG1},5,Ttr)
exten = s,2,Goto(status-${DIALSTATUS},1)
exten = status-BUSY,1,VoiceMail(b${MACRO_EXTEN})
exten = status-BUSY,2,Playback(vm-goodbye)
exten = status-BUSY,3,Hangup()
exten = status-NOANSWER,1,VoiceMail(u${MACRO_EXTEN})
exten = status-NOANSWER,2,Playback(vm-goodbye)
exten = status-NOANSWER,3,Hangup()
[internal]
exten = _ZXZ,1,Macro(voicemail,SIP/${EXTEN})
And this is the part of the features.conf I changed (just uncommented that
part)
[featuremap]
blindxfer = #1 ; Blind transfer
disconnect = *0; Disconnect
automon = *1 ; One Touch Record
atxfer = *2; Attended transfer
None of the shortcuts in [featuremap] works.
What am I doing wrong?
Regards, Jens
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