[asterisk-users] Backgroung usage
Hello, I try to use the background cmd for send incomings call on dial plan. I try in an internal number for resting: exten = 405,1,DigitTimeout,5 exten = 405,2,ResponseTimeout,10 exten = 405,3,Background(vm-accueilcreat) exten = 1,1,Goto(creat-in,s,1) exten = 2,1,Dial(IAX2/301,15,tr) exten = 3,1,Hangup But nothing happen when i hit 1, 2, or 3. Wher is the mistake?? Best regards, -- Olivier Saulnier STEGANUX 1er étage DIAMECANS BEL AIR 03410 St-Victor T: 04.70.02.27.62 F: 04.70.09.97.41 http://www.steganux.com ___ --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] AGI and some informations
Ok, i understand,
But i don't know how to get the IP Adress when a softphone is registred,
and how to send to this IP adress, and call number to the softphone, for
an incoming call.
best regards,
Olivier S
Anton Frolov a écrit :
it was not a real code, but just a schema.
I can't write a more precise snippet of code, since I'm completely
unaware of your configuration.
But in any case, I would delegate all of the logic to your Ruby script
and keep the minimum in the extensions.conf.
AF.
Olivier Saulnier wrote:
Anton Frolov a écrit :
When registering the softphone:
OK, in which file do i do that??
SoftPhonesDB.insert(olivier, $ip);
Could you explain me what means SoftPhonesDB.insert?? It's not an AGI
commande, how can i use it??
In extensions.conf:
exten = 302,s,agi,script.rb,${EXTEN}
OK, i understand
In script.rb:
$ip = SoftPhonesDB.select(olivier);
Dial(SIP/$ip, $arg);
Hummm...Does Dial working with Ip adress?? Or should i write ine the
database the softphone number too??
Best regards,
Olivier S
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Olivier Saulnier
STEGANUX
1er étage DIAMECANS
BEL AIR
03410 St-Victor
T: 04.70.02.27.62
F: 04.70.09.97.41
http://www.steganux.com
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Re: [asterisk-users] AGI and some informations
Hello, I'll try to explain better what i want to do: 1 - i've developped a softphone, with iaxclient.dll, in Windev Langage (French langage - PC SOFT). This DLL doesn't work well with Windev, some of pieces are ok. the ring, is not OK!! 2 - For detect the ring, i have make a listen server on the softphone. On the Asterisk server, i've make a Ruby Script which give the information as the RING must bell, and when it must stop. 3 - I modify the extensions.conf file fir call the Ruby script. but it's not a good work, because: a) I must specify for each softphone a new context, where i call the script with the IP address. b) i must give at softphone the phone nimber incoming, for external calls Do you have any idea for process that?? Best regards, -- Olivier Saulnier STEGANUX 1er étage DIAMECANS BEL AIR 03410 St-Victor T: 04.70.02.27.62 F: 04.70.09.97.41 http://www.steganux.com ___ --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] AGI Script with parameters
Hello, I want to use AGI for give some information for a softphone, as: exten = 0470022762,2,AGI(/ruby/ring.rb 192.168.0.10 5010) We use Ruby langage. The line doesn't worksin as this, but works with shell command. Also, if i modify my ruby script for give in the code the ip adress and port number, it works! Do you have any idea?? best regards, -- Olivier Saulnier STEGANUX 1er étage DIAMECANS BEL AIR 03410 St-Victor T: 04.70.02.27.62 F: 04.70.09.97.41 http://www.steganux.com ___ --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] AGI Script with parameters
hello, I try: exten = 0470022762,2,agi,/ruby/ring.rb|192.168.0.10 5010 And it doesn't works again... best regards, Olivier S Anton Frolov a écrit : bonjour, Olivier I think that your line should be something like: exten = phone,2,agi,script.py|args -- Olivier Saulnier STEGANUX 1er étage DIAMECANS BEL AIR 03410 St-Victor T: 04.70.02.27.62 F: 04.70.09.97.41 http://www.steganux.com ___ --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] AGI Script with parameters
YES!! It works ;-)) Best regards, Olivier S. Anton Frolov a écrit : you should use a separator between the arguments as well. try exten = 0470022762,2,agi,/ruby/ring.rb|192.168.0.10|5010 -- Olivier Saulnier STEGANUX 1er étage DIAMECANS BEL AIR 03410 St-Victor T: 04.70.02.27.62 F: 04.70.09.97.41 http://www.steganux.com ___ --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] AGI and some informations
Hello, i would like to use AGI and Ruby for communicate with a softphone. I would like send the IP adress of the softphone, directly for the extensions.conf file, as: exten = 302,1,agi,/ruby/ipphone.rb|ip_adress I know, with extensions.conf file, that i work on the softphone 302. But, how can i read the ip adress (the softphone give me when it registered, but after??? best regards, -- Olivier Saulnier STEGANUX 1er étage DIAMECANS BEL AIR 03410 St-Victor T: 04.70.02.27.62 F: 04.70.09.97.41 http://www.steganux.com ___ --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] AGI and some informations
Anton Frolov a écrit :
When registering the softphone:
OK, in which file do i do that??
SoftPhonesDB.insert(olivier, $ip);
Could you explain me what means SoftPhonesDB.insert?? It's not an AGI
commande, how can i use it??
In extensions.conf:
exten = 302,s,agi,script.rb,${EXTEN}
OK, i understand
In script.rb:
$ip = SoftPhonesDB.select(olivier);
Dial(SIP/$ip, $arg);
Hummm...Does Dial working with Ip adress?? Or should i write ine the
database the softphone number too??
Best regards,
Olivier S
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[asterisk-users] Catch an event
Hello, I would like for some reasons, catch the ring event since Asterisk, in real-time. Is this information record in a database? How can I read it, immediatly? I either think to catch the information by a little shell script as: asterisk -r |tail -1|grep ring|awk ... and redirect the internal number to an application fir process? Do you see a best way to do that?? The reason of this exercise is simply that i develop a softphone with an iax dll, but we use the french language Windev. All functions are OK, except sounds functions, as ring or dialling tone... So, i try to get some solutions... Best regards, OLS ___ --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] french promt
Hello Khaled, Follow this link: http://svn.digium.com/view/asterisk/sounds/fr/trunk/?rev=34575 Best regards, Olivier S. Khaled Chehab a écrit : Please any one knows from where I can download asterisk French sounds /var/lib/asterisk/sounds. //Regards// * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --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 -- Olivier Saulnier STEGANUX 1er étage DIAMECANS BEL AIR 03410 St-Victor T: 04.70.02.27.62 F: 04.70.09.97.41 http://www.steganux.com ___ --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] Problem of Quality
Hello, Sometimes, when i call an outside people, he said me that the communication is bad: The voice is low, far, bad poor quality. How can i know where is the problem, which tests can i make? Best regards, -- Olivier Saulnier STEGANUX 1er étage DIAMECANS BEL AIR 03410 St-Victor T: 04.70.02.27.62 F: 04.70.09.97.41 http://www.steganux.com ___ --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] Phone Ring
Hello, Do you know where i can download some rings for a PA1688 based Phone? All rings on this link are not very nice...: http://www.aredfox.com/edownloadsring.htm Best regards, -- Olivier Saulnier STEGANUX 1er étage DIAMECANS BEL AIR 03410 St-Victor T: 04.70.02.27.62 F: 04.70.09.97.41 http://www.steganux.com ___ --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] request @default
Hello, I creat two context in extensions.conf: [creat-in] exten = 400,Dial(IAX2/400,20,tr) exten = 400,2,Voicemail(u400) exten = 400,10,Hangup exten = 401,Dial(IAX2/401,20,tr) exten = 401,2,Voicemail(u401) exten = 401,10,Hangup [steganux-in] exten = 300,Dial(IAX2/300,20,tr) exten = 300,2,Voicemail(u300) exten = 300,10,Hangup exten = 301,Dial(IAX2/301,20,tr) exten = 301,2,Voicemail(u301) exten = 301,10,Hangup exten = 0470022762,1,Dial(IAX2/300,20,tr) exten = 0470022762,1,Dial(IAX2/301,20,tr) exten = 0470022762,2,Voicemail(u300) exten = 0470022762,10,Hangup But when i want to call 400 from 300, it doesn't work, and log file return: chan_iax2.c: Rejected connect attempt from 192.168.0.60, request '[EMAIL PROTECTED]' does not exist Do you have any idea? Best regards, -- Olivier Saulnier STEGANUX 1er étage DIAMECANS BEL AIR 03410 St-Victor T: 04.70.02.27.62 F: 04.70.09.97.41 http://www.steganux.com ___ --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] request @default
Hi, I've forgot thos part, but default context contains: [default] include = demo include = free-out include = bri-out include = steganux-in include = creat-in include = steganux-out include = creat-out include = commun ; contexte pour usage interne I trie to creat a new context, nammed commun which contains creat-in and steganux-in, but not better. Best regards, Olivier S Kai Fürstenberg a écrit : Hi, Olivier Saulnier wrote: Hello, I creat two context in extensions.conf: [creat-in] exten = 400,Dial(IAX2/400,20,tr) exten = 400,2,Voicemail(u400) exten = 400,10,Hangup exten = 401,Dial(IAX2/401,20,tr) exten = 401,2,Voicemail(u401) exten = 401,10,Hangup [steganux-in] exten = 300,Dial(IAX2/300,20,tr) exten = 300,2,Voicemail(u300) exten = 300,10,Hangup exten = 301,Dial(IAX2/301,20,tr) exten = 301,2,Voicemail(u301) exten = 301,10,Hangup exten = 0470022762,1,Dial(IAX2/300,20,tr) exten = 0470022762,1,Dial(IAX2/301,20,tr) exten = 0470022762,2,Voicemail(u300) exten = 0470022762,10,Hangup But when i want to call 400 from 300, it doesn't work, and log file return: chan_iax2.c: Rejected connect attempt from 192.168.0.60, request '[EMAIL PROTECTED]' does not exist Do you have any idea? Check in which context 300 is working. It seems to be in the [default] context. If you haven't included [creat-in] into the default context, then dialling 400 will not work. Kai ___ --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 -- Olivier Saulnier STEGANUX 1er étage DIAMECANS BEL AIR 03410 St-Victor T: 04.70.02.27.62 F: 04.70.09.97.41 http://www.steganux.com ___ --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] request @default
Doug Lytle a écrit : This is incorrect. It should be: [creat-in] exten = 400,1,Dial(IAX2/400,20,tr) exten = 400,2,Voicemail(u400) exten = 400,3,Hangup Ok, i forgot priority!! Best regards, -- Olivier Saulnier STEGANUX 1er étage DIAMECANS BEL AIR 03410 St-Victor T: 04.70.02.27.62 F: 04.70.09.97.41 http://www.steganux.com ___ --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] Forward call
Hie,
I trie to use a simply call forward, found on this mailing list (:-),
when i'm not near my phone:
i creat a global set:
olscell=123456789 ; my cell phone number
A macro for forwarding the call:
[macro-cell_user]
exten = s,1,Playback(Call_Transfer)
exten = s,2,Flash()
exten = s,3,SendDTMF(${ARG1})
exten = s,4,Hangup()
I put in m incoming context:
exten = 0470022762,1,Dial(IAX2/300,20,tr)
exten = 0470022762,2,Macro(cell_user,${olscell})
But, when the call is being, the phone is hangup!
What do i do in macro for forward the call??
Best regards,
--
Olivier Saulnier
STEGANUX
1er étage DIAMECANS
BEL AIR
03410 St-Victor
T: 04.70.02.27.62
F: 04.70.09.97.41
http://www.steganux.com
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Re: [Asterisk-Users] Xorcom Rapid
Hello,
I've receive no response, no idea??
Bets regards,
Olivier S;
Tzafrir Cohen a écrit :
Jut as usual with Zaptel: Zap/NNN (e.g: Zap/1 , Zap/2) for individual
channels. And gNNN and similar work just the same.
OK, in extensions.conf, i put the contexts PSTN and INTERNAL as:
[PSTN] ; for in coming calls - defin in zapata.conf
exten = s,1,Dial(IAX2/300,20)
exten = s,2,Voicemail, u300)
[INTERNAL] ; for internal AND outgoing call - actually just outgoing calls
exten = _0.,1,Dial(ZAP/g1/${EXTEN:1})
For hardware, how can i know on which interface is connected my ISDN line??
For outgoing call, i name the channel ZAP/1 in extensions.conf file, but
i dont know if it's correct.
And i always have the message timeout, but no rule 't' in context
What's mean??
There is no extension named t in that context to handle timeouts.
Your dialplan reads:
[PSTN]
exten = 1,1,Dial (IAX2/300,20)
exten = s,2,Voicemail, u300)
So no timeout action is specified. Ignore it if you don't just want to
have the call disconnected on timeout without taking any other action.
I'm not sure if the space after Dial is legal. I figure it may be the
source to your problem. Do you get an error in the CLI when reloading?
Before reloading:
set verbose 1
to see only the relevant warnings.
I have the same message!
Do you know how i can stop messages from qozap (they fill the screen
either asterisk is down!!!)
Best regards,
--
Olivier Saulnier
STEGANUX
1er étage Diamecans
Bel Air
03410 St Victor
T: 04.70.02.27.62
F: 04.70.09.97.41
http://www.steganux.com
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Re: [Asterisk-Users] Xorcom Rapid
Tzafrir Cohen a écrit :
I'm still not hapy with that as a default. It should provide you a basis
for manual editing at this stage. But I wonder what else could the
script configured there differently. Are those sane defaults for BRI on
France?
I've modified zaptel-channels.conf file , because, nothing happen when i
call from an external phone inside the company.
It's my problem, i don't know how name the QuadBRI interface, and how to
use it in extensions files
Do you hace some samples to give me, or explain me how i can detect the
name to use?
Best regards,
Olivier Saulnier
# Global data
loadzone= fr
defaultzone= fr
zaptel-channels.conf:
; Autogenerated by /usr/sbin/genzaptelconf -- do not hand edit
; Zaptel Channels Configurations (zapata.conf)
;
; This is not intended to be a complete zapata.conf. Rather, it is intended
; to be #include-d by /etc/zapata.conf that will include the global settings
;
; Span 1: ztqoz/2/1 quadBRI PCI ISDN Card 1 Span 1 [TE] (cardID 0)
group=0
context=PSTN
switchtype = euroisdn
signalling = bri_cpe
channel = 1-2
; Span 2: ztqoz/2/2 quadBRI PCI ISDN Card 1 Span 2 [TE] (cardID 0)
group=0
context=PSTN
switchtype = euroisdn
signalling = bri_cpe
channel = 4-5
; Span 3: ztqoz/2/3 quadBRI PCI ISDN Card 1 Span 3 [TE] (cardID 0)
group=0
context=PSTN
switchtype = euroisdn
signalling = bri_cpe
channel = 7-8
; Span 4: ztqoz/2/4 quadBRI PCI ISDN Card 1 Span 4 [TE] (cardID 0)
group=0
context=PSTN
switchtype = euroisdn
signalling = bri_cpe
channel = 10-11
extensions.conf:
[general]
static=yes
; we don't want asterisk to write the configuration, as it will write
; everything to a single file
writeprotect=yes
[globals]
#include extensions-defs.conf
; another #include. This one includes complete contetexts.
; What happens if a section that has existed is re-added?
;
; Currently Asterisk ignores the new section. And thus is is very simple
; to override existing extensions. However nobody guarantees that the
; configurations will be paserd the same way in the future. This is
intended
; for immediate hacks and for long-run system breakage.
#include extensions.d/*.conf
; Basically you should not edit this file to add new stuff: add/edit
; files in extensions.d/ instead. Fr instance: to add an IVR: look at
; extensions.d/ivr.conf and later on 'include = ivr' instead of
; 'include =phone'
[macro-stdexten]
;
; Standard extension macro:
; ${ARG1} - Device(s) to ring
; ${ARG2} - flags for Dial: if empty: tr. pass '-' for no flags.
; ${ARG3} - voicemail box. If empty: use the extension number.
exten = s,1,SetVar(VMBOX=${MACRO_EXTEN}); default for VMBOX, if no ARG3
exten = s,2,GotoIf($[${LEN(${ARG3})} = 0]?4)
exten = s,3,SetVar(VMBOX=${ARG3})
; Ring the interface, 20 seconds maximum
exten = s,4,SetVar(FLAGS=r)
; why 'x'? see bourne shell 101
exten = s,5,GotoIf($[ x${ARG2} = x- ]?7); '-' as the 'flags' argument
exten = s,6,SetVar(FLAGS=${ARG2})
exten = s,7,Dial(${ARG1},20,${ARG2})
; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = s,8,Goto(s-${DIALSTATUS},1)
; If unavailable, send to voicemail w/ unavail announce
exten = s-NOANSWER,1,Voicemail(u${VMBOX})
; If they press #, return to start
exten = s-NOANSWER,2,Goto(${MACRO_CONTEXT},s,1)
; If busy, send to voicemail w/ busy announce
exten = s-BUSY,1,Voicemail(b${VMBOX})
; If they press #, return to start
exten = s-BUSY,2,Goto(${MACRO_CONTEXT},s,1)
; Treat anything
exten = _s-.,1,Goto(s-NOANSWER,1)
;
; You may want to improve this one
;
[macro-stdmeetme]
exten = s,1,MeetMe(${MACRO_EXTEN},M)
[macro-dialout]
;
; a macro for setting up a trunk
; usage:
;
; Arguments:
;
; ARG1: trunk channels: a ''-separated list of channels
; ARG2: number: the number to dial.
;
; Example:
;
; exten = _9.,Macro(dialout,Zap/1Zap2,${EXTEN:1})
;
exten = s,1,ChanIsAvail(${ARG1}); use
exten = s,102,Goto(s-CHANUNAVAIL,1) ; this indicates that all lines
exten = s,2,SetVar(DIALLINE=${AVAILORIGCHAN})
exten = s,3,Goto(start,1) ;
include = trunk-macros-common
[macro-trunksip]
;
; a macro for setting up a trunk
; usage:
;
; Arguments:
;
; ARG1: trunk channel: a *single* channel name: SIP/peer, IAX2/peer
; Does this work for OH323?
; ARG2: number: the number to dial.
; ARG3 (optional): maximal number of calls allowed in this trunk.
; If not given: unlimited.
;
; Example:
;
; exten = _9.,Macro(Zap/1Zap2,${EXTEN:1})
;
exten = s,1,GotoIf($[${ARG3} = ]?6)
; The group name is the sip/iax peer
exten = s,2,Cut(GROUPNAME,ARG1,,1); leave only the first target
exten = s,3,Cut(GROUPNAME,GROUPNAME,/,2); extract peer name
exten = s,4,SetGroup(${GROUPNAME})
exten = s,5,CheckGroup(${ARG3})
exten = s,106,Goto(s-CHANUNAVAIL,1)
exten = s,6,SetVar(DIALLINE=${ARG1})
exten = s,7,Goto(start,1)
include = trunk-macros-common
[trunk-macros-common]
;
; a macro for setting up a trunk
; usage:
;
; Arguments:
;
; DIALLINE: trunk channels: The channel
Re: [Asterisk-Users] Xorcom Rapid
Tzafrir Cohen a écrit :
Jut as usual with Zaptel: Zap/NNN (e.g: Zap/1 , Zap/2) for individual
channels. And gNNN and similar work just the same.
OK, in extensions.conf, i put the contexts PSTN and INTERNAL as:
[PSTN] ; for in coming calls - defin in zapata.conf
exten = s,1,Dial(IAX2/300,20)
exten = s,2,Voicemail, u300)
[INTERNAL] ; for internal AND outgoing call - actually just outgoing calls
exten = _0.,1,Dial(ZAP/g1/${EXTEN:1})
For hardware, how can i know on which interface is connected my ISDN line??
For outgoing call, i name the channel ZAP/1 in extensions.conf file, but
i dont know if it's correct.
And i always have the message timeout, but no rule 't' in context
What's mean??
There is no extension named t in that context to handle timeouts.
Your dialplan reads:
[PSTN]
exten = 1,1,Dial (IAX2/300,20)
exten = s,2,Voicemail, u300)
So no timeout action is specified. Ignore it if you don't just want to
have the call disconnected on timeout without taking any other action.
I'm not sure if the space after Dial is legal. I figure it may be the
source to your problem. Do you get an error in the CLI when reloading?
Before reloading:
set verbose 1
to see only the relevant warnings.
I have the same message!
Do you know how i can stop messages from qozap (they fill the screen
either asterisk is down!!!)
Best regards,
--
Olivier Saulnier
STEGANUX
1er étage Diamecans
Bel Air
03410 St Victor
T: 04.70.02.27.62
F: 04.70.09.97.41
http://www.steganux.com
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[Asterisk-Users] Xorcom Rapid
,GotoIf($[${ARG3} = ]?6)
; The group name is the sip/iax peer
exten = s,2,Cut(GROUPNAME,ARG1,,1); leave only the first target
exten = s,3,Cut(GROUPNAME,GROUPNAME,/,2); extract peer name
exten = s,4,SetGroup(${GROUPNAME})
exten = s,5,CheckGroup(${ARG3})
exten = s,106,Goto(s-CHANUNAVAIL,1)
exten = s,6,SetVar(DIALLINE=${ARG1})
exten = s,7,Goto(start,1)
include = trunk-macros-common
[trunk-macros-common]
;
; a macro for setting up a trunk
; usage:
;
; Arguments:
;
; DIALLINE: trunk channels: The channel through which to dial
; ARG2: number: the number to dial.
;
; Example:
;
; exten = _9.,Macro(Zap/1Zap2,${EXTEN:1})
;
exten = start,1,Dial(${DIALLINE}/${ARG2})
exten = start,2,Goto(s-${DIALSTATUS},1)
exten = s-ANSWER,1,Goto(s-HANGUP,1)
exten = s-HANGUP,1,Hangup
exten = s-NOANSWER,1,Goto(s-HANGUP,1)
exten = s-CHANUNAVAIL,1,Playtones(congestion)
exten = s-CHANUNAVAIL,2,Wait(3)
exten = s-CHANUNAVAIL,3,Goto(s-HANGUP,1)
exten = s-BUSY,1,Playtones(busy)
exten = s-BUSY,2,Wait(3)
exten = s-BUSY,3,Goto(s-HANGUP,1)
exten = s-CONGESTION,1,Goto(s-BUSY,1)
exten = s-CANCEL,1,Goto(s-HANGUP,1)
[phones]
; conf files in the extensions-phones.d subdirectory should have no context.
; They are all to be part of the 'phones' context
#include extensions-phones.d/*.conf
include = phones-zap
[PSTN]
exten = 1,1,Dial (IAX2/300,20)
exten = s,2,Voicemail, u300)
[INTERNAL]
;exten = 300,1,Dial(IAX2/10,20,tr)
;exten = 300,2,voicemail(u10)
;exten =300,hangup
;exten = 300,2,voicemail(b10)
;exten =300,103,hangup
exten = _0.,1,Dial(ZAP/2/${EXTEN:3})
exten = _3.,1,Dial(ZAP/2/${EXTEN:3})
Best regards,
--
Olivier Saulnier
STEGANUX
1er étage Diamecans
Bel Air
03410 St Victor
T: 04.70.02.27.62
F: 04.70.09.97.41
http://www.steganux.com
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Re: [Asterisk-Users] QuadBri card
It's mark on some documentations... Where do i laucnh qozap ?? Best regards, Olivier S. Tzafrir Cohen a écrit : On Wed, Jun 07, 2006 at 05:44:16PM +0200, Olivier Saulnier wrote: Hello, I install the latest release of Asterisk, QuadBri driver. I compile al; that, copy zaptel.conf file, modify /etc/rc.d/rc.local for launch qozap... Bad place. rc.local is just about the last place in the init sequence to be run. After Asterisk is started. -- Olivier Saulnier STEGANUX 1er étage Diamecans Bel Air 03410 St Victor T: 04.70.02.27.62 F: 04.70.09.97.41 http://www.steganux.com ___ --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] QuadBri card
Hello, I install the latest release of Asterisk, QuadBri driver. I compile al; that, copy zaptel.conf file, modify /etc/rc.d/rc.local for launch qozap... zaptel.conf: --- # hfc-s pci a span definition # most of the values should be bogus because we are not really zaptel loadzone=fr defaultzone=fr span=1,1,3,ccs,ami span=2,1,3,ccs,ami span=3,1,3,ccs,ami span=4,1,3,ccs,ami bchan=1-2 dchan=3 bchan=4-5 dchan=6 bchan=7-8 dchan=9 bchan=10-11 dchan=12 zapata.conf: ; ; Zapata telephony interface ; ; Configuration file [trunkgroups] [channels] switchtype=euroisdn pridialplan=local prilocaldialplan=local language=fr context=from-pstn ;signalling=fxs_ks ; OLS signalling=bri_cpe_ptmp rxwink=300; Atlas seems to use long (250ms) winks ; ; Whether or not to do distinctive ring detection on FXO lines ; ;usedistinctiveringdetection=yes usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=800 rxgain=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no ;faxdetect=both faxdetect=incoming ;faxdetect=outgoing ;faxdetect=no ;Include genzaptelconf configs #include zapata-auto.conf group=1 ;Include AMP configs #include zapata_additional.conf Asterisk is OK, but when i plug my ISDN phone lines, the leds of the QuadBri card stays red! Nothing happen when i call the phone number by external line. I always have at asterisk console the message: qozap: not re-activating layer1 span1 I see my channels with zap show status if I do: less /proc/zaptel/3 (par exemple), I have the return message for each channel: DEACTIVATING What's happen?? Best regards, -- Olivier Saulnier STEGANUX 1er étage Diamecans Bel Air 03410 St Victor T: 04.70.02.27.62 F: 04.70.09.97.41 http://www.steganux.com ___ --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] Compiling QuaBri cards
Hello, I've installed a QuadBri card from Junghanns, and have some problems for compiling software. Compiling Zaptel and LibPRI are OK. But when i want to compile ztgsm files, i have the error: link /usr/src/linux-2.6 to your kernel sources first ! I work with kernel 2.6.8-2-686, and i have a link: linux-2.6.8-2-686 from /lib/modules/2.6.8-2-686. When i creat a link for linux-2.6 on this directory, i have some news erreors; has: structure has no member... Could you help me on this problem?? Best regards, -- Olivier Saulnier STEGANUX 1er étage Diamecans Bel Air 03410 St Victor T: 04.70.02.27.62 F: 04.70.09.97.41 http://www.steganux.com ___ --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] Compiling QuaBri cards
Tzafrir Cohen a écrit : wget http://rapid.dotsrc.org/rapid/pool/main/z/zaptel/zaptel-modules-2.6.8-2-686_1.2.5-4+2.6.8-16sarge1_i386.deb It use as dependance the zaptel deb package, but in the website, only release for i386 is available, is it good?? When i depackage it, i have the errors: (Lecture de la base de données... 25603 fichiers et répertoires déjà installés.) Préparation du remplacement de zaptel 1:1.2.5-4 (en utilisant zaptel_1.2.5-4_i386.deb) ... Dépaquetage de la mise à jour de zaptel ... Paramétrage de zaptel (1.2.5-4) ... Zaptel cards initial configuration: FATAL: Error inserting ztdummy (/lib/modules/2.6.8-2-686/extra/ztdummy.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error running install command for ztdummy zaptel. Do you have any idea?? Also i am not sure that the zaptel package furnish on the site is the last one... Best regards, Olivier S. Alternatively, install zaptel-source from there and build using m-a a-i zaptel -- Olivier Saulnier STEGANUX 1er étage Diamecans Bel Air 03410 St Victor T: 04.70.02.27.62 F: 04.70.09.97.41 http://www.steganux.com ___ --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] Developping SoftPhone
Hello, I would like to use an ocx for integrated a softphone in an existant program developped in Windev (from PC Soft). I try IaxClientOCx, but nothing happen at initialising. Then, I try some softphone make with it, it doesn't function either... Do you know any other OCX for try? Best regards, Olivier S. ___ --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] Listening a conversation
Hello, is it possible to listen a conversation in real time, without recording it? Best regards, -- Olivier Saulnier STEGANUX 35 Quai Louis Blanc 03100 Montluçon T: 04.70.02.80.55 F: 04.70.02.80.57 http://www.steganux.com ___ --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] Listening a conversation
Ok, thanks everybody :-) -- Olivier Saulnier STEGANUX 35 Quai Louis Blanc 03100 Montluçon T: 04.70.02.80.55 F: 04.70.02.80.57 http://www.steganux.com ___ --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] IAX Configuration
Hello, I have some problems with a new configuration: I always have on my asterisk console the message: chan_iax2.c:5886 update registry: restricting registration for peer '19' to 60 secondes I connect only two ip phone with iax protocol. And when i want to call 19 phone, it's hangup. No information in console view, or in file /var/log/asterisk/messages. Do you have any idea? My files a there: extensions.conf: [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no priorityjumping=no [globals] CONSOLE=Console/dsp; Console interface for demo ;CONSOLE=Zap/1 ;CONSOLE=Phone/phone0 IAXINFO=guest; IAXtel username/password TRUNK=Zap/g2 TRUNKMSD=1 [INTERNAL] exten = 19,1,Dial(SIP/19,20,tr) exten = 19,2,Voicemail(u19) exten = 19,hangup exten = 19,102, Voicemail (b19) exten = 19,103,Hangup exten = 20,1,Dial(SIP/20,20,tr) exten = 20,2,Voicemail(u20) exten = 20,hangup exten = 20,102, Voicemail (b20) exten = 20,103,Hangup iax.conf: [general] bandwidth=low disallow=lpc10 jitterbuffer=no forcejitterbuffer=no [19] type = friend username = 19 secret = 19 host=dynamic context = INTERNAL mailbox=19 [20] type = friend username = 20 secret = 20 host=dynamic context = INTERNAL mailbox=20 Best regards, -- Olivier Saulnier STEGANUX 35 Quai Louis Blanc 03100 Montluçon T: 04.70.02.80.55 F: 04.70.02.80.57 http://www.steganux.com ___ --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] IAX Configuration
Yes Sean, I've just see that :-) I modify, and my communication is now OK. But i always have the message on the console... Bets regards, OLS Sean Cook a écrit : my guess is that you are trying to dial a sip channel to reach an iax peer. Dial(SIP/19) should be Dial(IAX2/19) Olivier Saulnier wrote: Hello, I have some problems with a new configuration: I always have on my asterisk console the message: chan_iax2.c:5886 update registry: restricting registration for peer '19' to 60 secondes I connect only two ip phone with iax protocol. And when i want to call 19 phone, it's hangup. No information in console view, or in file /var/log/asterisk/messages. Do you have any idea? My files a there: extensions.conf: [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no priorityjumping=no [globals] CONSOLE=Console/dsp; Console interface for demo ;CONSOLE=Zap/1 ;CONSOLE=Phone/phone0 IAXINFO=guest; IAXtel username/password TRUNK=Zap/g2 TRUNKMSD=1 [INTERNAL] exten = 19,1,Dial(SIP/19,20,tr) exten = 19,2,Voicemail(u19) exten = 19,hangup exten = 19,102, Voicemail (b19) exten = 19,103,Hangup exten = 20,1,Dial(SIP/20,20,tr) exten = 20,2,Voicemail(u20) exten = 20,hangup exten = 20,102, Voicemail (b20) exten = 20,103,Hangup iax.conf: [general] bandwidth=low disallow=lpc10 jitterbuffer=no forcejitterbuffer=no [19] type = friend username = 19 secret = 19 host=dynamic context = INTERNAL mailbox=19 [20] type = friend username = 20 secret = 20 host=dynamic context = INTERNAL mailbox=20 Best regards, ___ --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 -- Olivier Saulnier STEGANUX 35 Quai Louis Blanc 03100 Montluçon T: 04.70.02.80.55 F: 04.70.02.80.57 http://www.steganux.com ___ --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] Compiling zaptel
Hello, When I compile zaptel application, i have this error log file: cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -lm gendigits.c -o gendigits ./gendigits cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\makefw.c -o makefw ./makefw tormenta2.rbt tor2fw tor2fw.h Loaded 69900 bytes from file ./makefw pciradio.rbt radfw radfw.h Loaded 42096 bytes from file ZAPTELVERSION=SVN-trunk-r980 build_tools/make_version_h version.h.tmp if cmp -s version.h.tmp version.h ; then echo; else \ mv version.h.tmp version.h ; \ fi rm -f version.h.tmp gcc -I/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB -I/drivers/net -Wall -I. -Wstrict-prototypes -fomit-frame-pointer -I/drivers/net/wan -I/include/net -DSTANDALONE_ZAPATA -o zaptel.o -c zaptel.c In file included from zaptel.c:43: /usr/include/linux/kernel.h:72: error: erreur de syntaxe before size_t /usr/include/linux/kernel.h:74: error: erreur de syntaxe before size_t In file included from /usr/include/linux/timex.h:186, from /usr/include/linux/sched.h:11, from /usr/include/linux/module.h:10, from zaptel.c:45: /usr/include/linux/time.h:14: error: erreur de syntaxe before time_t /usr/include/linux/time.h:16: error: erreur de syntaxe before '}' token /usr/include/linux/time.h:20: error: erreur de syntaxe before time_t In file included from /usr/include/linux/timex.h:186, from /usr/include/linux/sched.h:11, from /usr/include/linux/module.h:10, from zaptel.c:45: /usr/include/linux/time.h: Dans la fonction « timespec_to_jiffies »: /usr/include/linux/time.h:198: error: dereferencing pointer to incomplete type /usr/include/linux/time.h:199: error: dereferencing pointer to incomplete type /usr/include/linux/time.h: Dans la fonction « jiffies_to_timespec »: /usr/include/linux/time.h:219: error: dereferencing pointer to incomplete type I cut here, because the end of file is in error too :-) I use the latest release of zaptel. The problem is that the type size_t in /usr/include/linux/kernel.h is not recognised. If i replace it by int type, the compilation is OK until i have the type time_t in time.h file :-( Di you have any idea for that (how to do accept this types, or remplace them by what?? Best regards, -- Olivier Saulnier STEGANUX 35 Quai Louis Blanc 03100 Montluçon T: 04.70.02.80.55 F: 04.70.02.80.57 http://www.steganux.com ___ --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] AGI and incoming call
Hello, I thought it's exactly what i ask!! Very well!! Bets regards, Olivier S. picciuX a écrit : anyway, you could put the routing stuff in an external file included in extensions.conf: ... some dialplan stuff ... [extension-routing] #include ext-routing.conf ... some dialplan stuff ... in ext-routing.conf you have your routing stuff: exten = 1234567,1,Dial(11) exten = 7654321,1,Dial(12) and so on When you have a new customer, you only need to regenerate ext-routing.conf from the db and asterisk -rx extensions reload without having asterisk wait an AGI script to re-parse the routing stuff on each and every call... hope this helps 2006/4/26, Olivier Saulnier [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: Hello, It's not possible, because the flat file is generated since a database, and each day, there is news customers. Best regards, Olivier S. Innocent Evil a écrit : Why don't you do something like this: exten = 12345678,1,Dial(10) exten = 45874521,1,Dial(11) exten = 32544884,1,Dial(12) replace Dial(10) and so on with apppriate extension. Thanks, -- You don't have any choice, you already made it before you came here. -Original Message- From: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Sent: Wed, 26 Apr 2006 08:47:03 +0200 To: asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com Subject: [Asterisk-Users] AGI and incoming call Hello, I would like to intercept each incoming call and with an awk script, search the internal phone number ask. For example: I have a text database as this: External phone Internal Phone 12345678 10 45874521 11 32544884 12 When the client 45874521 call, Asterisk must routed the incoming call to the internal phone 11 I have an awk script able to find the good internal phone, but i don't know how to interface it with Asterisk. I thought that AGI is the best way. Is it? Best regards, -- Olivier Saulnier STEGANUX 35 Quai Louis Blanc 03100 Montluçon T: 04.70.02.80.55 F: 04.70.02.80.57 http://www.steganux.com ___ --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 ___ --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 -- Olivier Saulnier STEGANUX 35 Quai Louis Blanc 03100 Montluçon T: 04.70.02.80.55 F: 04.70.02.80.57 http://www.steganux.com http://www.steganux.com ___ --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 ___ --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 -- Olivier Saulnier STEGANUX 35 Quai Louis Blanc 03100 Montluçon T: 04.70.02.80.55 F: 04.70.02.80.57 http://www.steganux.com ___ --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] AGI and incoming call
Hello, I would like to intercept each incoming call and with an awk script, search the internal phone number ask. For example: I have a text database as this: External phone Internal Phone 12345678 10 45874521 11 32544884 12 When the client 45874521 call, Asterisk must routed the incoming call to the internal phone 11 I have an awk script able to find the good internal phone, but i don't know how to interface it with Asterisk. I thought that AGI is the best way. Is it? Best regards, -- Olivier Saulnier STEGANUX 35 Quai Louis Blanc 03100 Montluçon T: 04.70.02.80.55 F: 04.70.02.80.57 http://www.steganux.com ___ --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] AGI and incoming call
Could you help me for AGI script and interface with asterisk server, please? Best regards, OLS Steve Totaro a écrit : Thats how I do it. AGI makes a call to a MSSQL server. A stored proc returns an extension based on ANI and DNIS and the call continues. Thanks, Steve Totaro Olivier Saulnier wrote: Hello, I would like to intercept each incoming call and with an awk script, search the internal phone number ask. For example: I have a text database as this: External phone Internal Phone 12345678 10 45874521 11 32544884 12 When the client 45874521 call, Asterisk must routed the incoming call to the internal phone 11 I have an awk script able to find the good internal phone, but i don't know how to interface it with Asterisk. I thought that AGI is the best way. Is it? Best regards, ___ --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 -- Olivier Saulnier STEGANUX 35 Quai Louis Blanc 03100 Montluçon T: 04.70.02.80.55 F: 04.70.02.80.57 http://www.steganux.com ___ --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] AGI and incoming call
Hello, It's not possible, because the flat file is generated since a database, and each day, there is news customers. Best regards, Olivier S. Innocent Evil a écrit : Why don't you do something like this: exten = 12345678,1,Dial(10) exten = 45874521,1,Dial(11) exten = 32544884,1,Dial(12) replace Dial(10) and so on with apppriate extension. Thanks, -- You don't have any choice, you already made it before you came here. -Original Message- From: [EMAIL PROTECTED] Sent: Wed, 26 Apr 2006 08:47:03 +0200 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] AGI and incoming call Hello, I would like to intercept each incoming call and with an awk script, search the internal phone number ask. For example: I have a text database as this: External phone Internal Phone 12345678 10 45874521 11 32544884 12 When the client 45874521 call, Asterisk must routed the incoming call to the internal phone 11 I have an awk script able to find the good internal phone, but i don't know how to interface it with Asterisk. I thought that AGI is the best way. Is it? Best regards, -- Olivier Saulnier STEGANUX 35 Quai Louis Blanc 03100 Montluçon T: 04.70.02.80.55 F: 04.70.02.80.57 http://www.steganux.com ___ --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 -- Olivier Saulnier STEGANUX 35 Quai Louis Blanc 03100 Montluçon T: 04.70.02.80.55 F: 04.70.02.80.57 http://www.steganux.com ___ --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
