[asterisk-users] Using a feature from AMI or CLI
Hi, I've defined a feature using a macro in features.conf : special = #2,peer,Macro,special Everything is working if the user use the phone key. But i would like to call the feature (or the Macro on the peer channel) from AMI or CLI. First i thought i would be simple, but i did not find any solution. Does someone has an idea ? Thank you very much. Matthieu NICAISE Responsable technique GSM : 06 72 19 09 55 techni...@thinkrosystem.com Thinkro System http://www.thinkrosystem.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Parking problem with outgoing calls
Hi everybody,
I have a problem using parking for outgoing call.
A is an local sip phone. A is using the local extension :
[local]
exten = _XXX.,1,Wait(0)
exten = _XXX.,n,Dial(SIP/${EXTEN:0...@trunk_sip_2,0,TK)
exten = _XXX.,n,Dial(DAHDI/4/${EXTEN:0},0,TK)
exten = _XXX.,n,Playback(callbox-thinkro-trunkDefautIndispo)
exten = _XXX.,n,Hangup()
exten = h,1,Hangup()
When A is calling an outside number, trunk_sip_2 is used.
Then A is parking the call. A can hear the parking number.
But after the call is parked, the displan recall the outside number
using DAHDI/4.
Here are the logs :
-- Executing [03838xx...@local:1] Wait(SIP/*15-0849ea88, 0)
in new stack
-- Executing [03838xx...@local:2] Dial(SIP/*15-0849ea88, SIP/
0383...@trunk_sip_2,0,TK) in new stack
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
-- Called 0383824...@trunk_sip_2
-- SIP/trunk_sip_2-084a5e18 is ringing
-- SIP/trunk_sip_2-084a5e18 is making progress passing it to SIP/
*15-0849ea88
-- SIP/trunk_sip_2-084a5e18 answered SIP/*15-0849ea88
callbox*CLI
callbox*CLI
callbox*CLI
-- Started music on hold, class 'default', on SIP/
trunk_sip_2-084a5e18
== Parked SIP/trunk_sip_2-084a5e18 on 900 (lot default). Will
timeout back to extension [trunk_sip_2] , 1 in 300 seconds
-- Added extension '900' priority 1 to parkedcalls (0x84868d8)
-- SIP/*15-0849ea88 Playing 'digits/9.alaw' (language 'fr')
-- SIP/*15-0849ea88 Playing 'digits/0.alaw' (language 'fr')
-- SIP/*15-0849ea88 Playing 'digits/0.alaw' (language 'fr')
-- Executing [...@local:1] Hangup(SIP/*15-0849ea88, ) in new
stack
== Spawn extension (local, h, 1) exited non-zero on 'SIP/
*15-0849ea88'
-- Executing [03838xx...@local:3] Dial(SIP/*15-0849ea88, DAHDI/
4/03838x,0,TK) in new stack
-- Called 4/0383824377
[Dec 27 17:59:54] WARNING[25542]: app_dial.c:1620 dial_exec_full:
Invalid timeout specified: '0'. Setting timeout to infinite
callbox*CLI
callbox*CLI
callbox*CLI
-- DAHDI/4-1 answered SIP/*15-0849ea88
Thank you very much for any help !
Matthieu NICAISE
Responsable technique
GSM : 06 72 19 09 55
techni...@thinkrosystem.com
Thinkro System
http://www.thinkrosystem.com/
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Re: [asterisk-users] Calls Dropping
Hi, try using screen : http://www.rackaid.com/resources/linux-screen-tutorial-and-how-to/ I think it's the best way of doing this. Regards, Matthieu NICAISE Responsable technique GSM : 06 72 19 09 55 techni...@thinkrosystem.com Thinkro System http://www.thinkrosystem.com/ Le 2 mai 10 à 15:52, Dan Journo a écrit : Hi Bob, Thanks for that. Is there any way I can make the task run in the background and free up the console? Also so that I can disconnect my ssh session without losing the task. Thanks Dan Sent from my Windows Mobile® phone. -Original Message- From: Bob Smither smit...@c-c-i.com Sent: 02 May 2010 14:04 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Calls Dropping On Sun, 2010-05-02 at 08:34 -0400, Dan Journo wrote: snip How can i log a continuous ping test to a file and include the date and time of each ping? Try this: #!/bin/sh for (( ; ; )) do NOW=$(date +%T %m/%d/%Y) PING=$(ping -qc 1 example.com) echo $NOW: $PING pinger.log done exit 0 You can then monitor the log file using: $ tail -f pinger.log You will need to use ^C to kill the script. Hope this helps. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] B410P and DTMF
Hi, I have 3 ISDN lines using the digium B410P card. Incoming and outgoing call are working. I use the following version : * libpri-1.4.10.2 * dahdi-2.2.1.1 * asterisk-1.6.2.6 On incoming calls, DTMF is not working, i can't see any logs. Here are the main configuration files : dahdi-channels.conf ;; Span 2: B4/0/1 B4XXP (PCI) Card 0 Span 1 signalling=bri_cpe callerid=asreceived group=5 context=trunk_5_0 language=fr channel = 5-6 ;; Span 3: B4/0/2 B4XXP (PCI) Card 0 Span 2 signalling=bri_cpe callerid=asreceived group=6 context=trunk_7_0 language=fr channel = 8-9 ;; Span 4: B4/0/3 B4XXP (PCI) Card 0 Span 3 signalling=bri_cpe callerid=asreceived group=7 context=trunk_9_0 language=fr channel = 11-12 /etc/dahdi/system.conf # Span 2: B4/0/1 B4XXP (PCI) Card 0 Span 1 span=2,1,0,ccs,ami # termtype: te bchan=5-6 hardhdlc=7 echocanceller=mg2,5-6 # Span 3: B4/0/2 B4XXP (PCI) Card 0 Span 2 span=3,2,0,ccs,ami # termtype: te bchan=8-9 hardhdlc=10 echocanceller=mg2,8-9 # Span 4: B4/0/3 B4XXP (PCI) Card 0 Span 3 span=4,3,0,ccs,ami # termtype: te bchan=11-12 hardhdlc=13 echocanceller=mg2,11-12 # Span 5: B4/0/4 B4XXP (PCI) Card 0 Span 4 span=5,4,0,ccs,ami # termtype: te bchan=14-15 hardhdlc=16 echocanceller=mg2,14-15 Thank you ! Matthieu NICAISE Responsable technique GSM : 06 72 19 09 55 techni...@thinkrosystem.com Thinkro System http://www.thinkrosystem.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Parking function problem ?
Hello,
i'm using the parking feature.
When the call is parked by A (number *15) , B is correctly parked, by
A did not hangup automatically.
Here are the dialplan
[local]
exten = _XXX.,1,Wait(0)
exten = _XXX.,n,Dial(SIP/${EXTEN:0...@trunk_sip_2,0,TK)
exten = _XXX.,n,Dial(DAHDI/4/${EXTEN:0},0,TK)
exten = _XXX.,n,Playback(callbox-thinkro-trunkDefautIndispo)
exten = _XXX.,n,Hangup()
exten = h,1,Hangup()
and the logs :
-- Executing [0383824...@local:1] Wait(SIP/*15-0849ea88, 0)
in new stack
-- Executing [0383824...@local:2] Dial(SIP/*15-0849ea88, SIP/
0383...@trunk_sip_2,0,TK) in new stack
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
-- Called 0383824...@trunk_sip_2
-- SIP/trunk_sip_2-084a5e18 is ringing
-- SIP/trunk_sip_2-084a5e18 is making progress passing it to SIP/
*15-0849ea88
-- SIP/trunk_sip_2-084a5e18 answered SIP/*15-0849ea88
callbox*CLI
callbox*CLI
callbox*CLI
-- Started music on hold, class 'default', on SIP/
trunk_sip_2-084a5e18
== Parked SIP/trunk_sip_2-084a5e18 on 900 (lot default). Will
timeout back to extension [trunk_sip_2] , 1 in 300 seconds
-- Added extension '900' priority 1 to parkedcalls (0x84868d8)
-- SIP/*15-0849ea88 Playing 'digits/9.alaw' (language 'fr')
-- SIP/*15-0849ea88 Playing 'digits/0.alaw' (language 'fr')
-- SIP/*15-0849ea88 Playing 'digits/0.alaw' (language 'fr')
-- Executing [...@local:1] Hangup(SIP/*15-0849ea88, ) in new
stack
== Spawn extension (local, h, 1) exited non-zero on 'SIP/
*15-0849ea88'
-- Executing [0383824...@local:3] Dial(SIP/*15-0849ea88, DAHDI/
4/0383824377,0,TK) in new stack
-- Called 4/0383824377
[Dec 27 17:59:54] WARNING[25542]: app_dial.c:1620 dial_exec_full:
Invalid timeout specified: '0'. Setting timeout to infinite
callbox*CLI
callbox*CLI
callbox*CLI
-- DAHDI/4-1 answered SIP/*15-0849ea88
Why Executing [...@local:1] Hangup(SIP/*15-0849ea88, ) did not
hangup A call ?
I hope i'm clear enough !
Thank you
Matthieu NICAISE
Responsable technique
techni...@thinkrosystem.com
Thinkro System
http://www.thinkrosystem.com/
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Re: [asterisk-users] Echo issue
Hi, I think you need to remove the line echocanceller in system.conf You could also try to use fxotune, it'a really improving things. You also need to put echocancel=yes in chan_dahdi.conf Matthieu NICAISE Responsable technique GSM : 06 72 19 09 55 techni...@thinkrosystem.com Thinkro System http://www.thinkrosystem.com/ Le 15 déc. 09 à 23:15, hin lee a écrit : If I installed a Digium echo cancellation module on my TE121 card, do I need to remove the echocanceller line under the system.conf? How should I have it? This is my system.conf: bchan=1-23 dchan=24 echocanceller=mg2,1-23 Thank you! Hin From: hin lee hi...@yahoo.com To: noahisaacmil...@gmail.com Cc: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Fri, December 11, 2009 8:56:13 AM Subject: Re: [asterisk-users] Echo issue The echo between our extensions (using Polycom 550 handsets) disappears once I removed the Digium echo module. Are you routing internal calls from SIP - DAHDI - SIP? The digium echo module will not have any effect on pure SIP - SIP calls. Do you have acoustic echo cancellation active on the Polycom phones? Internal calls should be SIP to SIP. Yes we do have the acoustic echo cancellation active on the Polycom phones. This is my system.conf: bchan=1-23 dchan=24 echocanceller=mg2,1-23 Did you use these same settings when you were using the hardware echo module? Yes, I believe so. I asked an Asterisk expert to make sure everything is working correctly when installing the hardware module. If the setting don't look correct, what should be there when we use the hardware module? Thank you! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call on hold through DTMF
Hi everybody, I have a sip phone (Siemens) which has no sip functions at all. Is is possible to press #4 by example to put the call on hold then dial #2 to get the call back ? I'have look at features.conf but i did not find the solution. I know the call parking functionnality, but i would like a much simple system. I hope i'm clear enough. Thank you Matthieu NICAISE Responsable technique GSM : 06 72 19 09 55 techni...@thinkrosystem.com Thinkro System http://www.thinkrosystem.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VoiceMail greetings
Hello everybody, I'm using Asterisk ( 1.6.1.9 ) Voicemail. For example, if i call extension *11 which is not registered, from extension *12, i have no greetings at all, i only have the please leave a message after the beep. I tried to record the busy, unavailable and temporary greetings for extension *11 using VoiveMailMain and the file are well created on the file system. I cannot understand why those files are not played. If i use VoiceMail(*11) in the extension.conf i have exactly the same behaviour. If i user VoiceMail(*11,b) the busy message is read. Is that a normal behaviour ? I can't understand why Asterisk is not using the Dial status automaticaly. Thank you for your help ! Matthieu NICAISE Responsable technique GSM : 06 72 19 09 55 techni...@thinkrosystem.com Thinkro System http://www.thinkrosystem.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoiceMail greetings
Here is the output of the CLI with verbose and debug set to 3 : == Using SIP RTP CoS mark 5 == Using SIP VRTP CoS mark 6 -- Executing [...@local:1] Dial(SIP/*15-0849a370, SIP/*11,60) in new stack == Using SIP RTP CoS mark 5 == Using SIP VRTP CoS mark 6 [Nov 29 03:38:13] WARNING[24635]: app_dial.c:1528 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [...@local:2] VoiceMail(SIP/*15-0849a370, *11) in new stack -- SIP/*15-0849a370 Playing 'vm-intro.alaw' (language 'fr') -- SIP/*15-0849a370 Playing 'beep.alaw' (language 'fr') -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/default/*11/ tmp/40taTt format: wav49, 0x849b338 -- x=1, open writing: /var/spool/asterisk/voicemail/default/*11/ tmp/40taTt format: gsm, 0x849c7c0 -- x=2, open writing: /var/spool/asterisk/voicemail/default/*11/ tmp/40taTt format: wav, 0x849cb08 -- User hung up == Parsing '/var/spool/asterisk/voicemail/default/*11/INBOX/ msg.txt': == Found == Spawn extension (local, *11, 2) exited non-zero on 'SIP/ *15-0849a370' -- Executing [...@local:1] Hangup(SIP/*15-0849a370, ) in new stack == Spawn extension (local, h, 1) exited non-zero on 'SIP/ *15-0849a370' Th Warren Matthieu NICAISE Responsable technique GSM : 06 72 19 09 55 techni...@thinkrosystem.com Thinkro System http://www.thinkrosystem.com/ Le 29 nov. 09 à 03:19, Warren Selby a écrit : Do you have *11 registered in your voicemail.conf file? What does the cli output look like when you try to leave a voicemail? Thanks, --Warren Selby On Nov 28, 2009, at 7:22 PM, matthieu Nicaise techni...@thinkrosystem.com wrote: Hello everybody, I'm using Asterisk ( 1.6.1.9 ) Voicemail. For example, if i call extension *11 which is not registered, from extension *12, i have no greetings at all, i only have the please leave a message after the beep. I tried to record the busy, unavailable and temporary greetings for extension *11 using VoiveMailMain and the file are well created on the file system. I cannot understand why those files are not played. If i use VoiceMail(*11) in the extension.conf i have exactly the same behaviour. If i user VoiceMail(*11,b) the busy message is read. Is that a normal behaviour ? I can't understand why Asterisk is not using the Dial status automaticaly. Thank you for your help ! Matthieu NICAISE Responsable technique GSM : 06 72 19 09 55 techni...@thinkrosystem.com Thinkro System http://www.thinkrosystem.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoiceMail greetings
The content of the voicemail directory is :
ls -lh /var/spool/asterisk/voicemail/default/*11/
total 324K
drwxr-xr-x 2 root root 4.0K 2009-11-28 23:49 INBOX/
drwxr-xr-x 2 root root 4.0K 2009-11-28 23:46 Old/
drwxr-xr-x 2 root root 4.0K 2009-11-28 23:46 Urgent/
-rw-r--r-- 1 root root 3.5K 2009-11-28 23:47 busy.WAV
-rw-r--r-- 1 root root 3.5K 2009-11-28 23:47 busy.gsm
-rw-r--r-- 1 root root 34K 2009-11-28 23:47 busy.wav
-rw-r--r-- 1 root root 17K 2009-11-28 23:44 greet.WAV
-rw-r--r-- 1 root root 17K 2009-11-28 23:44 greet.gsm
-rw-r--r-- 1 root root 163K 2009-11-28 23:44 greet.wav
drwxr-xr-x 2 root root 4.0K 2009-11-28 23:49 tmp/
-rw-r--r-- 1 root root 4.1K 2009-11-28 23:47 unavail.WAV
-rw-r--r-- 1 root root 4.1K 2009-11-28 23:47 unavail.gsm
-rw-r--r-- 1 root root 40K 2009-11-28 23:47 unavail.wav
I made an error in my first mail, i'm calling voicemail in
extensions.conf this way :
exten = _*.,1,Dial(SIP/${EXTEN:0},60)
exten = _*.,n,VoiceMail(${EXTEN:0},u)
exten = _*.,n,Playback(ss-noservice)
Matthieu NICAISE
Responsable technique
GSM : 06 72 19 09 55
techni...@thinkrosystem.com
Thinkro System
http://www.thinkrosystem.com/
Le 29 nov. 09 à 04:26, Warren Selby a écrit :
On Sat, Nov 28, 2009 at 8:39 PM, matthieu Nicaise techni...@thinkrosystem.com
wrote:
Here is the output of the CLI with verbose and debug set to 3 :
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
-- Executing [...@local:1] Dial(SIP/*15-0849a370, SIP/
*11,60) in new stack
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
[Nov 29 03:38:13] WARNING[24635]: app_dial.c:1528 dial_exec_full:
Unable to create channel of type 'SIP' (cause 20 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [...@local:2] VoiceMail(SIP/*15-0849a370, *11)
in new stack
-- SIP/*15-0849a370 Playing 'vm-intro.alaw' (language 'fr')
-- SIP/*15-0849a370 Playing 'beep.alaw' (language 'fr')
-- Recording the message
-- x=0, open writing: /var/spool/asterisk/voicemail/default/*11/
tmp/40taTt format: wav49, 0x849b338
-- x=1, open writing: /var/spool/asterisk/voicemail/default/*11/
tmp/40taTt format: gsm, 0x849c7c0
-- x=2, open writing: /var/spool/asterisk/voicemail/default/*11/
tmp/40taTt format: wav, 0x849cb08
-- User hung up
== Parsing '/var/spool/asterisk/voicemail/default/*11/INBOX/
msg.txt': == Found
== Spawn extension (local, *11, 2) exited non-zero on 'SIP/
*15-0849a370'
-- Executing [...@local:1] Hangup(SIP/*15-0849a370, ) in new
stack
== Spawn extension (local, h, 1) exited non-zero on 'SIP/
*15-0849a370'
Th Warren
Matthieu NICAISE
Responsable technique
GSM : 06 72 19 09 55
techni...@thinkrosystem.com
Thinkro System
http://www.thinkrosystem.com/
What is the output of 'ls -lh /var/spool/asterisk/voicemail/default/
*11/' ?
--
Thanks,
--Warren Selby
http://www.selbytech.com
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Re: [asterisk-users] VoiceMail greetings
Thank you Jonathan and Warren, I now have the answer i needed ! Matthieu NICAISE Responsable technique GSM : 06 72 19 09 55 techni...@thinkrosystem.com Thinkro System http://www.thinkrosystem.com/ Le 29 nov. 09 à 04:41, Jonathan Thurman a écrit : On Sat, Nov 28, 2009 at 5:22 PM, matthieu Nicaise techni...@thinkrosystem.com wrote: Hello everybody, I'm using Asterisk ( 1.6.1.9 ) Voicemail. For example, if i call extension *11 which is not registered, from extension *12, i have no greetings at all, i only have the please leave a message after the beep. I tried to record the busy, unavailable and temporary greetings for extension *11 using VoiveMailMain and the file are well created on the file system. I cannot understand why those files are not played. If i use VoiceMail(*11) in the extension.conf i have exactly the same behaviour. If i user VoiceMail(*11,b) the busy message is read. Is that a normal behaviour ? I can't understand why Asterisk is not using the Dial status automaticaly. Thank you for your help ! The default option for voicemail is to play only the instructions. Take a look at http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+VoiceMail for more details on the options. You will have to parse the Dial status in the dialplan, and pass 'u' for unavailable message to be played. You can see one way to parse the dial status in the sample extensions.conf file under the stdexten subroutine. There are lots of reasons to let the admin decide which greeting to play. For example, my canned 'receptionist' context plays the busy greeting as the after-hours greeting, otherwise playing the unavailable greeting. -Jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
