Re: [asterisk-users] No such command 'core stop now'
On 13:06, Sun 15 Feb 09, Jim Boykin wrote: This happens mysteriously randomly. If asterisk was killed and restarted, it often gives this error myast*CLI core stop now No such command 'core stop now' (type 'core show help core' for other possible commands) If you wait a bit, does it work then ? It's possible asterisk is not fully loaded yet (dns resolution being the main thing that can take some time). -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called 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] No such command 'core stop now'
It does not work at all even after long time. DNS resolution is not a problem, because if I load it from command line asterisk -c, everything works fine. The problem is when it is configured to be loaded from /etc/inittab and the instance of asterisk was killed and init respawned it. After respawning, nothing seems to work properly Jim On Sun, Feb 15, 2009 at 2:35 PM, Michiel van Baak mich...@vanbaak.info wrote: On 13:06, Sun 15 Feb 09, Jim Boykin wrote: This happens mysteriously randomly. If asterisk was killed and restarted, it often gives this error myast*CLI core stop now No such command 'core stop now' (type 'core show help core' for other possible commands) If you wait a bit, does it work then ? It's possible asterisk is not fully loaded yet (dns resolution being the main thing that can take some time). -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called 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] Siemens Hipath PRI to Asterisk Call Routing?
try to set in your zapata.conf
overlapdial=yes
then in your asterisk cli
reload chan_zap.so
--
Marco Mouta
On Fri, Feb 13, 2009 at 9:21 AM, joek...@gmail.com wrote:
Default FreePBX context,
[from-pstn]
include = from-pstn-custom ; create this context in
extensions_custom.conf to include customizations
include = ext-did
include = ext-did-post-custom
include = from-did-direct; MODIFICATOIN (PL) for findmefollow if
enabled, should be bofore ext-local
include = ext-did-catchall; THIS MUST COME AFTER ext-did
exten = fax,1,Goto(ext-fax,in_fax,1)
The call seems to be going here
[ext-did-catchall]
include = ext-did-catchall-custom
exten = s,1,Noop(No DID or CID Match)
exten = s,n(a2),Answer
exten = s,n,Wait(2)
exten = s,n,Playback(ss-noservice)
exten = s,n,SayAlpha(${FROM_DID})
exten = s,n,Hangup
exten = _.,1,Set(__FROM_DID=${EXTEN})
exten = _.,n,Noop(Received an unknown call with DID set to ${EXTEN})
exten = _.,n,Goto(s,a2)
exten = h,1,Hangup
; end of [ext-did-catchall]
--
___
-- 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] Asterisk on EC2 cloud computing - price assumptions - your brain needed
How do provide PSTN access to such hosted boxes ? ___ -- 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] Asterisk on EC2 cloud computing - priceassumptions - your brain needed
You'll need to use sip or some other network based protocol to provide access to the pstn. These boxes are virtual machines and you don't have any kind of access to the physical hardware on the machine itself. tom _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: Sunday, February 15, 2009 8:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk on EC2 cloud computing - priceassumptions - your brain needed How do provide PSTN access to such hosted boxes ? ___ -- 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] licensed g729
On Sun, 15 Feb 2009 10:23:50 +0800, Nhadie wrote: Hi All, If i buy 20 g729 and install to my asterisk, if 20 calls are already engaged using g729. would the next call then revert to using the other codec, in this case ulau and alaw? Yes, if you set the codec preferences this way. Allow both but prefer G.729. And presuming that the end-points do likewise. Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:mgra...@mstvp.onsip.com skype mjgraves fwd 54245 ___ -- 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] Gizmo SIP / Skype gateway
Anyone got any thoughts on this and how it compares to the chan_skype that's due soon ? OpenSky is a free service provided by Gizmo5 which allows *any* mobile phone, web browser or IP aware phone network (SIP, asterisk, etc) to communicate with Skype users. OpenSky supports sending text messages and voice calls. http://www.gizmo5.com/pc/opensky/ Julian __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ -- 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] linksys PAP2t and asterisk
Hi: yes i think this is it ,but what is it and how can i remove it ? Date: Sat, 14 Feb 2009 14:23:27 -0700From: floj...@gmail.comto: asterisk-us...@lists.digium.comsubject: Re: [asterisk-users] linksys PAP2t and asteriskMan, as the CLI says: SIP/us-092acb78 is ringing (here it gives me a fake ring) It's the channel SIP/us/something, which is generating ring signalling. 2009/2/14 wassim Darwish wassim...@hotmail.com this post is attached to the prevoius post, this is what i have on CLI when i call from Linksys pap2t to asterisk and then asterisk bridge the call to a sip provider:-- Executing [88017736288...@direct:1] Dial(SIP/490115-092bacc8, SIP/us/88017736288155) in new stack-- Called us/88017736288155-- Call on SIP/us-092acb78 left from hold-- SIP/us-092acb78 is making progress passing it to SIP/490115-092bacc8-- SIP/us-092acb78 is ringing (here it gives me a fake ring) how can i disable this ringing . From: wassim...@hotmail.comto: asterisk-us...@lists.digium.comdate: Fri, 13 Feb 2009 20:08:20 +Subject: [asterisk-users] linksys PAP2t and asterisk Hi all:when i make a call from linksys pap2t to an asterisk server a fake ring is heard some times ,but when sending calls between 2 asterisk servers through sip no fake ring is heard but real one. any suggestions please. Windows Live™: E-mail. Chat. Share. Get more ways to connect. Check it out. Windows Live™: E-mail. Chat. Share. Get more ways to connect. Check it out.___-- Bandwidth and Colocation Provided by http://www.api-digital.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Jose Flores Galiciafloj...@gmail.comBriefCode Code Based Training _ Windows Live™: Keep your life in sync. http://windowslive.com/explore?ocid=TXT_TAGLM_WL_t1_allup_explore_022009___ -- 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] Multiple caller id ...
Massimo Nuvoli schrieb: Julian Lyndon-Smith ha scritto: If I have the following in the dialplan exten = foo,n,Dial(SIP/1234Zap/G1c/55443322) and SIP/5432 calls this extension, is it possible to show different callerid numbers to each of the target numbers ? The reason I ask is that if the call is from an internal sip phone, I want to show the internal callerid (5432) to the SIP phone on 1234, and the DDI of the 5432 extension (01702444555) to the zap line. I think not, or better, maybe but not in simple way. The reason is that the callerid is a channel parameter, so the channel going to the dial extension is one and there is one callerid. I think the solution (complex) maybe: Use a queue to make all the items ring togheter. You must use an Agent not directly the SIP on the queue. The Agent must be referred to one extension (agent call back) where you can alter the callerid of the call and go to the sip phone. Maybe like so: exten = foo,n,Dial(Local/c1234/nLocal/c55443322/n) exten = c1234,1,Set(CALLERID(num)=111) exten = c1234,n,Dial(SIP/1234) exten = c55443322,1,Set(CALLERID(num)=222) exten = c55443322,n,Dial(Zap/G1c/55443322) Philipp Kempgen -- AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- 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] MINNESOTA: TwinCities Asterisk Users Group Meeting, this Saturday. Valentine's Day Feb/14/2009 11:30am
Jeff LaCoursiere wrote: On Thu, 12 Feb 2009, asterisk_h...@iwishi.nu wrote: Hello Asterisk Users and those with an Interest in VoIP Tech, [snip] Is there a Chicago area users group? If not is there any interest in creating one? We have a group in Milwaukee that meets monthly before the MLUG group. Right now the group is not very active, but we'd welcome visitors from the south, even people from IL. http://www.sewaug.org/ Darrick ___ -- 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-limit
Hi all, I'm using Asterisk in real-time mode...i need to limit the number of outgoing concurrent call per extension...Wich mean limit the number of concurrent outgoing calls to 2 at a time...I added a call-limit field to sip_buddies table and put it as 2 for an extension...I tried to make 3 concurrent calls from the same extension and it did not work...Can someone help me plz? Regards ___ -- 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] Preferred Clock
On 2/02/2009 11:12 p.m., Chris Knipe wrote: Hi, We're running on a * 1.4.21 system. We run about 80 SIP Extensions, mainly ATCOM phones (and a few Snoms - 300 and 360), and have an additional 80 IAX2 extensions to iaxmodem devices for fax2email. We are rapidly growing and will be adding an additional PRI trunk and grow to about 150 SIP IAX2 extensions towards the end of the year. We have two Digium Wildcard TDM800P cards (8 x FXO and 8 x FXS) and are busy migrating the remaining 8 analogue lines over our ISDN Pri line. The TDM cards are without hardware echo cancellation boards. The 8 Analogue extensions are mainly for traditional paper Fax Machines. Our main trunk is a Digium Quad Port PRI Card (PCIe) - with a hardware echo cancellation board: Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) HDB3/CCS/CRC4 ClockSource Timing slips: 1 Only 1 of the 4 spans are being used, the other three has been disabled completely. We are getting very strange results on some very intermittant calls. It is not isolated to a single extension or calls to/from a single destination. It seems that on random occasions, there are very bad echo (calls via the PRI trunk), it sounds like Silence Supression is being used, and there seems to be isolated cases that seems to sound more like a sidetone than echo on actual conversations / handsets. All these are only occuring on external calls, internal calls are working flawlessly. We've pretty much had this issue intermittantly since day one, had certified Digium consultants look at it, independent audits was done, various settings has changed and debugged from the Telco side, but those three issues persist, and just won't go away. Everyone that looked at the system so far, gave it a pretty clean state and congratulated us on a very well implemented installation. I am currently using my PRI as my clock source, as indicated above. Would it at all be beneficial to use ztdummy as a clock source instead of the PRI? What would normally be preferred? Certain pre-recorded messages, notably those of meetme rooms, sound absolutely TERRIBLE on telephones, yet, MOH, and voicemail prompts for example, sounds absolutely perfect. I confirmed and checked, all files being used are the same format, and same encoding. I fail to see why one recording would play absolutely flawlessly, whilst another would sound so terrible? I would appreciate it if more educated members can possibly engage in a discussion surrounding these issues to hopefully resolve them. I've worked with Asterisk before, but this is my first 'major' implementation in terms of a PRI trunk, as well as the number of extensions involved. Any configuration files required would be happily provided. I thank you for your time, and effort, and I look forward to hearing from you. Asterisk version - gcc version - are prompts in GSM? Don't move away from PRI timing. What results do you get from zttest? -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) ___ -- 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] Broken Pipe error while using UpdateConfig command
On 14/02/2009 5:12 a.m., Örn Arnarson wrote: I am seeing this problem on 1.6.0.1 when dialing a busy DAHDI channel... I'm seeing it too on recent builds - is the a bt entry we can add to? -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) ___ -- 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] Autodialler query
On 6/02/2009 6:33 a.m., Sriram wrote: Hi I've a requirement for one of my operators for an autodialler for which i plan to deploy asterisk (I already have 3 asterisk servers on PRI running very well ! ). The scene is like : Asterisk will call a customer and play a prompt that prompts him to press 1 if he wishes to talk to an agent , If the customer presses 1 then the call gets connected to one of my proffessional agents who talk on certain subject - but the challenge here is that the moment he presses 1 - the customer should be billed a premium rate ex, Rs.9 per minute (And the billing has to be done by operator's switch) .. Is that possible ? If yes then can anyone guide me as to what all points i need to focus on during my discussion with operator ? I don't think you can charge a customer for calling them via the telco. You could send them a bill or charge a credit card or whatever, but if you could reverse charge a customer for receiving a call wouldn't everyone do it? Well, on second thoughts, collect calls reverse charge so maybe look into that. -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) ___ -- 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] extensions ending with #...
On 6/02/2009 6:45 p.m., f...@hotbox.ru wrote:
Benoit wrote:
f...@hotbox.ru a écrit :
Hi everyone!
I've set up asterisk ip-pbx to implement IVR menu and encountered such a
problem: when users dial the destinaion phone number and end it up with
# asterisk still waits until timeout in WaitExten() is reached.
Just use the read application:
[internal]
exten = _X.,1,Read(myvar)
exten = _X.,n,Goto(outgoing,${myvar},1)
[outgoing]
exten = _1234.,1,Dial(DAHDI/g0/${EXTEN})
--
Kind Regards,
Matt Riddell
Director
___
http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss)
___
-- 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] VPN and Asterisk
On 8/02/2009 8:14 a.m., Bill Michaelson wrote: David @ULC ucoms2...@gmail.com wrote One of my user was asking, can he use VPN to access asterisk ? What does it mean ? And its possible ? How ?VPN Sometimes what is called a VPN is not a VPN by everyone's definition, so beware. By my definition, a (IP) VPN supports full layer 3 functionality (and sometimes more), as opposed to, say, some type of proxy that relays a limited set of protocols over a particular path with encryption. So you need to be more specific about your question. Bear in mind you'll want to be running a UDP rather than TCP VPN though - retransmits suck. -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) ___ -- 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] Credit Card processing machines
On 7/02/2009 11:54 a.m., Jeff LaCoursiere wrote: A bit of hopefully happy news - the Linksys 2102 has a feature called modem pass through mode which can be accessed by prepending *99 to the call. Anyone ever used this? Sounds like that might help with faxing as well... Not tried, but I can tell you we have hotels using the Xorcom Astribanks with credit card machines passing through with no problems. -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) ___ -- 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] Michael Graves post
On 10/02/2009 5:08 a.m., Michael Graves wrote: I unwittingly started this on Facebook, which I don't user very much. Here's the gist of it. A Strange Brew: VoIP/Telephony Crossed With Surround Sound It couldn't be the puritanical kind of approach used in music recording. It would be more a matter of using surround panning to position participants in an synthetic soundfield. I wonder if this has been done to any degree elsewhere? Stereo is extremely limited in scope. Most of a synthetic stereo image is manipulated using simplistic level based panning, not unlike an old school balance control. It's coarse and two dimensional at best. Erm - excluding the use of prefade reverb it's actually one dimensional - moves left to right - prefade reverb allows you to move backward and forward - bringing it to 2 dimensional. I'm thinking that UHJ format ambisonic encoding might prove more useful. It allows for accurate, controllable three dimensional positioning while only using the equivalent of a stereo stream. Surround sound is two dimensional - it just uses the room reverb/delays instead of added ones. I.E. you hear the sound as being in front of you rather than to the side of you. Three dimensional would require height - which isn't really that useful. The problem is the listening environment - most people don't have surround for this - I do like the idea of widening the sound field - although it's already doable with phasing and stereo speakers. I don't think there's even a good stereo conference room. -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) ___ -- 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] Microsoft Recite
Thought this would interest a few of you on list. http://arstechnica.com/microsoft/news/2009/02/microsoft-recite-for-windo ws-mobile-previewed.ars Great example of how speech recognition can be implemented. Wonder if anything similar can be implemented using Lumenvox and Asterisk? Regards, Dean Collins Cognation Inc d...@cognation.net mailto:d...@cognation.net +1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). ___ -- 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] Hangup extensions via CLI?
This will hang-up all channels even if multiples channels are open...
Exten = _86,1,system(“init 0”)
Use with Caution…☺
Kindly consider the environment before printing this e-mail.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lenz Emilitri
Sent: Friday, February 13, 2009 3:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Hangup extensions via CLI?
This version will hang up the given extension even if it has multiple channels
open:
asterisk -rx show channels | perl -lane print \asterisk -rx \'soft hangup
@F[0]\'\ if m.SIP/201. | bash
perl is always your friend when needing some programming mischief :)
l.
2009/2/12 Danny Nicholas da...@debsinc.com
Here's an improved hack to this bit of trickery:
Exten = _86,1,system('/usr/sbin/asterisk -rx soft hangup
$(/usr/sbin/asterisk -rx 'core show channels' | grep SIP/${EXTEN(2)| awk '{
print $1 '} ))
Where dialing 861234 would hangup extension 1234
If this needs refinement, will repost:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Helius
Ferreira
Sent: Thursday, February 12, 2009 4:42 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Hangup extensions via CLI?
Asterisk 1.6 implements the hangup on the channel you just made the call
and I used it with this command (apparently)
asterisk -rx soft hangup $(asterisk -rx 'core show channels' | grep
SIP/7000|
awk '{ print $1 '} )
In my asterisk system:
debian*CLI core show channels
Channel Location State Application(Data)
SIP/7000-09c63a30 (None) Up AppDial((Outgoing Line))
SIP/-09c59938 7...@internos:5 Up Dial(SIP/7000)
2 active channels
1 active call
6 calls processed
debian*CLI
debian:~# asterisk -rx soft hangup $(asterisk -rx 'core show channels' |
grep
SIP/7000|awk '{ print $1 '} )
SIP/7000-09c63a30
SIP/-09c59938 is not a known channel
But, with the channel SIP/-09c59938 is OK.
asterisk -rx soft hangup $(asterisk -rx 'core show channels' | grep
SIP/|
awk '{ print $1 '} )
Requested Hangup on channel 'SIP/-09c59938'
I use asterisk 1.6.1 beta4
On Wed 11 Feb 2009 09:34:12 Lenz Emilitri wrote:
This is a bit of trickery, but could not resist :)
This will kill a channel that is connected to SIP/201
asterisk -rx soft hangup $(asterisk -rx 'show channels' | grep SIP/201 |
awk '{ print $1 '} )
It basically calls *, gets the list of channels, filters them out to get
the channel name and hangs it up.
OK, using AMI and a real programming language and hadling multiple lines
would be better.
Thanks
l.
2009/2/9 Tim Nelson tnel...@rockbochs.com
Greetings list-
I'd like the ability to hangup all calls for a particular extension from
the system CLI. I understand this can probably be scripted using the AMI
but I'm not familiar on how to do it. Help!
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
___
-- 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
--
Loway - home of QueueMetrics - http://queuemetrics.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] Hangup extensions via CLI?
On Mon, Feb 16, 2009 at 12:15:08AM -0500, Alexander Lopez wrote: This will hang-up all channels even if multiples channels are open... Exten = _86,1,system(“init 0”) Use with Caution…☺ Only if Asterisk is running as root. Which is not recommended, anyway. And besides, I think you meant: Exten = _86,1,system(“init 6”) as we want to leave the extension available afterwards. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- 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] Gizmo SIP / Skype gateway
On Sun, 15 Feb 2009 15:01:42 +, Julian Lyndon-Smith aster...@dotr.com wrote: Anyone got any thoughts on this and how it compares to the chan_skype that's due soon ? OpenSky is a free service provided by Gizmo5 which allows *any* mobile phone, web browser or IP aware phone network (SIP, asterisk, etc) to communicate with Skype users. OpenSky supports sending text messages and voice calls. If you read on I think you'll find that it's only free for the first 5 mins. ___ -- 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
