[asterisk-users] How to hangup all channels
Hi guys, I'm using Asterisk 1.4 and i need a way to hangup all channels. I want to use the teleyapper system for broadcasting call for security reason but i need that all channels are free when a security call is ready to start! I already search in the old post without success. Can anyone help me? Thanks and sorry for my newbie english-- _ -- 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] change date
Hi, why have many files on http://downloads.asterisk.org/pub/telephony/asterisk/releases/ the change date 18 aug 2009? See: asterisk-1.2.24-patch.gz07-Aug-2007 17:10 3.2K asterisk-1.2.24-patch.gz.asc07-Aug-2007 17:10 1.1K asterisk-1.2.24-patch.gz.sha1 07-Aug-2007 17:1067 asterisk-1.2.24.tar.gz 18-Aug-2009 16:3328M asterisk-1.2.24.tar.gz.asc 18-Aug-2009 16:33 1.0K asterisk-1.2.24.tar.gz.sha1 18-Aug-2009 16:3365 asterisk-1.2.25-patch.gz29-Nov-2007 15:59 1.5K asterisk-1.2.25-patch.gz.asc29-Nov-2007 15:59 567 I try to repair the openembedded recipes an the recipe have also an different checksum. NOTE: fetch http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1.2.24.tar.gz NOTE: The checksums for '/home/klaus/development/oe/downloads/asterisk-1.2.24.tar.gz' did not match. Expected MD5: '63dc8b7be4cd10375c5fbda893c780bc' and Got: 'db7bcaaa494804af361157a37c224dfa' Expected SHA256: '9debaf410636fa477e1e1f09fe0b16a1c2814afaf7195f34f29e4ce5b8debbbd' and Got: 'eed3493b1409d7100e0f983af0486bd7f8965e9e47b7a6d5ab8539b2dd3609aa' NOTE: Your checksums: SRC_URI[md5sum] = db7bcaaa494804af361157a37c224dfa SRC_URI[sha256sum] = eed3493b1409d7100e0f983af0486bd7f8965e9e47b7a6d5ab8539b2dd3609aa Greetings Klaus -- _ -- 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
Re: [asterisk-users] How to hangup all channels
On Sat, 27 Nov 2010, Giuseppe D'alessio wrote:
Hi guys, I'm using Asterisk 1.4 and i need a way to hangup all channels.
1) sudo /etc/init.d/asterisk restart
2) Write a script to do asterisk -r -x 'core show channels', parse the
output and do asterisk -r -x 'channel request hangup ${CHANNEL_NAME} for
each channel.
3) Write a script to do #2 using AMI.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] How to hangup all channels
Can also do asterisk -r -x 'restart now'
asterisk*CLI help restart
restart gracefully Restart Asterisk gracefully
restart now Restart Asterisk immediately
restart when convenient Restart Asterisk at empty call volume
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Nov 27, 2010, at 8:45 AM, Steve Edwards wrote:
On Sat, 27 Nov 2010, Giuseppe D'alessio wrote:
Hi guys, I'm using Asterisk 1.4 and i need a way to hangup all channels.
1) sudo /etc/init.d/asterisk restart
2) Write a script to do asterisk -r -x 'core show channels', parse the
output and do asterisk -r -x 'channel request hangup ${CHANNEL_NAME} for
each channel.
3) Write a script to do #2 using AMI.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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http://www.asterisk.org/hello
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_
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[asterisk-users] DAHDI 2.4.0 produces HDLC errors with echo canceler
Hi, After upgrading to DAHDI 2.4.0 from Zaptel, I noticed a lot of HDLC errors on my console: [Nov 27 01:15:09] NOTICE[2743] chan_dahdi.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 [Nov 27 01:15:09] NOTICE[2743] chan_dahdi.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 These errors prevented calls from being made and received on my PRI spans. This seems similar to bug 15498: https://issues.asterisk.org/view.php?id=15498 Which says this was fixed in 2.2...so maybe it got back into 2.4? I can get rid of the errors by disabling the mg2 echo canceler in /etc/dadhi/system.conf. The PRI card I'm using is a Digium TE122. I'd prefer not having to run with the echo canceler off of course... [1] active=yes alarms=OK description=Wildcard TE122 Card 0 name=WCT1/0 manufacturer=Digium devicetype=Wildcard TE122 location=PCI Bus 03 Slot 03 basechan=1 totchans=24 irq=35 type=digital-T1 syncsrc=1 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF Thanks. -- James -- _ -- 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
Re: [asterisk-users] sip echo server
On Sat, 27 Nov 2010, Ali Khalfan wrote: I'm trying to find a way to setup a SIP server that will mainly echo back a request from one agent only, my question is do i need to setup any of the other conf files besides extensions.conf and sip.conf? 0) Posting the same request an hour later doesn't speed up responses. I'm still not sure what you are asking for. Are you wanting to pick up a SIP [hard|soft] phone, dial an extension and hear yourself talk? If so, the relevant configuration files will be: 1) modules.conf - if app_echo.so is not loaded. If 'core show application echo' works, this is not an issue. 2) sip.conf - to authenticate the [hard|soft] phone and to specify where in extensions.conf calls from that user will start. 3) extensions.conf - to define the context referenced in sip.conf and to execute the echo() application. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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
Re: [asterisk-users] sip echo server
Original Message Subject: Re: [asterisk-users] sip echo server From: Steve Edwards asterisk@sedwards.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Sat 27 Nov 2010 12:43:57 PM EST On Sat, 27 Nov 2010, Ali Khalfan wrote: I'm trying to find a way to setup a SIP server that will mainly echo back a request from one agent only, my question is do i need to setup any of the other conf files besides extensions.conf and sip.conf? 0) Posting the same request an hour later doesn't speed up responses. I apologize, I have assumed that my e-mails (including the ones) I sent yesterday were not being received) since I haven't seen my own e-mails in the list, I see there has been a reply there too, thank you for that (and Robert Thomas) I'm still not sure what you are asking for. Are you wanting to pick up a SIP [hard|soft] phone, dial an extension and hear yourself talk? If so, the relevant configuration files will be: yes, this is exactly what i want, although i only need a softphone 1) modules.conf - if app_echo.so is not loaded. If 'core show application echo' works, this is not an issue. 2) sip.conf - to authenticate the [hard|soft] phone and to specify where in extensions.conf calls from that user will start. 3) extensions.conf - to define the context referenced in sip.conf and to execute the echo() application. -- _ -- 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
Re: [asterisk-users] sip echo server
From: Steve Edwards asterisk@sedwards.com I'm still not sure what you are asking for. Are you wanting to pick up a SIP [hard|soft] phone, dial an extension and hear yourself talk? If so, the relevant configuration files will be: On Sat, 27 Nov 2010, Ali Khalfan wrote: yes, this is exactly what i want, although i only need a softphone From: Steve Edwards asterisk@sedwards.com 1) modules.conf - if app_echo.so is not loaded. If 'core show application echo' works, this is not an issue. 2) sip.conf - to authenticate the [hard|soft] phone and to specify where in extensions.conf calls from that user will start. 3) extensions.conf - to define the context referenced in sip.conf and to execute the echo() application. So, after you configure sip.conf and extensions.conf, you should be good to go. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Preserve CallerID on transfers
Hi, its possible to mantain the original CallerId when making transfers? (atx or blind) Example: A calls to B, A transfer to C, C see the CallerID of B, and not A... Its possible? Thanks1 -- _ -- 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] Strange Logfile Entries.
Hi List, Anybody any ideas on these? [Nov 26 15:14:10] WARNING[3265] chan_sip.c: Remote host can't match request NOTIFY to call '1c4890a52552c39b0b81702353087...@192.168.33.12'. Giving up. [Nov 26 15:16:44] WARNING[3265] chan_sip.c: Remote host can't match request NOTIFY to call '7920154238a73e5643a899635b0cb...@192.168.33.12'. Giving up. [Nov 26 15:23:37] NOTICE[3161] channel.c: No/unknown event '0' on timer for 'Local/2...@from-inside-aa95,2'? Customer had a problem picking up a parked call yesterday at 15:21, probably more like 15:23 as 201 was the extension.. Any ideas? Thanks in advance, Brian -- _ -- 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
Re: [asterisk-users] Preserve CallerID on transfers
On Sat, Nov 27, 2010 at 11:40 AM, Fabiano Carlos Heringer b...@grupoheringer.com.br wrote: Hi, it´s possible to mantain the original CallerId when making transfers? (atx or blind) Example: A calls to B, A transfer to C, C see the CallerID of B, and not A... It´s possible? Asterisk 1.8 added Connected Party Identification Support. Try 1.8 in a test environment and see if it meets your needs. For more info, see: http://lists.digium.com/pipermail/asterisk-announce/2010-October/000277.html -Jonathan -- _ -- 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] AstLinux 0.7.4 Release now available
The AstLinux Team is happy to announce the release of AstLinux 0.7.4. This is a dual release which allows you to chose between Asterisk 1.4.36 or 1.8.0. There are several security updates and other improvements. All current AstLinux users should upgrade as soon as feasible. One of the more significant additions includes preliminary IPv6 support. The two releases can be viewed here. http://www.astlinux.org/release/074-asterisk-1436 http://www.astlinux.org/release/074-asterisk-180 A full changelog is available on those pages. Current users can upgrade either from the web interface or via the command line. upgrade-run-image check http://mirror.astlinux.org/firmware (http://mirror.astlinux.org/ast18-firmware for Asterisk 1.8 firmware) The version should be reported as 0.7.4 upgrade-run-image upgrade http://mirror.astlinux.org/firmware The AstLinux Team -- _ -- 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
