similar to: Issue when reloading

Hello, Cedric Hans has released an UNISTIM channel driver for asterisk (stable). You can download it at : http://mlkj.net/asterisk/chan_unistim-0.9.2.tar.bz2 Copy of README : This is a channel driver for Unistim protocol. You can use at least Nortel i2004 phones with it. Only few features are supported : Send/Receive CallerID, Redial, SoftKeys, SendText(), Music On Hold, Message Waiting

I have in "unistim.conf [violet] ... context=internal but it is not recognized. When I try to make a call it looks for context "default" Is it a bug or a limitation of unistim. -- #Joseph GPG KeyID: ED0E1FB7

Hi [Feb 17 07:59:45] WARNING[21539]: channel.c:3477 ast_request: No channel type registered for 'USTM' [Feb 17 07:59:45] WARNING[21539]: app_dial.c:1502 dial_exec_full: Unable to create channel of type 'USTM' (cause 66 - Channel not implemented) == Everyone is busy/congested at this time (1:0/0/1) I get this after I restart my asterisk 1.6, it all worked yesterday. I have the

Hi All, During last three month I have worked on improving functionality of Nortel phones working with asterisk to replace existing Nortel station by asterisk. Many improvments done, listed below. I have only i2002 phone and unable to test if new version of channel correctly works with i2204 phone. If anyone can test it and report issues, it would be great. Please visit mantis to find out patch

hi list can U help me caller id in USTM if now working -- Starting switch on '4211 at 4211-1' to 4203 -- Executing [4203 at office:1] DumpChan("USTM/4211 at 4211-0x7f7ba4228fd0", "") in new stack Dumping Info For Channel: USTM/4211 at 4211-0x7f7ba4228fd0: ================================================================================ Info: Name=

Hello, I am trying to configure new opus codec in asterisk 14, but unable to find any examples of codecs.conf settings for this codec. All I am trying to do - setup peer with using opus in narrow band mode (8kHz sampling rate). Does anybody know how to configure chan_opus? -- Regards, Igor Goncharovsky Unistim Dev: http://unistim.igorg.ru -------------- next part -------------- An HTML

Hi, Does anybody have an updated Chan Unistim that compiles on Asterisk 1.2beta? Below is the output when compiling on Red Hat 9.0 Thanks, [root@maui2 chan_unistim-0.9.2]# make gcc -pipe -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/usr/include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -Wno-missing-prototypes -Wno-missing-declarations -DCRYPTO -c -o chan_unistim.o

HI, I tried to configure Asterisk 1.8 on one of my test-hosts. I've installed from centos-asterisk.repo (http://packages.asterisk.org/centos/$releasever/tested/$basearch/): Nov 26 15:34:56 Installed: asterisk-sounds-core-en-gsm-1.4.20-1_centos5.noarch Nov 26 15:34:59 Installed: asterisk18-core-1.8.0-1_centos5.i386 Nov 26 15:35:02 Installed: asterisk18-voicemail-1.8.0-1_centos5.i386 Nov 26

The Asterisk Development Team has announced the release of Asterisk 11.14.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.14.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

The Asterisk Development Team has announced the release of Asterisk 11.14.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.14.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

Hi How do i set up so that everyone can dial, for example *21* to forward all calls to a cellphone or another extension and how do I enable so that cals can be transferd between extentions. I use asterisk 1.6 and have my phones in unistim.conf and my extensions in extensions.conf. Regards /ralf ________________________________________________ Ralf Tr?skman, IT AdLibris AB, Odengatan 106, 113

The Asterisk Development Team has announced the release of Asterisk 12.7.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 12.7.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

The Asterisk Development Team has announced the release of Asterisk 12.7.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 12.7.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

Hello, I have an Asterisk 1.8.11 installation. When I built up this Asterisk, I didn't install DAHDI channel, if I issue command connect*CLI> core show channeltypes I would have response like: connect*CLI> core show channeltypes Type Description Devicestate Indications Transfer ---------- -----------

I'm having a strange problem with a sip client and 2 asterisk servers connected together with a sip trunk. Here's a rough layout sip_client ------ Asterisk A -----[sip trunk] ------ Asterisk B when the sip client tries to dial an extension on Asterisk B, Asterisk A sends the invite to B using "sip_client@[ip address of asterisk A]" rather than the username A uses to talk to B.

Hi, Trying to configure my tdm410p card, my dahdi in asterisk cli was missing. ! ael agent agi cdr channel cli config console core database devstate dialplan dnsmgr dundi features file group hangup help http iax2 indication keys

Hello everyone! I installed Asterisk following the instructions of the book "Asterisk: The Future of Telephony". (very nice book) However, I failed. I installed zaptel, libpri and asterisk (in this order). The Installation of Zaptel is successful and my TDM400P is correctly detected: # zttool Alarms Span OK Wildcard S400P

#!/bin/bash # checksetexternip.sh # Author: John Cahill email at johncahill.net # Licence: GPL v3 # Description: script that queries checkip.dyndns.com to find the server's external IP address. Updates asterisk's externip value and does a sip reload if necessary. # Last modified 06/02/2012 is_ip(){ input=$1 octet1=$(echo $input | cut -d "." -f1) octet2=$(echo $input

The Asterisk Development Team has announced the release of Asterisk 13.1.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.1.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: New

The Asterisk Development Team has announced the release of Asterisk 13.1.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.1.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: New