similar to: Xapian 1.2.14 released

Hi, I'm trying to compile asterisk 1.2.14 on a Debian Sarge amd64 with kernel 2.6.8-12-amd64-k8 make[2]: Entering directory `/usr/src/asterisk-1.2.14/codecs/gsm' gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -fomit-frame-pointer -fPIC -c -DNeedFunctionPrototypes=1 -funroll-loops

Hi All, How to install bristuff on asterisk 1.2.14? install scripts are trying to download and compile those versions: asterisk-1.0.10 zaptel-1.0.10 libpri-1.0.9 and I'm running: asterisk-1.2.14 zaptel-1.2.12 libpri-1.2.4 I only need Pickup application from bristuff to be able to pickup channel independent calls e.g. when I have incoming call from PSTN and I would like to answer

I upgraded my Asterisk system to version 1.2.14 to check if the sound quality issues I was having with Chanspy in 1.2.7 remained. I'm still getting them, and I'm honestly out of ideas except from RTFS. The called party sounds normally fine, but it's impossible to hear the caller. Sometimes, when the called party is talking, the caller can also be heard. The conversation sounds broken,

The Asterisk Development Team is pleased to announce the release of Asterisk 1.2.14. This release contains a number of updates: - a bug fix for the ExternalIVR application and addition of 'silence' sound files to support it - various SIP interoperability improvements - memory and dialog leaks in the SIP channel driver - a fix to music-on-hold random mode that was not really random - an

The Asterisk Development Team is pleased to announce the release of Asterisk 1.2.14. This release contains a number of updates: - a bug fix for the ExternalIVR application and addition of 'silence' sound files to support it - various SIP interoperability improvements - memory and dialog leaks in the SIP channel driver - a fix to music-on-hold random mode that was not really random - an

The Asterisk and Zaptel development team has released version 1.2.14 of Zaptel. This release contains only minor changes, the most important of which relates to single-port module support on Digium's TDM800P analog interface card (previously these modules were not properly recognized by the driver). Thanks for supporting Asterisk and Zaptel!

The Asterisk and Zaptel development team has released version 1.2.14 of Zaptel. This release contains only minor changes, the most important of which relates to single-port module support on Digium's TDM800P analog interface card (previously these modules were not properly recognized by the driver). Thanks for supporting Asterisk and Zaptel!

I sent this yesterday, but saw zero traffic, so I think it got lost in the ether, so I'm sending again. I'm having an issue using Asterisk 1.2.14 and an Audiocodes Mediant 1000 ISDN gateway. For the most part, everything is working except for attended transfers. When I do an attended transfer, and complete the transfer before the 3rd party answers, the PSTN side hears dead air until the

Nothing much to be said.. I backported ringinuse, autofill and the QueueLog application from 1.4.0 to 1.2.14. Any or all may be applied - order doesn't matter. They have received minimal testing but appear to function correctly. As always with these things, don't blame me if they connect your callers to a phonesex line, etc. http://bum.net/patches/ Cheers, Gavin.

Hi All, I'm using Asterisk 1.2.14 with Wildcard TDM400P. I need app_pickup.so application so I can pickup channel-independent calls from any IP Phone headset. How to compile and install only this application from bristufff package? Thank you in advance, Dominik

Good Day List Users, Is there any way to play an audiofile or at least a beep into an established call, I want to do this event each 3 minutes in the call, for now I have a shell to get the call time and evaluate the 3 minutes.....do you know any way to play that sound? I tried app_inject, it works really nice in asterisk 1.4.X releases; but my PBX runs 1.2.14 and It can?t be upgraded (policy

> And all was fine, before I accidentally upgraded Centos to 7.1.1503 > After upgrade LXC can't start with diagnostic: > [root@node14 ~]# virsh start test1 > error: Failed to start domain test1 > error: internal error: guest failed to start: internal error: Child > process (ip link set macvlan0 netns 25263) unexpected exit status 2: > RTNETLINK answers: Invalid argument

Hi there I am using fedora21 for hosts and guests, using lxc containers, and was on vanilla 1.2.9 as exposed by the fedora yum repo The other day I was needing the ability to grant mknod capability to some guests, some chap on IRC kindly pointed me at the <capabilities> tag However I was told that needed 1.2.11 So I went and rebuilt from this download

hi all： my environment is centos6.5 libvirt version is 1.2.14-1 qemu version is 1.7.0-1 I use openstack create a windows guest about two days later I run virsh list but the the process is hang virsh list can not return any thing it hang then I run /etc/init.d/libvirtd restart it show libivrtd stop faild and start failed I run /etc/init.d/libvirtd status it show

hi all： my environment is centos6.5 libvirt version is 1.2.14-1 qemu version is 1.7.0-1 I use openstack create a windows guest about two days later I run virsh list but the the process is hang virsh list can not return any thing it hang look like this： then I run /etc/init.d/libvirtd restart it show libivrtd stop faild and start failed i will use the gdb and

hi, I see in dovecots logfile some error messages: Sep 29 15:33:43 auth(default): Error: LDAP: Connection lost to LDAP server, reconnecting Sep 29 15:34:15 auth(default): Error: LDAP: Connection lost to LDAP server, reconnecting Sep 29 15:34:50 auth(default): Error: LDAP: Connection lost to LDAP server, reconnecting Sep 29 15:35:22 auth(default): Error: LDAP: Connection lost to LDAP server,

Hi all. I use LXC on Centos 7 x86-64, with libvirt version 1.2.6 and 1.2.12 My container has bridged network: # virsh dumpxml test1 <domain type='lxc'> <name>test1</name> <uuid>518539ab-7491-45ab-bb1d-3d7f11bfb0b1</uuid> <memory unit='KiB'>1048576</memory> <currentMemory unit='KiB'>1048576</currentMemory>

Graham Bentley wrote: > Recently swapped out my Sony for a HP DAT and > got this first time in dmesg > > (sa0:ahc0:0:6:0): PREVENT ALLOW MEDIUM REMOVAL. CDB: 1e 0 0 0 0 0 > (sa0:ahc0:0:6:0): CAM Status: SCSI Status Error > (sa0:ahc0:0:6:0): SCSI Status: Check Condition > (sa0:ahc0:0:6:0): UNIT ATTENTION asc:28,0 > (sa0:ahc0:0:6:0): Not ready to ready change, medium may

Hi all, Are there any issues to be concerned about when calls come in from PSTN to a PRI card and are forwarded back out the same PRI card? Anything different have to be enabled in zaptel.conf or zapata.conf or the Sangoma configs to make this work? What about using .call files that join two ZAP channels? Channel: ZAP/1/4081234567 MaxRetries: 0 RetryTime: 60 WaitTime: 60 Application:

Am I doing something really stupid in this AEL macro, or is nesting an 'if' inside a 'switch', inside an 'if' not supported in the 1.2 AEL parser? macro stdexten( ext , dev ) { // First determine if the SIP peer is registered here Set(aretheyhere=${SIPPEER(${ext}:status)}); if(${aretheyhere:0:2}) == "OK") {