similar to: Asterisk 1.8.5.0 Now Available

The Asterisk Development Team has announced the final maintenance release of Asterisk, version 1.6.2.19. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ Please note that Asterisk 1.6.2.19 is the final maintenance release from the 1.6.2 branch. Support for security related issues will continue until April 21, 2012. For more information

The Asterisk Development Team has announced the final maintenance release of Asterisk, version 1.6.2.19. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ Please note that Asterisk 1.6.2.19 is the final maintenance release from the 1.6.2 branch. Support for security related issues will continue until April 21, 2012. For more information

The Asterisk Development Team has announced the release of Asterisk 1.4.30. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.4.30 resolves several issues reported by the community, and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by community

The Asterisk Development Team has announced the release of Asterisk 1.4.30. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.4.30 resolves several issues reported by the community, and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by community

Hi all, I'm getting one way audio when calling over the SIP trunk i.e. end device B (remote end of SIP trunk) can hear device A (softphone registered with Asterisk) but device A can't hear device B. Even though I configured same NAT configurations on other servers and they are working good. The NAT configuration is listed below; localnet=130.0.0.0/130.0.0.0 externhost=12.131.12.13

We are running Asterisk 1.8.5.0 with an uptime of 40 weeks. Just today our streaming music on hold stopped working. I remember when we had first installed 1.8 we had an issue where the streaming music on hold would not work because Music On Hold was using the DAHDI timing module. We needed the DAHDI timing module loaded so that paging would work. However, at that time we upgraded to 1.8.5.0 and

Trying to make this work, and Office 365 support is useless, giving me the following response when I asked them for help troubleshooting a 488 Not Acceptable Here. Regarding your service request about configuring your PBX system with Office 365, we do not support specific setups for PBX systems for Unified Messaging. Please contact the vendor for more specific instructions and configurations.

High gurus, Je souhaite mettre en place un serveur samba lorsque je vais dans le voisinnage r?seau je vois bien le partage MYGROUP cependantlorsque je veux y acc?der j'ai une erreur du type : MYGROUP est inaccessible. l'ordinateur ou le nom de partage n'a pas ?t? trouv? !! controler votre entr? et r??ssayez !! Une id?e ? # Samba config file created using SWAT # from 130.0.10.3

On a BT line, how do I determine whether the number on an incoming call has been deliberately withheld (by dialling 141) or is merely unavailable (e.g. because it originated from overseas or passed through some ancient switching equipment) ? In the first case, I want the caller to be played a message to the effect that we are not at home to anonymous cowards but if their business is

Trying to setup UM with Office 365 which requires TLS. I've tried under 1.8.5.0 and under 1.6.2.16.1 and I get the same error: [Aug 11 06:50:20] VERBOSE[3023] tcptls.c: SSL certificate ok [Aug 11 06:50:20] VERBOSE[3023] tcptls.c:?? == Problem setting up ssl connection: error:00000000:lib(0):func(0):reason(0) [Aug 11 06:50:20] WARNING[3023] tcptls.c: FILE * open failed! Following the

Bonjour, Je suis ?tudiant stagiaire ? Paris et je rencontre quelques difficult?s en programmation R. J'ai une data frame compos?e de 4 colonnes et 250 lignes et dont chaque ligne est une famille. J'ai fait un tirage al?atoire avec remise des familles 250 fois ce qui m'am?ne ? une nouvelle dataframe. A cette nouvelle dataframe, j'applique un programme qui calcule 2 param?tre X1

CentOS 5.5 Xen "standard" Xen Installation. I have two nics. I just put the second one to DHCP and modified the ifcfg-et01 and so far I am holding, but I am not confident. Prior they were sequential IP Addrs on same subnet. arpwatch has indicated flip flips. I can find no rhyme or reason to predict them. I know I missed I must have missed a step somewhere. I want to keep the

Hi, I am running asterisk 1.8.5.0 and have compiled in the srtp module All but Snom phones are working. I have set the srtp tag on the snoms to 80 and RTP/SAVP to mandatory and they worked for a few hours. This morning all snoms are reporting this when trying to make a call (this is snom calling snom). ---------snip------------------ == Using SIP RTP CoS mark 5 -- Executing [10000 at

Hi, When starting Asterisk (1.8.5.0) I see in messages: [Jul 18 15:47:50] WARNING[15491] loader.c: Error loading module 'chan_gtalk.so': /usr/lib/asterisk/modules/chan_gtalk.so: undefined symbol: ast_aji_get_client [Jul 18 15:47:50] WARNING[15491] loader.c: Module 'chan_gtalk.so' could not be loaded. Yet I do have iksemel installed: ls -l /usr/local/lib/libik* -rw-r--r-- 1

The Asterisk Development Team has announced the release of Asterisk 1.8.4.1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.4.1 resolves several issues reported by the community. Without your help this release would not have been possible. Thank you! Below is a list of issues resolved in this release: * Fix

The Asterisk Development Team has announced the release of Asterisk 1.8.4.1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.4.1 resolves several issues reported by the community. Without your help this release would not have been possible. Thank you! Below is a list of issues resolved in this release: * Fix

Hi! I would like some advices to combine security and automated backup... How rsync manage hosts allow parameter compared to /etc/hosts.allow? Is it possible to use ssh key just for authentification not for crypted transfert? I have also a problem to preserve owner:group properties even with archive parameter which is supposed to do that. I have to put gid =root and uid = root in

Bonjour, D?apr?s votre question ? Jean_Marc : ? Je m'int?resse ? la source 1.0.5, et je ne vois pas une _spx_lpc (). Il ya une _spx_autocorr (), qui est en lpc.c et est appel? ? proximit? du d?part de la Codeur fonction dans nb_celp.c. L'encodeur semble appeler la autocorr () la fonction, puis appelle wld () pour faire quelque chose qui s'appelle Levinson-Durbin.

Hi All; Asterisk version is: 1.8.5.0 But I see at the consol the following warning and really I did google but did not understand if it is bug or related to settings: [Sep 13 15:04:56] WARNING[2209]: chan_sip.c:19667 handle_response_invite: just did sched_add waitid(3429468) for sip_reinvite_retry for dialog 3c581fa96f2b-53yysntgjmwb in handle_response_invite But actually, we see some SNOM IP

Hi All; Suddenly, we restarted the Asterisk machine and the echo appeared. The lines are analoge. At the consol, I see this message: [Aug 10 14:36:05] WARNING[3789]: chan_dahdi.c:4831 dahdi_enable_ec: Unable to enable echo cancellation on channel 1 (No such device) [Aug 10 14:36:07] WARNING[3789]: chan_dahdi.c:4831 dahdi_enable_ec: Unable to enable echo cancellation on channel 1 (No such