similar to: Asterisk 1.4.27, 1.6.0.18, and 1.6.1.10 Now Available

Xapian 1.4.27 can now be downloaded from: https://xapian.org/download This release is mainly composed of bug fixes. The wiki has the usual summary of the most notable changes: https://trac.xapian.org/wiki/ReleaseOverview/1.4.27 A big thanks to Matthieu Gautier, David Gessel and ????? ???????? for helping to make this release a reality. If I've missed anyone out, you can claim an extra

Hello. I just upgraded from 1.6.0.9 to 1.6.1.10 and it seems that the Music On Hold functionality has changed (or is bugged?). I have Aastra 6757i and Aastra 6731i phones, and now when i press the MusicOnHold button / change lines on the phone, MOH no longer starts. It did this in v 1.6.0.9. The invites received are exactly the same, only 1.6.1.10 doesn't ever start MOH. Is there some

On 1.6.0.18-rc3 using app_fax.so, spandsp-0.0.5, anytime I use SendFAX asterisk restarts: [Nov 15 19:00:36] VERBOSE[17013] logger.c: -- Executing [s at fax-tx-test:1] ESC[1;36;40mNoOpESC[0;37;40m("ESC[1;35;40mSIP/nhi-rive rside-sip-00000000ESC[0;37;40m", "ESC[1;35;40mContext fax-tx-testESC[0;37;40m") in new stack [Nov 15 19:00:36] VERBOSE[17013] logger.c: --

The Asterisk Development Team has announced the release of libpri 1.4.15. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/libpri The release of libpri 1.4.15 resolves several issues reported by the community and would not have been possible without your participation. Thank you! The following are the issues resolved in this release: * --- Fix

The Asterisk Development Team has announced the release of libpri 1.4.15. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/libpri The release of libpri 1.4.15 resolves several issues reported by the community and would not have been possible without your participation. Thank you! The following are the issues resolved in this release: * --- Fix

The Asterisk Development Team has announced security releases for Asterisk as the following versions: * 1.2.36 * 1.4.26.3 * 1.6.0.17 * 1.6.1.9 These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of 1.2.36 resolves an issue where sending a REGISTER with a differing username in the From URI and Authorization header

The Asterisk Development Team has announced security releases for Asterisk as the following versions: * 1.2.36 * 1.4.26.3 * 1.6.0.17 * 1.6.1.9 These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of 1.2.36 resolves an issue where sending a REGISTER with a differing username in the From URI and Authorization header

I've run into a odd issue where inbound calls to the SIP client work fine, but outbound from the SIP client do not. The path between the client and the server is as below. N900 SIP client <-- OpenVPN --> Asterisk The version of Asterisk in question is 1.6.0.18. Any suggestions? -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type:

I'm trying to connect a sip call from sipgate to Asterisk A to Asterisk B. Both are behind NAT, but port forwarded. I get the connection, but no voice - either in or out. I can call on SIP from A to B (and from B to A). Do it all the time. Asterisk A receives SIP calls from Junction and Teliax. CLI on A looks right: == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 ==

Greetings- First, my apologies for the OT post. Yes, I understand this is not the FreePBX-users mailing list. But, there are a large number of people that use FreePBX and I'm hoping they can be of assistance. I have a system running Asterisk 1.4.27 (see... relevance!!!) and FreePBX 2.6.0. There are a large number of inbound routes configured for the various DID's coming in via PRI, SIP,

Hi out there, I think i've everything set up properly, outbound calls are working fine, but at incoming calls I can't hear anything, but the other one is able to hear me perfectly. I'm using an asterisk 1.6.1.10 in my internal network in a NAT, connected to my sip-provider using a trunk. Firewall settings on the router are: forward UDP port 5060,5004,10000-20000 to asterisk server

Hello. All. I am a bit new to asterisk, started from half a month ago. I am setting up a home asterisk server with analog card. I am using asterisk 1.4.27. At the moment, I bought a X100P card and installed it on my computer. I used it to connect my home phone line. For the moment, it works fine when dial in. Soon I noticed when I dial out through it to my mobile, it can't hang up

Hi, Does anyone know who did the prompts for French and Russian for Asterisk? I need some custom prompts. Regards, Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160511/ae5eea65/attachment.html>

Hello , I have a problem to install asterisk with ldap. I am doing the following: make clean . / configure make menuselect LIBS =- lldap export LIBS make ====> This is where my error #make CC = "cc" CXX = "g + +" LD = "" AR = "" RANLIB = "" CFLAGS = "" make-C menuselect CONFIGURE_SILENT = "- silent" menuselect make [1]:

Hello, When I run the 'smbclient -L apollo' command at the UNIX prompt a I get an error. I have gone as far as I can in this DIAGNOSIS.txt file. Does anyone have any thoughts on what may be causing my problem ? I have made my comments to the DIAGNOSIS.txt file via '---- davidw {comment}' so you can see what I've done to date. Also, I thought I'd pass on the output

Hello, I have an Asterisk system in the UK using ISDN service from BT. My problem is that the called number is always passed to the provider with the Type Of Number declared as ?national? despite pridialplan (and prilocaldialplan) is set to ?unknown?. My questions are: - How can I find out what the effective value is for the above two variables without actually making a call? (I have very

Hi, Does anybody know where I can get an untouched smb.conf file? When I used samba-swat, it rewrote smb.conf and deleted out the comments that was in the original file that came with samba. Minh --- samba-request@lists.samba.org wrote: > Send samba mailing list submissions to > samba@lists.samba.org > > To subscribe or unsubscribe via the World Wide Web, > visit >

Hi I'm a problem with installing "rJava" in Linux I have got in my system : java-sun, java-sun-tools, java-sun-jre but if I wont install rJava I got a problem like this: > install.packages("rJava", dependencies = TRUE) --- Please select a CRAN mirror for use in this session --- trying URL 'http://cran.mirroring.de/src/contrib/rJava_0.8-2.tar.gz' Content type

Hello List, it is a very long time since I wrote here.... It has been still in Zaptel times.... Today I am run into a related problem: I can't get a DAHDI setup to work 100%. I am configuring an Astribank XR00013 (BRI, two ISDN ports). At some degree the installation (latest DAHDI drivers, Asterisk 1.6.0.18 on Centos 5.x) works. I can get incoming calls to the dialplan context that is

I'm new to R, so apologies if the answer to my question is very obvious to everyone else! I use the carbon version of R (1.5.1), not the Darwin version. After a bit of a battle, I'm getting to grips with using R, and really getting to like it, but I'm still having trouble customising startup the way I'd like it. Reading ?Startup, I see that I can specify some commands to be run