similar to: Asterisk 1.2.0-beta1 tarball re-released

Does anyone have a working way of having a Cisco 7960 reload its config remotely. I have tried some of the scripts that I have found on the web, but to no avail. Thanks for the help. B. J. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040116/aa4eda3c/attachment.htm

I'm looking at version.h installed by Asterisk 1.2.3/4 - and the default value is 000000. I thought the value should be 010200. I know many people have problems compiling chan_bluetooth because of this inconsistency. Anyone has the last word on this?

I have a quick question for the list. For what reason would you have multiple instances of asterisk running on a single box? I can maybe see it if you have multiple IP addresses, but other than that I am drawing a blank. Thanks, B. J.

Hello all. We have an asterisk system set up, and we are seeing a lot of multiple DTMF digits being read by asterisk. In digging through the archives the only answer I have seen is to put in the statement relaxdtmf=yes in the zapata.conf file. Since we are not using any zapata devices, I have tried to put that statement in my sip.conf file to no avail. Any help would be appreciated as my end

I am still getting an error compiling the 1.2-Beta version. The tarball works fine, but I have never been able to compile the 1.2beta from CVS. I have been compiling CVS-HEAD on the machine for quite some time. It goes into this loop: if cmp -s include/asterisk/version.h.tmp include/asterisk/version.h ; then echo; else \ mv include/asterisk/version.h.tmp include/asterisk/version.h ; \

I am having a problem with Polycom auto-answer. I have the auto-answer working between PhoneA and PhoneB, but when I try to use the intercom between more then one phone I start having problems. PhoneA dials *3 which calls PhoneB, PhoneC, and PhoneD. All the phones ring, but only one will pick up, the rest will hang up and I get this error on Asterisk: Got SIP response 500 "Internal Server

hello friends, i have to configures asterisk n my hardware details are O.S - Ubuntu 8.04 Lts Memory - 1 GB Proccessor- core 2 duo is any one having a good link or how to related asterisk. any help,support will be higly appreciated thx -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello all. I just found out that I am no longer able to exit out of voicemail properly by hitting the 0 key, but the * key works. Asterisk comes back and says "I'm sorry, I did not understand that response" and goes on in the context. Is this a new "feature" or bug? Is anyone else having this problem? I am using Asterisk 1.0.3, and have tried it on two separate

Hello, first of all, sorry for the joke in the subject ;-) I have successfully update the speex library my software was using from 1.1.12 to 1.2beta1 (1.1.13). It is a VoIP software, where i set preprocessor VAD on and use the result of speex_preprocess() to determine if there is voice activity. I must say that it works fine...since i have update to the new version. Ive been testing, and

Hello, I'm having problem with a wildcard TE110P. As soon as I load the module (wcte11xp for kernel 2.6.10), it spawns a yellow error with or without an E1 plugged-in. Any one managed to set it up in France? Here are my files: zaptel.conf: span=1,0,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 zapata.conf: [channels] language=fr context=default switchtype=euroisdn pridialplan=unknown

Hello asterisk community, I have successfully set up asterisk as a SIP PBX and now would like to connect to the outside world using a Cisco 2600 with VIC-BRI as an ISDN gateway. This works already in the lab, but I have security concerns before conecting the gateway to the internet. I currently don't know exactly what VoIP services the Cisco runs by default besides SIP (H.323, MGCP, ...)

Hi, Between 1.6.1.9 and 1.6.1.18, handling of menuselect has changed in such a way that I cannot script non-english sound files downloading anymore. The following used to work (unattended) with 1.6.1.9 (for instance): cd /usr/src/asterisk-${ASTERISK_VERSION} ./configure make menuselect.makeopts echo "MENUSELECT_CORE_SOUNDS=CORE-SOUNDS-EN-GSM CORE-SOUNDS-FR-GSM" >

I wonder if there are CDs available for purchase. I don't have any way to burn one from a downloaded iso image. Any help appreciated. Tom

Just curious: has anyone seen or heard about a USB-based T1/E1 interface device? I've seen some serious T1/E1 testing equipment that is USB-based, but I was wondering if there was something more generic, like a Zaptel-ish T1/E1 that used USB instead of PCI/PCIx. Thanks! -MC -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello - 1. Is there any way to directly export R plots to MIF format for import into FrameMaker? 2. If not, how hard would it be to add such a device driver? I don't see device driver docs in the R extensions docs. Are drivers contained in their own source files with simple interfaces to the data structures with plot data? 3. If that's difficult, what workarounds are

Hello, I'm attempting to migrate from an existing dovecot .99 install to dovecot 1.0.rc15 on a fresh up to date CentOs 5.1. I'm not sure what's causing this permissions error on startup - I've tried 777,644, and 600 chmods but it's always this same error: Feb 1 12:43:05 ns2 dovecot: Dovecot v1.0.rc15 starting up Feb 1 12:43:06 ns2 dovecot: Auth process died too early -

Hello all. I know * doesn't directly support recording mp3 files, but I was wondering if anyone has created an AGI to do it indirectly. Thanks in advance. B. J. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040105/22d999c4/attachment.htm

Hello all. Has anyone had any success using ChanIsAvail with only SIP channels? Is there another, better way to check if an extension is busy without dialing it? Thanks, B. J. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040109/48ac2c3e/attachment.htm

The first beta of Asterisk 1.2.0 has been released! It is available from the ftp.digium.com FTP servers, as well as the Digium CVS servers (under the 'v1-2-0-beta1' tag). This version of Asterisk represents a significant improvement in features, stability and compatibility over the 1.0.x releases. Some of the major new (or upgraded) features include: * Asterisk Realtime

Guys. I was checking the changes for 1.2.0 beta1 and I read this: * Asterisk Realtime Architecture * Asterisk Manager Interface * Asterisk Extension Language * Dialplan functions * More powerful dialplan expression parser * Portability enhancements for FreeBSD, OpenBSD, Solaris and Mac OS X * ... and many more! Can somebody explain a bit mor ethe part regarding