search for: svnview

I'm currently reviewing the Digium wiki on migrating from chan_sip to res_pjip and I'm trying to access the script that is provided to help with conversion. https://wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip It would appear that said server hosting the script is no responding or the link is no longer valid. Doug

I am working on some fax tools for some of my users. I am reading the https://wiki.asterisk.org docs for faxing. Is see Application_SendFax and Application_SendeFax has one been discondinued? Any feed back on using the res_fax module would be apperciated. Any examples or other. Thanks Bryant -------------- next part -------------- An HTML attachment was scrubbed... URL:

...f servers hosting Asterisk services on Saturday, August 28, 2010 between 10am and 11am for maintenance. These services include the following sites: * packages.asterisk.org * svn.digium.com * svn.asterisk.org * svncommunity.digium.com * issues.asterisk.org * reviewboard.asterisk.org * svnview.digium.com * downloads.digium.com * downloads.asterisk.org * bamboo.asterisk.org Sorry for any inconvenience this may cause. - The Asterisk Team

I noticed that asterisk.org got a redesign, quite recently it seems, which is very nice, but the addons package isn't listed for download any longer, nor are releases posted to http://downloads.asterisk.org/pub/telephony/ . That said, looks like it's still available in svn, http://svnview.digium.com/svn/asterisk-addons/tags/1.6.1.1/ . So just wondering if addons will be around for the forseeable future? - Chris

Hi, I've gone through the source tree and I don't see a MIB description file for the SNMP agent in asterisk. can someone point me to it. Thanks, Bruce ferrell

...f servers hosting Asterisk services on Saturday, August 28, 2010 between 10am and 11am for maintenance. These services include the following sites: * packages.asterisk.org * svn.digium.com * svn.asterisk.org * svncommunity.digium.com * issues.asterisk.org * reviewboard.asterisk.org * svnview.digium.com * downloads.digium.com * downloads.asterisk.org * bamboo.asterisk.org Sorry for any inconvenience this may cause. - The Asterisk Team

Anybody know where to find some good information on the new CEL in asterisk 1.8? I'm very anxious to check out the new logging features but can't find anything but the cel.conf.sample file in the source package. I'd like to get this setup with ODBC. Thanks in advance.

Hello all, Is anyone else experiencing trouble accessing svn.digium.com? Neither webview nor svn checkout requests are successful. svnview.digium.com works but svn checkout at this address returns a "repository temporarily relocated" error... Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20121028/d0612cb5/attachment.htm...

...provide basic video conferencing in the ConfBridge() dialplan application. * Support for defining hints has been added to pbx_lua. * Replacement of Berkeley DB with SQLite for the Asterisk Database (AstDB). * Much, much more! A full list of new features can be found in the CHANGES file. http://svnview.digium.com/svn/asterisk/branches/10/CHANGES For a full list of changes in the current release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.0.0-rc1 Thank you for your continued support of Asterisk!

...provide basic video conferencing in the ConfBridge() dialplan application. * Support for defining hints has been added to pbx_lua. * Replacement of Berkeley DB with SQLite for the Asterisk Database (AstDB). * Much, much more! A full list of new features can be found in the CHANGES file. http://svnview.digium.com/svn/asterisk/branches/10/CHANGES For a full list of changes in the current release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.0.0-rc1 Thank you for your continued support of Asterisk!

...E: BLF and asterisk Queue To: ron_hartmann at hotmail.com In case my earlier email was a bit cryptic. I didn't commit the change, due to time zone differences, so don't have the actual patch that was commited. The patch that was comitted to asterisk 11 is at the following link. http://svnview.digium.com/svn/asterisk/branches/11/apps/app_queue.c?r1=37323 4 <http://svnview.digium.com/svn/asterisk/branches/11/apps/app_queue.c?r1=3732 34&r2=373235&view=patch> &r2=373235&view=patch Alec _____ From: Ron Hartmann [mailto:ron_hartmann at hotmail.com] Sent: Wedne...

Howdy, I am proud to announce that after a good bit of development, community feedback, testing, and code review, the brand new ConfBridge application has been officially merged into Asterisk Trunk!!! http://svnview.digium.com/svn/asterisk?view=revision&revision=314598 If you are already familiar with ConfBridge from Asterisk 1.6.X and 1.8, forget everything you know. This is a completely revamped, highly optimized, and feature rich conferencing application capable of mixing sample rates from 8khz all th...

Outoing calls I make using Motif Google Voice Calls continue ringing even after the other end picks up. I have to restart Asterisk to resolve the issue. I don't see any errors. It's not recognizing that the other party picked up the phone and restarting Asterisk fixes it only for a day. -- Co-op Vacation Rentals www.coopvr.com 15218 Summit Ave Suite #300-354 Fontana, CA 92336

Hello I just coding a AGI script for billing. - For external calls, I pass the call directly on a trunk. I do : Dial(trunk1/extension) -> OK ! - For internal calls (shortcode, others users ...) I am Dial(Local/extension at context/n) The problem is that through chan_local.so, I sound as it cut! Example if I call the voicemail ... "You have No messa ..." or "You have

Hi All, i would like to know if anyone has done or having idea regarding PRI terminations in asterisk. i have a requirement where i need to support 80 PRI in one machine i have found a machine which have 10 PCI slots available now i am thinking of arranging 8port sangoma card in this pci slots so i can arrenge 10 card in that. is it possible to run system like that ? is it good idea , can

Hi guys I have some questions regarding the above 1. Why are there different options for timing? 2. What are the differences between these types of timing sources? 3. When should you use what? 4. Is one timer type more "reliable" for an Asterisk system under heavy loads than another while NOT using any DAHDI hardware - SIP only? I'm of course referring to the familiar

...M CDT on September 2nd. We apologize in advance for any inconvenience this may cause. The affected services are: downloads.digium.com downloads.asterisk.org bamboo.asterisk.org git.asterisk.org code.asterisk.org packages.asterisk.org svn.asterisk.org issues.asterisk.org reviewboard.asterisk.org svnview.asterisk.org wiki.asterisk.org

...vide basic video conferencing in the ConfBridge() dialplan application. * Support for defining hints has been added to pbx_lua. * Replacement of Berkeley DB with SQLite for the Asterisk Database (AstDB). * Much, much more! A full list of new features can be found in the CHANGES file. http://svnview.digium.com/svn/asterisk/branches/10/CHANGES For a full list of changes in the current release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.0.0-beta2 Thank you for your continued support of Asterisk!

...gainst app_meetme.c in trunk, as your work will NOT make it into asterisk by posting it to asterisk-users. If you didn't already know, you should know that SVN is often way ahead of the 1.6.0. series for many reasons. For the latest versions for each 1.6 series, you should check through: http://svnview.digium.com/svn/asterisk/branches/

Hi, I am working on Trixbox. I want to create my own dial() function (named specificdial()) and I want to know how I can create a module and integrate the module in the trixbox plateform. thanks a lot Mickael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091006/b69d4103/attachment.htm