Comments for Asterisk Ideas and Wishlist Repository http://www.asteriskideas.org What's missing in your Asterisk? Add it to this repository! Fri, 12 Mar 2010 06:06:52 +0000 http://wordpress.org/?v=2.3.1 Comment on Memcached dialplan function by oej http://www.asteriskideas.org/functions/2008-01/memcached-dialplan-function/#comment-24 oej Sat, 03 Jan 2009 09:56:49 +0000 http://www.asteriskideas.org/functions/2008-01/memcached-dialplan-function/#comment-24 Thinking about it a bit more, this is a very good idea. Thinking about it a bit more, this is a very good idea.

]]> Comment on Set debug/verbose LEVEL peer XXXX by oej http://www.asteriskideas.org/core/2009-01/set-debugverbose-level-peer-xxxx/#comment-22 oej Sat, 03 Jan 2009 09:29:09 +0000 http://www.asteriskideas.org/core/2009-01/set-debugverbose-level-peer-xxxx/#comment-22 Interesting idea. The problem is that devices, like the SIP peer, are bound to channels and doesn't have a global namespace. A SIP peer xxx might have a different address than IAX peer xxx. Interesting idea. The problem is that devices, like the SIP peer, are bound to channels and doesn’t have a global namespace. A SIP peer xxx might have a different address than IAX peer xxx.

]]> Comment on Support for SIPConnect by david6p2 http://www.asteriskideas.org/channels/sip/2008-04/support-sipconnect/#comment-21 david6p2 Tue, 30 Sep 2008 17:25:28 +0000 http://www.asteriskideas.org/channels/sip/2008-04/support-sipconnect/#comment-21 I'm making my Bachelor Thesis in the technical recommendations of SIP Connect. And I'm making an scenario with Asterisk and Kamailio(OpenSER), with 2 Kamailio servers acting as SIP Proxy Servers as it is recommended in SIP Connect. Now I'm trying to have TLS connection between these 2 servers, so if anybody could help me, it will be very nice, because right now, I haven't find any example of an scenario like the one I want to use. I think that many of the recommendations of SIP Connect will or can be implemented in 1.6 version of asterisk. I hope I could help for this propose and you could help me also. I’m making my Bachelor Thesis in the technical recommendations of SIP Connect. And I’m making an scenario with Asterisk and Kamailio(OpenSER), with 2 Kamailio servers acting as SIP Proxy Servers as it is recommended in SIP Connect. Now I’m trying to have TLS connection between these 2 servers, so if anybody could help me, it will be very nice, because right now, I haven’t find any example of an scenario like the one I want to use. I think that many of the recommendations of SIP Connect will or can be implemented in 1.6 version of asterisk. I hope I could help for this propose and you could help me also.

]]> Comment on XML configuration driver by eliel http://www.asteriskideas.org/res/realtime-res/2008-01/realtime-xml/#comment-20 eliel Mon, 29 Sep 2008 17:37:05 +0000 http://www.asteriskideas.org/res/realtime-res/2008-01/realtime-xml/#comment-20 We are using libxml2 (in a abstracted way to support other xml library) for accessing the new XML applications/functions documentation. It will be possible to use this API to read the configuration. see: http://svn.digium.com/svn/asterisk/team/group/appdocsxml We are using libxml2 (in a abstracted way to support other xml library) for accessing the new XML applications/functions documentation. It will be possible to use this API to read the configuration.

see: http://svn.digium.com/svn/asterisk/team/group/appdocsxml

]]> Comment on Match SIP peers before SIP users by oej http://www.asteriskideas.org/channels/sip/2008-08/match-sip-peers-sip-users/#comment-19 oej Thu, 14 Aug 2008 17:43:08 +0000 http://www.asteriskideas.org/channels/sip/2008-08/match-sip-peers-sip-users/#comment-19 We need to revise peer matching for a new version of Asterisk or chan_sip, but this will break backwards compatibility and be a serious change. Maybe it can be done with an option and disabled by default. We need to revise peer matching for a new version of Asterisk or chan_sip, but this will break backwards compatibility and be a serious change.

Maybe it can be done with an option and disabled by default.

]]> Comment on AMI over Jabber by hellc2 http://www.asteriskideas.org/dev/ami/2008-02/ami-xmpp/#comment-17 hellc2 Sat, 17 May 2008 15:55:04 +0000 http://www.asteriskideas.org/dev/ami/2008-02/ami-xmpp/#comment-17 I did a very simple daemon coded in Perl that allow do it. You can see it here: http://www.sinologic.net/proyectos/astjabot/ It's very simple modify this daemon to get some AMI events into your jabber client. I did a very simple daemon coded in Perl that allow do it.
You can see it here:
http://www.sinologic.net/proyectos/astjabot/

It’s very simple modify this daemon to get some AMI events into your jabber client.

]]> Comment on XML configuration driver by IgorG http://www.asteriskideas.org/res/realtime-res/2008-01/realtime-xml/#comment-16 IgorG Tue, 04 Mar 2008 11:06:06 +0000 http://www.asteriskideas.org/res/realtime-res/2008-01/realtime-xml/#comment-16 Good idea, XML will very useful for auto-generated configs. But for human good idea to make configs Apache-like, it'll help to decrease number of config files. Good idea, XML will very useful for auto-generated configs. But for human good idea to make configs Apache-like, it’ll help to decrease number of config files.

]]> Comment on Dial by profile by randulo http://www.asteriskideas.org/apps/dial/2008-01/dialtemplates/#comment-14 randulo Sat, 26 Jan 2008 09:29:48 +0000 http://www.asteriskideas.org/apps/dial/2008-01/dialtemplates/#comment-14 Yes, excellent idea! Yes, excellent idea!

]]> Comment on H.323 support missing T.38 passthrough by oej http://www.asteriskideas.org/channels/h323/2008-01/h323-support/#comment-13 oej Wed, 23 Jan 2008 19:05:32 +0000 http://www.asteriskideas.org/channels/h323/2008-01/h323-support/#comment-13 Well, development relies on active users. We need more people that work with the H.323 channel. I would also say that video is missing from the H.323 channel today. Well, development relies on active users. We need more people that work with the H.323 channel.

I would also say that video is missing from the H.323 channel today.

]]> Comment on Dial by profile by vhatz http://www.asteriskideas.org/apps/dial/2008-01/dialtemplates/#comment-12 vhatz Wed, 23 Jan 2008 16:48:19 +0000 http://www.asteriskideas.org/apps/dial/2008-01/dialtemplates/#comment-12 Also, if we could specify options like codes or even protocol specific parameters it would be excellent. Also, if we could specify options like codes or even protocol specific parameters it would be excellent.

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