Re: [asterisk-dev] SIP request
Thank you! On Sun, Dec 1, 2013 at 9:02 PM, Joshua Colp wrote: > Stas Kobzar wrote: > >> Hello list, >> >> I am trying to develop my own Asterisk module. >> I need to create and send PUBLISH SIP message with special headers >> and/or message body. >> >> I found in that in include folder there is a sip_api.h (Asterisk 11), an >> API for INFO method. But I can not figure out how to access to other >> methods. >> >> Is it possible to use chan_sip methods in other modules? If yes, could >> you, please, give me a hint where to look? >> > > There is no way to do this. It doesn't provide any APIs to extend it. Any > additional functionality has to be built into chan_sip itself. > > In Asterisk 12 the new PJSIP based modules DO provide various APIs to > allow you to do exactly this. > > Cheers, > > -- > Joshua Colp > Digium, Inc. | Senior Software Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > Check us out at: www.digium.com & www.asterisk.org > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-dev mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-dev > -- Stas Kobzar VoIP Developer 514 284 2020 www.modulis.ca -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] SIP request
Stas Kobzar wrote: Hello list, I am trying to develop my own Asterisk module. I need to create and send PUBLISH SIP message with special headers and/or message body. I found in that in include folder there is a sip_api.h (Asterisk 11), an API for INFO method. But I can not figure out how to access to other methods. Is it possible to use chan_sip methods in other modules? If yes, could you, please, give me a hint where to look? There is no way to do this. It doesn't provide any APIs to extend it. Any additional functionality has to be built into chan_sip itself. In Asterisk 12 the new PJSIP based modules DO provide various APIs to allow you to do exactly this. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] SIP request
Hello list, I am trying to develop my own Asterisk module. I need to create and send PUBLISH SIP message with special headers and/or message body. I found in that in include folder there is a sip_api.h (Asterisk 11), an API for INFO method. But I can not figure out how to access to other methods. Is it possible to use chan_sip methods in other modules? If yes, could you, please, give me a hint where to look? Thank you, -- Stas Kobzar -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] [Code Review] 3036: res_pjsip_transport_websocket: Fix crash with security events and improve implementation
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3036/ --- (Updated Dec. 1, 2013, 1:56 p.m.) Status -- This change has been marked as submitted. Review request for Asterisk Developers. Changes --- Committed in revision 403256 Bugs: ASTERISK-22897 https://issues.asterisk.org/jira/browse/ASTERISK-22897 Repository: Asterisk Description --- The attached change fixes/tweaks a few things: Security events now determine the transport type using a saner method (by looking at the transport type on the message itself), which includes WebSocket based connections. This means no having to create a container of configured transports and no having to iterate them. Connection handling now uses the built-in PJSIP transport manager for figuring out what active transport/connection to use. This is based on the target IP address/port of the active WebSocket connection. Diffs - /branches/12/res/res_pjsip_transport_websocket.c 403236 /branches/12/res/res_pjsip/security_events.c 403236 /branches/12/res/res_pjsip/pjsip_options.c 403236 /branches/12/res/res_pjsip/location.c 403236 /branches/12/res/res_pjsip.c 403236 /branches/12/include/asterisk/res_pjsip.h 403236 Diff: https://reviewboard.asterisk.org/r/3036/diff/ Testing --- Connected using JsSIP, confirmed no crash and that traffic is sent out the proper connection. Thanks, Joshua Colp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] [Code Review] 3037: Add tests for CHANNEL function for PJSIP
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3037/ --- Review request for Asterisk Developers. Repository: testsuite Description --- This exercises the various parameters in the CHANNEL function that pertain to the PJSIP channel driver. Diffs - /asterisk/trunk/tests/channels/pjsip/dialplan_functions/pjsip_channel/test-config.yaml PRE-CREATION /asterisk/trunk/tests/channels/pjsip/dialplan_functions/pjsip_channel/sipp/uac-no-hangup.xml PRE-CREATION /asterisk/trunk/tests/channels/pjsip/dialplan_functions/pjsip_channel/configs/ast1/pjsip.conf PRE-CREATION /asterisk/trunk/tests/channels/pjsip/dialplan_functions/pjsip_channel/configs/ast1/extensions.conf PRE-CREATION Diff: https://reviewboard.asterisk.org/r/3037/diff/ Testing --- Thanks, Matt Jordan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[asterisk-dev] [Code Review] 3038: Add CHANNEL function support for PJSIP
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3038/ --- Review request for Asterisk Developers. Repository: Asterisk Description --- This patch adds CHANNEL function support to chan_pjsip. Since things were getting a bit large, all dialplan functions that were in chan_pjsip have also been moved into their own file (dialplan_functions). Information that can be retrieved: * rtp,type,[media_type] - Get RTP information, including media source/destination addresses, whether or not the media is secure, etc. * rtcp,statistic,[media_type] - Get RTCP statistic information * endpoint - Get the name of the endpoint associated with this channel. Use PJSIP_ENDPOINT to get more info. * pjsip,type - Get signalling related information, including source/destination addresses, URIs in the INVITE request, whether or not the signalling is using a secure transport, etc. Note that after this patch is committed, we should go back through the CHANNEL function documentation and move all of the channel technology specific information into blocks, so that the documentation is co-located with the channel drivers themselves. Diffs - /branches/12/res/res_pjsip_t38.c 403254 /branches/12/include/asterisk/res_pjsip_session.h 403254 /branches/12/funcs/func_channel.c 403254 /branches/12/channels/pjsip/include/dialplan_functions.h PRE-CREATION /branches/12/channels/pjsip/include/chan_pjsip.h PRE-CREATION /branches/12/channels/pjsip/dialplan_functions.c PRE-CREATION /branches/12/channels/chan_pjsip.c 403254 /branches/12/channels/Makefile 403254 Diff: https://reviewboard.asterisk.org/r/3038/diff/ Testing --- See https://reviewboard.asterisk.org/r/3037 Thanks, Matt Jordan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
