2015-05-21 18:43 GMT+02:00 Joshua Colp <[email protected]>: > Ludovic Gasc wrote: > >> 2015-05-21 17:59 GMT+02:00 Jean-Denis Girard <[email protected] >> <mailto:[email protected]>>: >> >> -----BEGIN PGP SIGNED MESSAGE----- >> Hash: SHA1 >> >> Le 21/05/2015 00:16, Joshua Colp a écrit : >> > If CCSS is needed then the only option is to use chan_sip. The >> > chan_pjsip module does not implement CCSS in any way. >> >> Is CCSS support planned for PJSIP? chan_sip is in "extended" state in >> asterisk-13, so chan_pjsip should be preferred for new installations, >> ri >> ght? >> >> >> If you really want CCSS support and to be fancy with PJSIP, you can >> easily implement a similar feature with AMI events, I already did that a >> long time ago before the integration of CCSS in Asterisk. >> I think it's possible to implement that only with dialplan and call files. >> >> In my mind, chan_sip will be dropped after asterisk 13, is it true ? >> > > It won't be dropped. It still has features which are not available in > PJSIP, and people still use it. The extended status refers to the support > level. Per the support states wiki page[1]: > > This module is supported by the Asterisk community, and may or may not > have an active developer. Some extended modules have active community > developers; others do not. Issues reported against these modules may have a > low level of support. >
Joshua, come on, you know as me that you have few people around the world to have the skills and the time to maintain a C module for Asterisk. For a critical feature like SIP in Asterisk, at least to me, it means that for a serious production with Asterisk 13, I won't use chan_sip but I'll prefer chan_pjsip. Personally, I don't care if it's pjsip or sip, I only want a telephony stack that won't piss on my shoes under the fire of a big production. However, I didn't know that some features are missing in chan_pjsip compare to chan_sip. A list exists somewhere ? Moreover, by curiosity, somebody has already benchmarked chan_sip vs chan_pjsip ? Somebody has a noticed an efficiency issue with pjsip ? Regards. > > [1] > https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States > > -- > Joshua Colp > Digium, Inc. | Senior Software Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - US > Check us out at: www.digium.com & www.asterisk.org > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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