PJSIP CCSS

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Asterisk Users 7 Comments

Hi list,

It looks like Call Completion Supplementary Services is not available for PJSIP channels, am I right? Is there another solution?

Thanks,
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Jean-Denis Girard

SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.79.75.27
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7 thoughts on - PJSIP CCSS

  • Jean-Denis Girard wrote:

    If CCSS is needed then the only option is to use chan_sip. The chan_pjsip module does not implement CCSS in any way.

    Cheers,

  • Jean-Denis Girard wrote:

    I know of noone currently working on CCSS support for PJSIP. In this case though since chan_pjsip doesn’t support what you need, chan_sip is it.

  • 2015-05-21 17:59 GMT+02:00 Jean-Denis Girard :

    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 ?

    Regards.

  • Ludovic Gasc wrote:

    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.

    [1]
    https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States

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    Joshua Colp Digium, Inc. | Senior Software Developer
    445 Jan Davis Drive NW – Huntsville, AL 35806 – US
    Check us out at: http://www.digium.com & http://www.asterisk.org

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  • 2015-05-21 18:43 GMT+02:00 Joshua Colp :

    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.