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jd.girard at sysnux.pf
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 Posted: Wed May 20, 2015 11:13 pm Post subject: [asterisk-users] PJSIP CCSS |
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Hi list,
It looks like Call Completion Supplementary Services is not available
for PJSIP channels, am I right? Is there another solution?
Thanks,
- --
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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jcolp at digium.com
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| Posted: Thu May 21, 2015 5:17 am Post subject: [asterisk-users] PJSIP CCSS |
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Jean-Denis Girard wrote:
| Quote: | -----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hi list,
It looks like Call Completion Supplementary Services is not available
for PJSIP channels, am I right? Is there another solution?
|
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,
--
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
--
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jd.girard at sysnux.pf
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| Posted: Thu May 21, 2015 10:59 am Post subject: [asterisk-users] PJSIP CCSS |
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Le 21/05/2015 00:16, Joshua Colp a écrit :
| Quote: | 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?
Thanks,
- --
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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=BFyM
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jcolp at digium.com
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| Posted: Thu May 21, 2015 11:04 am Post subject: [asterisk-users] PJSIP CCSS |
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Jean-Denis Girard wrote:
| Quote: | -----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Le 21/05/2015 00:16, Joshua Colp a écrit :
| Quote: | 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?
|
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.
--
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:
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gmludo at gmail.com
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| Posted: Thu May 21, 2015 11:39 am Post subject: [asterisk-users] PJSIP CCSS |
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2015-05-21 17:59 GMT+02:00 Jean-Denis Girard <jd.girard@sysnux.pf (jd.girard@sysnux.pf)>:
| Quote: | -----BEGIN PGP SIGNED MESSAGE-----
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Le 21/05/2015 00:16, Joshua Colp a écrit :
| Quote: | 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 ?
Regards.
Â
| Quote: |
Thanks,
- --
Jean-Denis Girard
SysNux        Systèmes  Linux  en  Polynésie  française
http://www.sysnux.pf/ Tél: [url=tel:%2B689%2040.50.10.40]+689 40.50.10.40[/url] / GSM: [url=tel:%2B689%2087.79.75.27]+689 87.79.75.27[/url]
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=BFyM
-----END PGP SIGNATURE-----
--
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New to Asterisk? Join us for a live introductory webinar every Thurs:
        http://www.asterisk.org/hello
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jcolp at digium.com
Guest
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| Posted: Thu May 21, 2015 11:43 am Post subject: [asterisk-users] PJSIP CCSS |
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Ludovic Gasc wrote:
| Quote: | 2015-05-21 17:59 GMT+02:00 Jean-Denis Girard <jd.girard@sysnux.pf
<mailto:jd.girard@sysnux.pf>>:
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Le 21/05/2015 00:16, Joshua Colp a écrit :
| Quote: | 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.
[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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gmludo at gmail.com
Guest
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| Posted: Thu May 21, 2015 12:06 pm Post subject: [asterisk-users] PJSIP CCSS |
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2015-05-21 18:43 GMT+02:00 Joshua Colp <jcolp@digium.com (jcolp@digium.com)>:
| Quote: | Ludovic Gasc wrote:
| Quote: | 2015-05-21 17:59 GMT+02:00 Jean-Denis Girard <jd.girard@sysnux.pf (jd.girard@sysnux.pf)
<mailto:jd.girard@sysnux.pf (jd.girard@sysnux.pf)>>:
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  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.
Â
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jd.girard at sysnux.pf
Guest
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| Posted: Thu May 21, 2015 12:09 pm Post subject: [asterisk-users] PJSIP CCSS |
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Le 21/05/2015 06:39, Ludovic Gasc a écrit :
| Quote: | | Quote: | 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 tha
|
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t a
| Quote: | | Quote: | long time ago before the integration of CCSS in Asterisk.
I think it's possible to implement that only with dialplan and call f
|
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iles.
Yes, that's what I'm going to do.
Thanks,
- --
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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--
_____________________________________________________________________
-- 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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