What phone is this being used on? I am able to do call pickup on various Polycom VVX’s, Yealink’s and even old Cisco SPA3xx/5xx phones. I think I even got it on an old snom but I would have to fire it up to double check.
Granted that each the phone configs for each of those brands do ask for a pickup code (some have default codes). So knowing what phones you are trying to do this with might help solve it. Tom From: asterisk-users <[email protected]> On Behalf Of Joshua C. Colp Sent: Tuesday, March 1, 2022 6:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]> Subject: Re: [asterisk-users] Pickup with pjsip not working On Tue, Mar 1, 2022 at 7:16 AM Karsten Wemheuer <[email protected] <mailto:[email protected]> > wrote: Am Dienstag, dem 01.03.2022 um 06:37 -0400 schrieb Joshua C. Colp: > On Tue, Mar 1, 2022 at 6:14 AM Karsten Wemheuer <[email protected] > <mailto:[email protected]> > wrote: > > Hi *, > > > > i am currently trying to migrate from chan_sip to pjsip. I am using > > Asterisk version 18.10. > > > > In chan_sip information about the pickup was sent in the XML body > > of > > the NOTIFY requests: > > > > /--- > > <?xml version="1.0"?> > > <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="3" > > state="full" > > entity="sip:[email protected] <mailto:sip%[email protected]> "> > > <dialog id="213" call-id="pickup-313634363132303037373438313434- > > wojojtzh4bgd" local-tag="1f4rovwnc1" remote-tag="as0a28a768" > > direction="recipient"> > > <remote> > > .... > > \--- > > > > > > If I use pjsip, the pickup information is missing: > > > > /--- > > <?xml version="1.0" encoding="UTF-8"?> > > <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="1" > > state="full" entity="sip:[email protected]:25060 > > <http://sip:[email protected]:25060> "> > > <dialog id="213" direction="recipient"> > > <remote> > > .... > > \--- > > > > Many phones expect this information and cannot perform a pickup. > > > > Where does this need to be configured or does this not work in > > pjsip? > > It does not appear as though anyone has written support for this in > PJSIP. > Do You know, if someone is working on this? Maybe I can help. Is it part of the upstream project or would it be built somewhere into res/res_pjsip.XXX? I know of noone working on this, and it would be part of Asterisk itself. -- Joshua C. Colp Asterisk Technical Lead Sangoma Technologies Check us out at <http://www.sangoma.com> www.sangoma.com and <http://www.asterisk.org> www.asterisk.org
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
