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-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Le 21/05/2015 00:16, Joshua Colp a ?crit : > 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...

Ludovic Gasc wrote: > 2015-05-21 17:59 GMT+02:00 Jean-Denis Girard <jd.girard at sysnux.pf > <mailto:jd.girard at sysnux.pf>>: > > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Le 21/05/2015 00:16, Joshua Colp a ?crit : > > 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...

-----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? Thanks, - -- Jean-Denis Girard SysNux Syst?mes Linux en Polyn?sie fran?aise http://www.sysnux.pf/ T?l: +689 40.50.10.40 / GSM: +689 87.79.75.27 -----BEGIN PGP SIGNATURE-----

2015-05-21 17:59 GMT+02:00 Jean-Denis Girard <jd.girard at sysnux.pf>: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Le 21/05/2015 00:16, Joshua Colp a ?crit : > > 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? &g...

Jean-Denis Girard wrote: > -----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

...d custom function 'FEATURE' == Registered custom function 'FEATUREMAP' == Parsing '/etc/asterisk/enum.conf': Found == Registered application 'CallCompletionRequest' == Registered application 'CallCompletionCancel' [2013-01-10 14:20:10] WARNING[27062]: ccss.c:4278 initialize_cc_max_requests: Could not find valid ccss.conf file. Using cc_max_requests default [2013-01-10 14:20:10] WARNING[27062]: ccss.c:4335 initialize_cc_devstate_map: Could not find valid ccss.conf file. Using cc_[state]_devstate defaults Asterisk Dynamic Loader Starting: == Parsing...

...9;: Found == Parsing '/usr/local/etc/asterisk/users.conf': Found == Parsing '/usr/local/etc/asterisk/enum.conf': Found == Registered application 'CallCompletionRequest' == Registered application 'CallCompletionCancel' == Parsing '/usr/local/etc/asterisk/ccss.conf': Found == Parsing '/usr/local/etc/asterisk/ccss.conf': Found Segmentation fault (core dumped) root at asterisk:~ # I used gdb, to get some backtrace data: (gdb) bt #0 0x0000000803261970 in strcasecmp_l () from /lib/libc.so.7 #1 0x0000000000532926 in media_info_cmp () #2 0x0...

Hi I need to set the number of incoming calls to one, but the outgoing calls should be unlimited. I think the busylevel parameter is for it(incoming calls), but not works. My config is: cat sip.conf [general] [template](!) qualify=yes cc_agent_policy=generic cc_monitor_policy=generic call-limit=2 busylevel=1 callcounter=yes subscribecontext = hint allowsubscribe=yes [100](template)

...o/rec_delete.o recno/rec_get.o recno/rec_open.o recno/rec_put.o recno/rec_search.o recno/rec_seq.o recno/rec_utils.o -> libdb1.a [LD] abstract_jb.o acl.o alaw.o aoc.o app.o ast_expr2.o ast_expr2f.o asterisk.o astfd.o astmm.o astobj2.o audiohook.o autochan.o autoservice.o bridging.o callerid.o ccss.o cdr.o cel.o channel.o chanvars.o cli.o config.o data.o datastore.o db.o devicestate.o dial.o dns.o dnsmgr.o dsp.o enum.o event.o features.o file.o fixedjitterbuf.o frame.o framehook.o fskmodem.o global_datastores.o hashtab.o heap.o http.o image.o indications.o io.o jitterbuf.o loader.o lock.o log...

..."yes". (Closes issue #17756. Reported by oej) * Remove current STUN support from chan_sip.c. This change removes the current broken/useless STUN support from chan_sip. (Closes issue #17622. Reported by philipp2. Review: https://reviewboard.asterisk.org/r/855/) * PRI CCSS may use a stale dial string for the recall dial string. If an outgoing call negotiates a different B channel than initially requested, the saved original dial string was not transferred to the new B channel. CCSS uses that dial string to generate the recall dial string. (Patched by...

..."yes". (Closes issue #17756. Reported by oej) * Remove current STUN support from chan_sip.c. This change removes the current broken/useless STUN support from chan_sip. (Closes issue #17622. Reported by philipp2. Review: https://reviewboard.asterisk.org/r/855/) * PRI CCSS may use a stale dial string for the recall dial string. If an outgoing call negotiates a different B channel than initially requested, the saved original dial string was not transferred to the new B channel. CCSS uses that dial string to generate the recall dial string. (Patched by...

...o/rec_delete.o recno/rec_get.o recno/rec_open.o recno/rec_put.o recno/rec_search.o recno/rec_seq.o recno/rec_utils.o -> libdb1.a [LD] abstract_jb.o acl.o alaw.o aoc.o app.o ast_expr2.o ast_expr2f.o asterisk.o astfd.o astmm.o astobj2.o audiohook.o autochan.o autoservice.o bridging.o callerid.o ccss.o cdr.o cel.o channel.o chanvars.o cli.o config.o data.o datastore.o db.o devicestate.o dial.o dns.o dnsmgr.o dsp.o enum.o event.o features.o file.o fixedjitterbuf.o frame.o framehook.o fskmodem.o global_datastores.o hashtab.o heap.o http.o image.o indications.o io.o jitterbuf.o loader.o lock.o log...

...isk... same => n,Set(CONNECTEDLINE(name)=User is Busy) ...sort of 'Reverse Caller ID Name' to immediately change what the caller sees on their phone display. More involved is the CALLCOMPLETION function eg. for automating redials to the busy user when they hang up their call (see the ccss.conf.sample file.) Kind Regards, -- 🤠 C. Maj, Technology Captain @ Penguin PBX Solutions 📞 USA Toll Free 1-833-PNGNPBX (1-833-764-6729) 🤙 International & SMS Texting +1.720.32.42.72.9 🐧 Visit on the World Wide Web at PENGUINPBX.COM

Hi All, One of our user asked the question, when she tries to call another local extension but the other end is engaged she will keep on trying until she finally can get thru. So she asked would it be possible to request for an auto-callback from the user she's trying to call to once it's not engaged anymore. is this possible on asterisk? what is that feature called? i am using

Everybody, I've had a request from my manager that I figure out how to get our Asterisk 13.x system using chan_sip to be able to display on the Polycom VVX series phone display (firmware 5.9.5), when an extension is called and the person on the other end is on the phone. He said, "Our old Analog phone system could do it, how hard can it be?" I've gone down the path of trying

...Fix voicemail sequencing for file based storage. (Closes issue #18498, #18486. Reported by JJCinAZ, bluefox. Patched by jpeeler) * Set hangup cause in local_hangup so the proper return code of 486 instead of 503 when using Local channels when the far sides returns a busy. Also affects CCSS in Asterisk 1.8+. (Patched by twilson) * Fix issues with verbose messages not being output to the console. (Closes issue #18580. Reported by pabelanger. Patched by qwell) * Fix Deadlock with attended transfer of SIP call (Closes issue #18837. Reported, patched by alecdavis. Tested by...

...Fix voicemail sequencing for file based storage. (Closes issue #18498, #18486. Reported by JJCinAZ, bluefox. Patched by jpeeler) * Set hangup cause in local_hangup so the proper return code of 486 instead of 503 when using Local channels when the far sides returns a busy. Also affects CCSS in Asterisk 1.8+. (Patched by twilson) * Fix issues with verbose messages not being output to the console. (Closes issue #18580. Reported by pabelanger. Patched by qwell) * Fix Deadlock with attended transfer of SIP call (Closes issue #18837. Reported, patched by alecdavis. Tested by...

...;/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/users.conf': Found == Parsing '/etc/asterisk/enum.conf': Found == Registered application 'CallCompletionRequest' == Registered application 'CallCompletionCancel' == Parsing '/etc/asterisk/ccss.conf': Found == Parsing '/etc/asterisk/ccss.conf': Found Asterisk Dynamic Loader Starting: == Parsing '/etc/asterisk/modules.conf': Found [Dec 10 20:21:06] NOTICE[16058]: loader.c:1446 load_modules: 263 modules will be loaded. Killed [root at localhost sounds]# asterisk -vv...

Hi, I would like to implement an auto-redial function in a context. The idea is about like this: Dial a number Hear busy Hangup Pick up again Dial a code like *123 => jumps into a context which redials until callresult is not busy Maybe like this: [autoredial] exten => s,1,Set(number=${CHANNEL(lastdialed)}) exten => s,2,Dial(SIP/${number}@account,60,g) exten => s,3,Wait(15) exten

...FAXOPT(gateway)=yes with Local channels (Reported by Etienne Lessard) * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in T.38 query option with features bridging code (Reported by David Brillert) * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels. (Reported by Richard Mudgett) * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have multiple native formats (Reported by Matt Jordan) * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-...