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...a table. But I'm struggling to figure out what that configuration is. It seems the object is called "contact_status", and I've tried [2] in sorcery.conf, But I'm not seeing any output or seeing anything else to follow with GoogleFu. [1], A mailinglist crawler, i know: https://asteriskfaqs.org/2015/10/29/asterisk-users/pjsip-and-rtt-in-realtime.html [2], sourcery.conf [res_pjsip] ******stuff*********** contact_status=realtime,ps_contact_status (Which I've also defined in extconfig) *Nick Olsen* Network Engineer Office: 321-408-5000 x103 Mobile: 321-794-0763 -------------- next p...

...ref=http%253A%252F%252Fwww.freepbx.o rg%252Fforum%252Ffreepbx%252Fusers%252Ftransfer-bug-on-asterisk-1-4-38-1-6-2 -15-and-1-8-0-1> http://www.freepbx.org/forum/freepbx/users/transfer-bug-on-asterisk-1-4-38-1 -6-2-15-and-1-8-0-1 <http://www.odesk.com/leaving_odesk.php?ref=http%253A%252F%252Fasteriskfaqs. org%252F2010%252F10%252F21%252Fasterisk-users%252Fblind-transfer-failed-sip- refer-method.html> http://asteriskfaqs.org/2010/10/21/asterisk-users/blind-transfer-failed-sip- refer-method.html <http://www.odesk.com/leaving_odesk.php?ref=https%253A%252F%252Fissues.aster isk.org%252Fview.php%...

...complete-2.4.1.2+2.4.1/linux/drivers/dahdi/dahdi-base.c:8652: error: invalid use of undefined type ???struct module??? /usr/src/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi/dahdi-base.c:8652: error: ???struct dahdi_chan??? has no member named ???pulsecount??? etc etc This article (http://asteriskfaqs.org/2011/01/30/asterisk-users/invalid-use-of-undefined-type-struct-module.html) indicates that those errors are the result of not having CONFIG_MODULES set in the kernel config. cd /home/user/rpmbuild/BUILD/kernel-2.6.18/linux-2.6.18.x86_64 make menuconfig [*] Enable loadable module support Lege...

....so: undefined symbol: pjsip_tls_transport_start2 ERROR[6253]: loader.c:2396 load_modules: Error loading module 'res_pjsip_config_wizard.so': /usr/lib/asterisk/modules/res_pjsip_config_wizard.so: undefined symbol: ast_sip_get_sorcery After a lot of investigation I found this post ( https://asteriskfaqs.org/2018/09/25/asterisk-users/asterisk-1561-symbol-pjsip_tls_transport_start2-not-found.html) which pointed me to the with-ssl parameter. So if I run all previous commands, but remove the "--with-ssl" options from the configure, then I don't have the pjsip errors when I start Asterisk...

I am doing dynamic conference bridges using confbridge in asterisk 11. Is there a way to toggle off an on recording of an ongoing conference call I have figured out how to record a conference if it is turned on when someone enters. Also I have noticed that when setting music_on_hold_class dynamically it does not override what is set on the channel. exten =>

...ng to figure out what that > configuration is. It seems the object is called "contact_status", and I've > tried [2] in sorcery.conf, But I'm not seeing any output or seeing anything > else to follow with GoogleFu. > > [1], A mailinglist crawler, i know: > https://asteriskfaqs.org/2015/10/29/asterisk-users/pjsip-and-rtt-in-realtime.html > [2], sourcery.conf > [res_pjsip] > ******stuff*********** > contact_status=realtime,ps_contact_status (Which I've also defined in > extconfig > ) > Contact status is not a sorcery object and is kept strictly in...

Hi, I have a problem with asterisk 10.8, hints and fxs dahdi phones: if a remote peer and an fxs phone gets connected and the remote peer hangsup, then asterisk sends the "Idle" state to notify the watcher before you hangup the fxs phone! Such a way if the user forgets to hangup the fxs phone (which is a cordless for example) then the operators will keep sending calls to him

...ds things like multiply defined symbols and includes testing? Thanks in advance. [1] https://issues.asterisk.org/jira/browse/ASTERISK-21720 [2] http://clang.debian.net/status.php?version=3.3&key=MULTIPLE_DEF [3] http://lists.digium.com/pipermail/asterisk-users/2011-June/263933.html [4] http://asteriskfaqs.org/tag/information/page/6 ...

...at >> configuration is. It seems the object is called "contact_status", and I've >> tried [2] in sorcery.conf, But I'm not seeing any output or seeing anything >> else to follow with GoogleFu. >> >> [1], A mailinglist crawler, i know: >> https://asteriskfaqs.org/2015/10/29/asterisk-users/pjsip-and-rtt-in-realtime.html >> [2], sourcery.conf >> [res_pjsip] >> ******stuff*********** >> contact_status=realtime,ps_contact_status (Which I've also defined in >> extconfig >> ) >> > > Contact status is not a s...

...ransport_start2 > ERROR[6253]: loader.c:2396 load_modules: Error loading module > 'res_pjsip_config_wizard.so': > /usr/lib/asterisk/modules/res_pjsip_config_wizard.so: undefined symbol: > ast_sip_get_sorcery > > After a lot of investigation I found this post ( > https://asteriskfaqs.org/2018/09/25/asterisk-users/asterisk-1561-symbol-pjsip_tls_transport_start2-not-found.html) > which pointed me to the with-ssl parameter. > So if I run all previous commands, but remove the "--with-ssl" options > from the configure, then I don't have the pjsip errors when I...

...of syntax processing and not trivial to deal with. PJSIP doesn't dial all contacts when dialing an aor, so one needs to use PJSIP_DIAL_CONTACTS. However, that can return the empty string and thus lead to a syntax error, leading to the need to write code to fix formatting: https://asteriskfaqs.org/2019/06/09/asterisk-users/dialpjsip_dial_contactsalice-pjsip_dial_contactsbob-how-not-to-fail-if-one-endpoint-has-no-registered-aor.html For the second issue, it would be nice if Dial just discarded empty destinations, as in Dial(PJSIP/foo&) Dial(PJSIP/foo&&PJSIP/baz) as woul...

(I'm new to Asterisk, after having started VOIP with vat on the mbone in the 90s.) I am setting up my first Asterisk system, and trying to read docs/guidance and follow best practices. I have read the 5th Edition of "Asterisk: The Definitive Guide" and like the 3rd Edition on the web it recommends that hardphones and softphones both have a unique name distinct from any concept of

...OR[6253]: loader.c:2396 load_modules: Error loading module >> 'res_pjsip_config_wizard.so': >> /usr/lib/asterisk/modules/res_pjsip_config_wizard.so: undefined symbol: >> ast_sip_get_sorcery >> >> After a lot of investigation I found this post ( >> https://asteriskfaqs.org/2018/09/25/asterisk-users/asterisk-1561-symbol-pjsip_tls_transport_start2-not-found.html) >> which pointed me to the with-ssl parameter. >> So if I run all previous commands, but remove the "--with-ssl" options >> from the configure, then I don't have the pjsip e...

Hello, I just made a asterisk / dahdi fresh install on Debian 8.4, and ended up with the following packages : $ sudo dpkg -l|grep -Ei 'dahdi|asterisk|libpri' ii asterisk 1:11.13.1~dfsg-2+b1 amd64 Open Source Private Branch Exchange (PBX) ii asterisk-config 1:11.13.1~dfsg-2 all Configuration