search for: app_transfer

...atus isdn switch-type primary-ni isdn incoming-voice voice no isdn outgoing display-ie no cdp enable ! ip http server ! ip classless ip route 0.0.0.0 0.0.0.0 66.xxx.xxx.xxx ! ! access-list 52 permit 65.xx.xx.xxx snmp-server community 3Z8pbXn3 RO 52 ! ! ! control-plane ! ! call application voice app_transfer flash:current.tcl call application voice app_transfer max-fwd-cnt 2 call application voice app_transfer language 1 en call application voice app_transfer set-location en 0 flash:/prompt ! voice-port 1/0:23 description dtci 2x9 ! voice-port 2/0 idle-voltage low ! voice-port 2/1 ! voice-port 2/2 !...

I've added an important new application: app_transfer. This application is designed to allow Asterisk to request the transfer of an incoming call to a different extension. Consider the following diagram: Caller -> [ PBX1 ] -> SIP or IAX2 -> [PBX2] -> Transfer App A caller calls an extension on PBX1 which forwards to PBX2. PBX2 execute...

...this is happening! It have checked everything but running out of options! [app_voicemail2.so] => (Comedian Mail (Voicemail System)) == Parsing '/etc/asterisk/voicemail.conf': Found == Registered application 'VoiceMail2' == Registered application 'VoiceMailMain2' [app_transfer.so] => (Transfer) == Registered application 'Transfer' [codec_ilbc.so] => (iLBC/PCM16 (signed linear) Codec Translator) Illegal instruction thanks hallian _________________________________________________________________ The new MSN 8: advanced junk mail protection and 2 months FR...

...943825 at internal_extensions-f967;1 and a Dial via Local channel to the SIP provider is made. However, two channels are consumed and bandwith is wasted. Second approach: I tried to catch the redirect in the call forward context which is defined before the Dial statement: and do the redirect with app_transfer. e.g. snippet from macro-stdexten: exten => s,n,Set(_FORWARD_CONTEXT=from_sip_forward) exten => s,n,Dial(${ARG2},${ARG4},tTfwW) [from_sip_forward] include => internal_devices exten => _X.,1,Verbose(1,${CALLERID(num)} tries call forward to ${EXTEN} for device ${CALLERID(rdnis)}) exten...

Does * support ISDN Primary Rate Interface (PRI) - 2B Transfer Capability for T-1/PRI? In other words the ability to take a call and join it to another call and then drop off letting the CO-switch take over. -Kevin Kevin Fjelsted, President AltiCom CTI, Inc. Track Me Down! One number Access, Press 11# during the voice mail message greeting to have me F-O-U-N-D! Phone: 612.259.0722 Fax:

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, As I've been unable to get app_transfer to work, could someone explain how it is supposed to work? Currently I have two Asterisk boxes. A call comes in via zaptel to ast1. ast1 dials ast2 using iax2 and gets instructed to transfer the call to a different extension. iax2 debug shows that a transfer cmd is sent to ast1, but nothing ha...

...e is a kind of handle leak here. Nothing else runs in the box If there is a way to find out what happens, let me know. The dialplan is confidential, for it belongs to my customer,but I can give you access to the box. In short , the app receives a call, checks the number against a database and calls app_transfer. That is it. This is what I see when the command fails: asterisk -r Asterisk SVN-branch-13-r431555M, Copyright (C) 1999 - 2014, Digium, Inc. and others. Created by Mark Spencer <markster at digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details...

...p.so] => (ENUM Lookup) == Registered application 'EnumLookup' [app_voicemail2.so] => (Comedian Mail (Voicemail System)) == Registered application 'VoiceMail2' == Registered application 'VoiceMailMain2' == Parsing '/etc/asterisk/voicemail.conf': Found [app_transfer.so] => (Transfer) == Registered application 'Transfer' [app_setcidnum.so] => (Set CallerID Number) == Registered application 'SetCIDNum' [codec_ilbc.so] => (iLBC/PCM16 (signed linear) Codec Translator) == Registered translator 'ilbctolin' from format ILBC to...

Hello List! We're having trouble making call deflection on ISDN PRI. We would like to transfer a call to an external extension but keeping the callerid of the caller so it can be presented to the receiver of the transferred call. At the time we're using Zaptel 1.4.5.1, Asterisk 1.4.11 and Digium hardware TE420B. We've ordered the service (CD) from the phone company. The

...text.so load => app_setcallerid.so load => app_setcdruserfield.so load => app_setcidname.so load => app_setcidnum.so load => app_sms.so load => app_softhangup.so load => app_striplsd.so load => app_substring.so load => app_system.so load => app_talkdetect.so load => app_transfer.so load => app_txtcidname.so load => app_url.so load => app_userevent.so load => app_voicemail.so load => app_waitforring.so load => app_zapateller.so ; [global] chan_phone.so=yes ---------------------- phone.conf: [interfaces] mode=fxo format=slinear echocancel=medium ;silencesup...

...rt 1 app_macro.so Extension Macros 0 app_random.so Random goto 19 codec_ulaw.so Mu-law Coder/Decoder 0 codec_alaw.so A-law Coder/Decoder 0 app_transfer.so Transfer 0 app_verbose.so Send verbose output 0 app_setcdruserfield.so CDR user field apps 0 app_enumlookup.so ENUM Lookup 0 pbx_config.so Text Exten...

...under MariaDB (Reported by Richard Miller) * ASTERISK-24752 - Crash in bridge_manager_service_req when bridge is destroyed by ARI during shutdown (Reported by Richard Mudgett) * ASTERISK-24741 - dtls_handler causes Asterisk to crash (Reported by Zane Conkle) * ASTERISK-24015 - app_transfer fails with PJSIP channels (Reported by Private Name) * ASTERISK-24727 - PJSIP: Crash experienced during multi-Asterisk transfer scenario. (Reported by Mark Michelson) * ASTERISK-24771 - ${CHANNEL(pjsip)} - segfault (Reported by Niklas Larsson) * ASTERISK-24716 - Improve pjsip l...

...under MariaDB (Reported by Richard Miller) * ASTERISK-24752 - Crash in bridge_manager_service_req when bridge is destroyed by ARI during shutdown (Reported by Richard Mudgett) * ASTERISK-24741 - dtls_handler causes Asterisk to crash (Reported by Zane Conkle) * ASTERISK-24015 - app_transfer fails with PJSIP channels (Reported by Private Name) * ASTERISK-24727 - PJSIP: Crash experienced during multi-Asterisk transfer scenario. (Reported by Mark Michelson) * ASTERISK-24771 - ${CHANNEL(pjsip)} - segfault (Reported by Niklas Larsson) * ASTERISK-24716 - Improve pjsip l...

.../usr/lib/asterisk/modules/app_chanisavail.so...done. Loaded symbols for /usr/lib/asterisk/modules/app_chanisavail.so Reading symbols from /usr/lib/asterisk/modules/app_enumlookup.so...done. Loaded symbols for /usr/lib/asterisk/modules/app_enumlookup.so Reading symbols from /usr/lib/asterisk/modules/app_transfer.so...done. Loaded symbols for /usr/lib/asterisk/modules/app_transfer.so Reading symbols from /usr/lib/asterisk/modules/app_setcidnum.so...done. Loaded symbols for /usr/lib/asterisk/modules/app_setcidnum.so Reading symbols from /usr/lib/asterisk/modules/app_cdr.so...done. Loaded symbols for /usr/lib...

...0 app_sayunixtime.so Say time 0 app_hasnewvoicemail.so Indicator for whether a voice mailbox ha 0 app_cdr.so Make sure asterisk doesn't save CDR for 0 app_setcidnum.so Set CallerID Number 0 app_transfer.so Transfer 0 app_enumlookup.so ENUM Lookup 0 app_chanisavail.so Check if channel is available 0 app_db.so Database access functions for Asterisk e 0 app_privacy.so Requir...

...symbols from /opt/asterisk/lib/modules/app_talkdetect.so...done. Loaded symbols for /opt/asterisk/lib/modules/app_talkdetect.so Reading symbols from /opt/asterisk/lib/modules/app_test.so...done. Loaded symbols for /opt/asterisk/lib/modules/app_test.so Reading symbols from /opt/asterisk/lib/modules/app_transfer.so...done. Loaded symbols for /opt/asterisk/lib/modules/app_transfer.so Reading symbols from /opt/asterisk/lib/modules/app_url.so...done. Loaded symbols for /opt/asterisk/lib/modules/app_url.so Reading symbols from /opt/asterisk/lib/modules/app_userevent.so...done. Loaded symbols for /opt/asterisk/...

...sterisk/lib/modules/app_talkdetect.so...done. > Loaded symbols for /opt/asterisk/lib/modules/app_talkdetect.so > Reading symbols from /opt/asterisk/lib/modules/app_test.so...done. > Loaded symbols for /opt/asterisk/lib/modules/app_test.so > Reading symbols from /opt/asterisk/lib/modules/app_transfer.so...done. > Loaded symbols for /opt/asterisk/lib/modules/app_transfer.so > Reading symbols from /opt/asterisk/lib/modules/app_url.so...done. > Loaded symbols for /opt/asterisk/lib/modules/app_url.so > Reading symbols from /opt/asterisk/lib/modules/app_userevent.so...done. > Loaded s...

...Registered application 'System' [app_talkdetect.so] => (Playback with Talk Detection) == Registered application 'BackgroundDetect' [app_test.so] => (Interface Test Application) == Registered application 'TestClient' == Registered application 'TestServer' [app_transfer.so] => (Transfer) == Registered application 'Transfer' [app_txtcidname.so] => (TXTCIDName) == Registered application 'TXTCIDName' == Parsing '/etc/asterisk/enum.conf': Found [app_url.so] => (Send URL Applications) == Registered application 'SendURL' [...

...pp_sms.so ; Requires N/A ;load => app_softhangup.so ; Requires N/A ;load => app_striplsd.so ; Requires N/A ;load => app_substring.so ; Requires N/A ;load => app_system.so ; Requires N/A load => app_talkdetect.so ; Requires N/A ;load => app_test.so ; Requires N/A ;load => app_transfer.so ; Requires N/A ;load => app_txtcidname.so ; Requires N/A ;load => app_url.so ; Requires N/A ;load => app_userevent.so ; Requires N/A load => app_verbose.so ; Requires N/A load => app_voicemail.so ; Requires res_adsi.so ;load => app_waitforring.so ; Requires N/A ;load =&g...

...Park' == Manager registered action ParkedCalls [app_milliwatt.so] => (Digital Milliwatt (mu-law) Test Application) == Registered application 'Milliwatt' [app_cdr.so] => (Make sure asterisk doesn't save CDR for a certain call) == Registered application 'NoCDR' [app_transfer.so] => (Transfer) == Registered application 'Transfer' [app_waitforring.so] => (Waits until first ring after time) == Registered application 'WaitForRing' [format_g726.so] => (Raw G.726 (16/24/32/40kbps) data) == Registered file format g726-40, extension(s) g726-40...