search for: transferstatus

...rmation for example) I don't think that this is possible as once my dialplan starts using Dial() there's no way to interrupt it. But: [internal-transfer] exten => _5XXXX,1,SetVar(CALLED=${EXTEN:1:4}) exten => _5XXXX,2,Dial(${CALLED},40,TrM,macro-hookback) exten => _5XXXX,3,DBGet(TRANSFERSTATUS=transferstatus/${EXTEN:2:4}) exten => _5XXXX,4,gotoIf($["${TRANSFERSTATUS}" = "ATTEMPT_RECALL" ]?5:6) exten => _5XXXX,5,Dial(12345) exten => _5XXXX,6,NoOp(Dial back ending) exten => *7XXXX,1,DBPut(transferstatus/${EXTEN:2:4}=ATTEMPT_RECALL) exten => h,1,DBDel(tra...

...destination. If TECH (SIP, IAX2, LOCAL etc) is used, only an incoming call with the same channel technology will be transfered. Note that for SIP, if you transfer before call is setup, a 302 redirect SIP message will be returned to the caller. The result of the application will be reported in the TRANSFERSTATUS channel variable: SUCCESS Transfer succeeded FAILURE Transfer failed UNSUPPORTED Transfer unsupported by channel driver The option string many contain the following character: 'j' -- jump to n+101 priority if the channel transfer attempt fails Thanks!! PB -------------- next part ---------...

...ardware TE420B. We've ordered the service (CD) from the phone company. The zapata.conf file inlcludes: Transfer= yes Facilityenable=yes Callerid=asreceived In extensions.conf we try to transfer a call to an external extension as: Transfer(ZAP/g0/ xxxxxxxx) but that fails with the ${TRANSFERSTATUS} = UNSUPPORTED. Ideas anyone? We would really appreciate it! Kind regards, Hanna Hanna Wallin System Development Direct: +46 (0)8 736 77 29 Mobile: +46 (0)73 414 13 38 Fax: +46 (0)8 736 77 91 E-mail: hanna.wallin at pocketmobile.se <mailto:hanna.wallin at pocketmobil...

I am having trouble understanding how to use this. I want to transfer certain incoming calls from an IAX ITSP based on caller ID. From what I can make of the docs, I thought I need to do something like this... exten => _NXXNXXXXXX,n(nocid),transfer(1000) exten => _NXXNXXXXXX,n,noop(boo,${TRANSFERSTATUS}) exten => _NXXNXXXXXX,n,hangup exten => 1000,1,Dial(IAX2/jnctn_out/16665551234,45,t) exten => 1000,n,hangup When the call comes in, the console shows that TRANSFERSTATUS contains SUCCESS, but there is no evidence that the lines at extension 1000 ever execute. The caller hears silenc...

...ward) 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 ${CALLERID(rdnis)}) exten => _X.,n,Transfer(${EXT_TRUNK}/${EXTEN}) exten => _X.,n,NoOp(Transfer STATUS: ${TRANSFERSTATUS}) However, this does not work, Is there a way to send the 302 response to the VoIP provider ? Thanks. Hans

...  n,Set(MYnewHEADER=${REPLACE(MYHEADER,A,)})    same =>      n,Set(PJSIP_HEADER(update,To)=${MYnewHEADER})    ; The previous block did not work because the INVITE message is not sent altered    same =>      n,Transfer(PJSIP/sip:B${MYDESTINY}@10.1.1.2)    same =>      n,NoOp(Transferencia=${TRANSFERSTATUS})    same =>      n,Goto(end)    same =>      n(black),Verbose(Fraudulento)    same =>      n,Answer()    same =>      n,Playback(bye)    same =>      n,HangUp()    same =>      n(end),Verbose(Terminado) What I need is to be able to change the TO: header so that the softswitch re...

I all, I'm trying to transfer a iax2 channel trought dialplan app transfer to another extensions (IAX). The variable TRANSFERSTATUS report SUCCESS but the call isn't trasfered. I haven't other information, in console I see only hangup of a channel. My scenario is 3 asterisk box connected with iax trunk, I talk from BOX1 to BOX2 and I want to transfer user on BOX2 to another user on BOX3. After transfer command is execu...

...I send a Transfer cmd (such as the following) Action: AGI ActionID: C8 Channel: SIP/1004-00000002 CommandID: C8 Command: EXEC Transfer SIP/1003 Destination phone starts ringing. If it answers the call, everything works fine. I am notified the agiexec completed successfully and given a TRANSFERSTATUS of SUCCESS. I am also notified when the call is hungup so that I can cleanup information regarding the call. Event: Hangup Privilege: call,all Channel: SIP/1004-00000002 Uniqueid: 1367761382.0 CallerIDNum: 1004 CallerIDName: 1004 - Asterisk ConnectedLineNum: <unknown> ConnectedLin...

Hi, i've some trouble with an * installation when the following scenario happen. 1) Inbound call to SIP/xxxxxxxxxxxx ; 2) Call is redirected to ring group 6xx 3) SIP extension 1xx answer. 4) caller want to speak with john doe on his mobile 5) assistant put caller on hold 6) assistant start a call to john doe mobile using a php script (AMI - Originate with custom context to force outbound

...when not specifying "dbfile" in res_config_sqlite3.conf (Reported by Dennis) * ASTERISK-28478 - Crash performing "core reload" with modified res_config_sqlite3.conf (Reported by Dennis) * ASTERISK-26968 - chan_pjsip: Transfer() does not result in TRANSFERSTATUS reflecting SIP response to transfer (Reported by Dan Cropp) * ASTERISK-28282 - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in chan_sip) (Reported by Walter Doekes) New Features made in this release: ----------------------------------- * ASTERISK-17808 - [patch]...

...when not specifying "dbfile" in res_config_sqlite3.conf (Reported by Dennis) * ASTERISK-28478 - Crash performing "core reload" with modified res_config_sqlite3.conf (Reported by Dennis) * ASTERISK-26968 - chan_pjsip: Transfer() does not result in TRANSFERSTATUS reflecting SIP response to transfer (Reported by Dan Cropp) * ASTERISK-28282 - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in chan_sip) (Reported by Walter Doekes) * ASTERISK-28457 - [patch] Fix crash in chan_dahdi on 32-bit systems caused by ASTERISK-2831...

...when not specifying "dbfile" in res_config_sqlite3.conf (Reported by Dennis) * ASTERISK-28478 - Crash performing "core reload" with modified res_config_sqlite3.conf (Reported by Dennis) * ASTERISK-26968 - chan_pjsip: Transfer() does not result in TRANSFERSTATUS reflecting SIP response to transfer (Reported by Dan Cropp) * ASTERISK-28282 - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in chan_sip) (Reported by Walter Doekes) * ASTERISK-28457 - [patch] Fix crash in chan_dahdi on 32-bit systems caused by ASTERISK-2831...

...ASTERISK-26006 - Show offending IP for TLS setup failures in logs (Reported by Oleksandr Natalenko) * ASTERISK-28444 - chan_pjsip: Peer IP for SSL handshake errors not logged (Reported by Bernhard Schmidt) * ASTERISK-26968 - chan_pjsip: Transfer() does not result in TRANSFERSTATUS reflecting SIP response to transfer (Reported by Dan Cropp) * ASTERISK-28419 - app_amd: Does not work with silence suppression (Reported by Nasir Iqbal) * ASTERISK-28018 - IP Fragmentation happening instead of DTLS fragmentation on handshake server hello certificate...