search for: predial

Displaying 20 results from an estimated 41 matches for "predial".

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2020 Feb 05
1
Hangup hook to put back a call into a queue
...fun-and-income/57718 but still not able to implement it:-( what I’ve done: * found out in a hard way how to detect the current destination extension (because it’s turn out that CALLERID(dnid) is not working in case of forwarded call it’s show the original destination) * write a macro-dialout-one-predial-hook and a hook marco like this: [macro-dialout-one-predial-hook] exten => s,1,Noop(Entering user defined context macro-dialout-one-predial-hook in extensions_custom.conf) exten => s,n,GotoIf($["${DEXTEN}"=“2001”]?special) exten => s,n,GotoIf($["${DEXTEN}"=“2002”]?spe...
2014 Sep 23
1
Change codec when dial from SIP to DAHDI
...when dial from SIP to DAHDI. I tried to use IP_CODEC/SIP_CODEC_OUTBOUND at dialplan. but the SIP codec change after dahdi answered the channel. so everything is broken. the call log like below: [2014-09-23 21:18:46] VERBOSE[11634][C-0000000d] pbx.c: -- Executing [s at macro-dialout-trunk-predial-hook:2] Set("SIP/222-00000004", "SIP_CODEC=alaw") in new stack [2014-09-23 21:18:46] VERBOSE[11634][C-0000000d] pbx.c: -- Executing [s at macro-dialout-trunk-predial-hook:3] Set("SIP/222-00000004", "SIP_OUT_CODEC=alaw") in new stack [2014-09-23 21:18:46]...
2010 Jun 11
2
Call ended after 31 seconds
...s is the log, but I've not been able to find something wrong... Any ideas? [Jun 11 15:50:46] DEBUG[26071] app_macro.c: Executed application: ExecIf [Jun 11 15:50:46] VERBOSE[26071] logger.c: -- Executing [s at macro-dialout-trunk:16] Macro("SIP/3000-00006d07", "dialout-trunk-predial-hook|") in new stack [Jun 11 15:50:46] VERBOSE[26071] logger.c: -- Executing [s at macro-dialout-trunk-predial-hook:1] MacroExit("SIP/3000-00006d07", "") in new stack [Jun 11 15:50:46] DEBUG[26071] app_macro.c: Executed application: Macro [Jun 11 15:50:46] VERBOSE[26071...
2019 Nov 15
2
pre-dial handler, how to access variables from calling channel?
...> n,Set(PAI=${PJSIP_HEADER(read,P-Asserted-Identity)}) same => n,Set(FROM=${CALLERID(Number)}) same => n,Set(TO=${DESTINATION}) same => n,Set(DIVERSION=${PJSIP_HEADER(read,Diversion)}) same => n,AGI(router.agi) same => n,GoTo(dial-out,s,1) [predial-handler]; Manipulate Header on OUTBOUND channel exten => screen-update,1,NoOp(PREDIAL FROM: ${CALLERID(Number)} TO: ${DESTINATION} PAI: ${PAI}) same => n,Set(PJSIP_HEADER(update,P-Asserted-Identity)=${PAI}) same => n,Return [dial-out] exten => s,1,NoOp(DIAL FROM: ${CALL...
2006 May 09
2
exten statement execution order
...he call is answered. ; Standard extension logic [macro-stdexten] ; ${ARG1}=Extension ${ARG2}=Device(s) to ring exten => s,1,NoOp(stdexten ${EXTEN}) exten => s,n,Set(cname=${CALLERID(number)}@asterisk.deskoptional.com) exten => s,n,Set(CALLERID(number)=${cname}) exten => s,n,Macro(psa-predial) exten => s,n,Dial(${ARG2},20,tTwW) exten => s,n,Set(savestatus=${DIALSTATUS}) exten => s,n,Macro(psa-postdial) exten => s,n,Goto(s-${savestatus},1);ANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER exten => s-ANSWER,1,NoOp(call was answered!!!!!) exten => s-CHANUNAVAIL,1,VoiceMail(u${AR...
2008 Apr 30
2
Sending caller name out PRI?
...n: Set -- Executing [s at macro-dialout-trunk:15] GotoIf("SIP/3991-b7900488", "1?gocall") in new stack -- Goto (macro-dialout-trunk,s,17) Executed application: GotoIf -- Executing [s at macro-dialout-trunk:17] Macro("SIP/3991-b7900488", "dialout-trunk-predial-hook|") in new stack Context 'macro-dialout-trunk-predial-hook' for macro 'dialout-trunk-predial-hook' lacks 's' extension, priority 1 Executed application: Macro -- Executing [s at macro-dialout-trunk:18] GotoIf("SIP/3991-b7900488", "0?bypass|1&q...
2010 Jun 16
0
H323 Trunk Problem calling from Asterisk to Avaya PBX
...ot;0|Set|DIAL_TRUNK_OPTIONS=M(setmusic^)") in new stack > [Jun 16 15:29:27] DEBUG[8721] app_macro.c: Executed application: ExecIf > [Jun 16 15:29:27] VERBOSE[8721] logger.c: -- Executing > [s at macro-dialout-trunk:16] Macro("SIP/16000-00000002", > "dialout-trunk-predial-hook|") in new stack > [Jun 16 15:29:27] VERBOSE[8721] logger.c: -- Executing > [s at macro-dialout-trunk-predial-hook:1] MacroExit("SIP/16000-00000002", "") > in new stack > [Jun 16 15:29:27] DEBUG[8721] app_macro.c: Executed application: Macro > [Jun 16...
2009 Oct 09
0
calls ansowered for 1 second or less
...;, "custom=IAX2/99999") in new stack -- Executing [s at macro-dialout-trunk:15] ExecIf("SIP/100-b609f9c0", "0|Set|DIAL_TRUNK_OPTIONS=M(setmusic^)") in new stack -- Executing [s at macro-dialout-trunk:16] Macro("SIP/100-b609f9c0", "dialout-trunk-predial-hook|") in new stack -- Executing [s at macro-dialout-trunk-predial-hook:1] MacroExit("SIP/100-b609f9c0", "") in new stack -- Executing [s at macro-dialout-trunk:17] GotoIf("SIP/100-b609f9c0", "0?bypass|1") in new stack -- Executing [s at...
2010 Mar 26
1
SIP/2.0 403 Forbidden
...uot;) in new stack > -- Executing [s at macro-dialout-trunk:15] ExecIf("SIP/75002-b7705298", > "1|Set|DIAL_TRUNK_OPTIONS=M(setmusic^none)Tt") in new stack > -- Executing [s at macro-dialout-trunk:16] Macro("SIP/75002-b7705298", > "dialout-trunk-predial-hook|") in new stack > -- Executing [s at macro-dialout-trunk-predial-hook:1] > MacroExit("SIP/75002-b7705298", "") in new stack > -- Executing [s at macro-dialout-trunk:17] GotoIf("SIP/75002-b7705298", > "0?bypass|1") in new stack &g...
2010 Mar 26
1
send a call from A to B use sip trunk prablem
...t; -- Executing [s at macro-dialout-trunk:15] > ExecIf("SIP/192.168.0.151-088e7938", > "1|Set|DIAL_TRUNK_OPTIONS=M(setmusic^none)Tt") in new stack > -- Executing [s at macro-dialout-trunk:16] > Macro("SIP/192.168.0.151-088e7938", "dialout-trunk-predial-hook|") in new > stack > -- Executing [s at macro-dialout-trunk-predial-hook:1] > MacroExit("SIP/192.168.0.151-088e7938", "") in new stack > -- Executing [s at macro-dialout-trunk:17] > GotoIf("SIP/192.168.0.151-088e7938", "0?bypass|1&q...
2010 Nov 03
1
Ring back problem on SIP calls. Order of 183 Session Progress and 180 Ringing
Hello everyone! I've had this problem for a while and cant figure it out. When an outside caller calls an extension on my asterisk system, they do not hear any sort of ringing. Inside extensions calling other extensions do hear ringing. We have 3 other asterisk systems that are configured the same way, but do not have this problem. We think it has something to do with asterisk 1.6. The other
2018 Jun 05
2
How to execute priorities following a caller hangup in a successful Dial?
This has been super-helpful, Eric. However, the handleHangupByPeer priorities below are still not run when the peer hangs-up. The last line in the cli when the peer hangs-up is still: Strict RTP learning complete - Locking on source address (Although sometimes there is also: Retransmission timeout reached on transmission) same =>
2008 Aug 11
1
Intermittent T.38 pass through
...d db levels on the FXS ports, higher and lower, no effect. I increased jitter, reduced jitter, disabled jitter, no effect. Ensured echo can's were off, no effect. Manually set faxes to 14.4bps, ecm off, no effect. Even switched telephone cord, no effect. On these Linksys 2102's, you can predial #99 to force the ATA to enable fax t.38, this works and is reliable, no RTP is setup, just UDPTL. So my question is this: Can I setup Asterisk to only allow t.38 pass through from these ATA's, without the need to use the #99 in every dial string from the fax machine? Thanks. JR -- --------...
2010 May 05
0
T38 trunk configuration for relay appears to affect default trunks for voip
...9] pbx.c: -- Executing [s at macro-dialout-trunk:15] ExecIf("SIP/21-00000058", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^))") in new stack [Apr 30 09:34:31] VERBOSE[12649] pbx.c: -- Executing [s at macro-dialout-trunk:16] Macro("SIP/21-00000058", "dialout-trunk-predial-hook,") in new stack [Apr 30 09:34:31] VERBOSE[12649] pbx.c: -- Executing [s at macro-dialout-trunk-predial-hook:1] MacroExit("SIP/21-00000058", "") in new stack [Apr 30 09:34:31] VERBOSE[12649] pbx.c: -- Executing [s at macro-dialout-trunk:17] GotoIf("SIP/21-0...
2015 Mar 20
3
outbound calls
..._OPTIONS=M(setmusic^default))") in new stack -- Executing [s at macro-dialout-trunk:16] ExecIf("SIP/101-00000103", "0?Set(DIAL_TRUNK_OPTIONS=M(confirm))") in new stack -- Executing [s at macro-dialout-trunk:17] Macro("SIP/101-00000103", "dialout-trunk-predial-hook,") in new stack -- Executing [s at macro-dialout-trunk-predial-hook:1] MacroExit("SIP/101-00000103", "") in new stack -- Executing [s at macro-dialout-trunk:18] GotoIf("SIP/101-00000103", "0?bypass,1") in new stack -- Executing [s at mac...
2004 Jun 14
1
making * more like a normal pbx (cisco ata-186)
...nctionality on sipura (pick-up phone and go right into context) successfully. It's conceivable you could change the sipura's dial-plan to include a singular "9", which takes you into the incoming context, and then you could... [incoming] 9,1,Dialtone 9,2,.... Of course, predialing (my phones let me dial a number on display, and once I pick up the receiver, they're dialed) would now become a problem, as establishing the connection may cause the first DTMF digit after the 9 to be missed by *, and you'd have to add a pause to all speed-dial (or pre-dial) numbers. If...
2010 Jun 21
1
How to tell if a dropped call is my fault
...-b7b9ae38", "0|Set|DIAL_TRUNK_OPTIONS=M(setmusic^)") in new stack [Jun 21 08:53:29] DEBUG[21559] app_macro.c: Executed application: ExecIf [Jun 21 08:53:29] VERBOSE[21559] logger.c: -- Executing [s at macro-dialout-trunk:16] Macro("SIP/611-b7b9ae38", "dialout-trunk-predial-hook|") in new stack [Jun 21 08:53:29] VERBOSE[21559] logger.c: -- Executing [s at macro-dialout-trunk-predial-hook:1] MacroExit("SIP/611-b7b9ae38", "") in new stack [Jun 21 08:53:29] DEBUG[21559] app_macro.c: Executed application: Macro [Jun 21 08:53:29] VERBOSE[21559]...
2011 Sep 28
2
PSTN connectivity
Hi All, I am trying to connect my asterisk box with freepbx to PSTN. I have purchased x100p FXO card and installed in my asterisk server. My freepbx detected the x100p FXO card and i can see the card specific details in command line. I have configured the following things. 1. OUTBOUND caller id and Dialing rules in Freepbx. 2. INBOUND route When i call to the PSTN number before
2015 Mar 27
0
call between snom 300 and aastra 6731i
..._OPTIONS=M(setmusic^default))") in new stack -- Executing [s at macro-dialout-trunk:16] ExecIf("SIP/300-00000192", "0?Set(DIAL_TRUNK_OPTIONS=M(confirm))") in new stack -- Executing [s at macro-dialout-trunk:17] Macro("SIP/300-00000192", "dialout-trunk-predial-hook,") in new stack -- Executing [s at macro-dialout-trunk-predial-hook:1] MacroExit("SIP/300-00000192", "") in new stack -- Executing [s at macro-dialout-trunk:18] GotoIf("SIP/300-00000192", "0?bypass,1") in new stack -- Executing [s at mac...
2015 Mar 20
0
outbound calls
...default))") in new stack > -- Executing [s at macro-dialout-trunk:16] ExecIf("SIP/101-00000103", > "0?Set(DIAL_TRUNK_OPTIONS=M(confirm))") in new stack > -- Executing [s at macro-dialout-trunk:17] Macro("SIP/101-00000103", > "dialout-trunk-predial-hook,") in new stack > -- Executing [s at macro-dialout-trunk-predial-hook:1] > MacroExit("SIP/101-00000103", "") in new stack > -- Executing [s at macro-dialout-trunk:18] GotoIf("SIP/101-00000103", > "0?bypass,1") in new stack >...