search for: ringtimer

Hey guys, I've got a part of my dialplan that dials multiple people: exten => direct,n,Dial(${QUEUEEXTS},${RINGTIME}) Multiple extensions are in the ${QUEUEEXTS} from an external script - e.g. SIP/100&SIP/101&SIP/105 etc This works great, however I want to see if I can find a way to work out (and run an AGI script) when the call is picked up by someone. Thanks all!

...r >>reference: >> >>[ext-sales] >>exten => 1,1,Answer() ; >>exten => 1,2,SetCIDName(Sales) ; >>exten => 1,3,GotoIfTime(${REGTIME}|${REGDAYS}|*|*?ext-cs-vm,1,1) >>exten => 1,4,Playback(custom/PleaseWait) >>exten => 1,5,Macro(dial,${RINGTIMER},tm,717-6197585949#) >>exten => 1,6,Macro(dial,${RINGTIMER},tm,707-6199208398#) >>exten => 1,7,Macro(dial,${RINGTIMER},tm,8323686410#) >>exten => 1,8,Macro(dial,${RINGTIMER},tm,717-6197585949#) >>exten => 1,9,Macro(dial,${RINGTIMER},tm,707-6199208398#) >>ex...

...gt; reference: >> >> [ext-sales] >> exten => 1,1,Answer() ; >> exten => 1,2,SetCIDName(Sales) ; >> exten => 1,3,GotoIfTime(${REGTIME}|${REGDAYS}|*|*?ext-cs-vm,1,1) >> exten => 1,4,Playback(custom/PleaseWait) >> exten => 1,5,Macro(dial,${RINGTIMER},tm,717-6197585949#) >> exten => 1,6,Macro(dial,${RINGTIMER},tm,707-6199208398#) >> exten => 1,7,Macro(dial,${RINGTIMER},tm,8323686410#) >> exten => 1,8,Macro(dial,${RINGTIMER},tm,717-6197585949#) >> exten => 1,9,Macro(dial,${RINGTIMER},tm,707-6199208398#) >&g...

Priorityjumping is totally ignored by my asterisk (tested 1.4.4 and 1.4.7.1 on FreeBSD 6.2) [general] priorityjumping=yes With n+101: exten => 1337,1,Dial(SIP/zytek,5,Ttj) exten => 1337,102,Dial(SIP/zytek,${RINGTIME},${OPTIONS}) exten => 1337,n,Hangup -- Executing [1337 at firma:1] Dial("SIP/113-087a3000", "SIP/zytek|5|Ttj") in new stack -- Called zytek

We have now got our new PRI line (10 channels, 100 numbers) connected and everything is working except the outgoing caller ID. Whatever SIP phone I'm using, the CID that's shown is the very first number... ----- s n i p ----- [default] include => outgoing include => priin [outgoing] exten => _NXXXXX.,1,Macro(dial,08${EXTEN},${RINGTIME}) ; Local number (w/o areacode) -

Regarding this I've read the specs linked to in detail, but I can find no mention anywhere of any change that implies or states that no ring time will be recorded anymore in Asterisk 13 and that all times in start and answer columns will now be equal for all calls. Can this be because I nowhere use the Answer() application in my dialplan when dialing out? -----Original Message----- From:

...tensions.conf I see the following marcro's: ; Ring an extension, if the extension is busy or there is no answer send it ; to voicemail ; ARGS: $VMBOX, $EXT [macro-exten-vm] exten => s,1,Setvar(FROMCONTEXT=exten-vm) exten => s,2,Macro(record-enable,${ARG2},IN) exten => s,3,Macro(dial,${RINGTIMER},${DIAL_OPTIONS},${ARG2}) exten => s,4,GotoIf($[${CHANNEL:0:5} = Local]?s-${DIALSTATUS},1) ; if the channel is Local, then do not go to voicemail. This is primarily to avoid vm for call-forwarded extensions in ring groups exten => s,5,GotoIf($[${ARG1} = novm]?s-${DIALSTATUS},1) ; no voicemai...

...all would be well, but this didn't do any good. After playing a bit longer I just set the following: exten => 900,2,set(CALLERID(num)="606-555-1212") exten => 900,3,set(CALLERID(name)="Outside Call") exten => 900,4,Dial(${DIALEXTENSIONS},${RINGTIMER},${DIAL_OPTIONS}) Now all works well. So is there another option somewhere to keep asterisk from killing a transfer without callerid? This happened on both 1.4.17 & 1.4.18.1. Thanks, Ken -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.co...

Hello, i am trying to enable call forwarding on asterisk 1.6 with asterisk-gui If i set my stdexten as follows (with the lines i marked) everything seems like working. But if i make any change on asterisk-gui and apply it.. it recreates the macro-stdexten and deletes my configuration regarding to it. So where should i add my call-forward configuration??? Where am i making a mistake??

Hi All, Asterisk 1.4.12 CentOS 5 My ISP account includes nearly 500 minutes of VOIP calls per month but the service is expensive for unbundled minutes. So I'm trying to find a way to keep an accumulated total of calls made through that trunk so that I can automatically switch to a lower-cost provider when my bundled minutes are used. The plan is to store the accumulated time in AstDB and

Hello all, This is a VICIDial server and I am looking to send calls to VM box 2100 after 3 minutes of sitting in the queue(via the VICIDial AGI). This would be inserted between exten => s,8,Background(open) and exten => s,9,AGI.

I have two sites running Asterisk PBX. Normally the inbound calls go through a 3rd (colocated) server and are routed via IAX to the site (the site registers with the main server) I created a macro that tries to ring one location and then another. Each site explicitly Answer() the call even though it will only ring all the sip phones at the relevant location. When fall back is in effect it goes to

...callerid) exten => s,n,Set(FROMCONTEXT=exten-vm) exten => s,n,Set(VMBOX=${ARG1}) exten => s,n,Set(EXTTOCALL=${ARG2}) exten => s,n,Set(CFUEXT=${DB(CFU/${EXTTOCALL})}) exten => s,n,Set(RT=${IF($[$["${VMBOX}"!="novm"] | $["foo${CFUEXT}"!="foo"]]?${RINGTIMER}:"")}) exten => s,n,Macro(record-enable,${EXTTOCALL},IN) exten => s,n,GotoIf($["${CHANNEL:0:5}" = "Local"]?dolocaldial,1) ; if the channel is Local exten => s,n,Macro(dial,${RT},${DIAL_OPTIONS},${EXTTOCALL}) exten => s,n,GosubIf($[$["${DIALSTATUS}&quot...

I just can't get kphone to register with asterisk, i can make calls to the demos and even get into the mailbox but kphone cannot register. Here's my story. Can you help me?? Please I have installed asterisk on debian using apt-get install asterisk. I have configured an extension in extensions.conf as follows exten => 8003,1,Dial(Sip/8003,${RINGTIME},rt) exten =>

...r() exten => _SIP011XX,n,NoOp(Call Parked on: ${PARKINGSLOT}) exten => _SIP011XX,n,NoOp(This is who parked us: ${EXTEN}) exten => _SIP011XX,n,Set(PARKINGEXTEN=${PARKINGSLOT}) ;This sets the PARKINGEXTEN to the parking slot we were parked in. exten => _SIP011XX,n,Dial(SIP/${EXTEN:4:4},${RINGTIMER},${INTERNAL_DIAL_OPTIONS}) ;This send the call back to the person who parked it. There are a couple of global variables I use here. Nothing unusual here. So what is the problem? Well the problem is that the PARKINGEXTEN variable gets reset after the dial command in parkedcallstimeout. That...

...${RGPREFIX})}}" != "${RGPREFIX}"]?4:3) [pbx_config] 2. Set(VMGAIN=${IF($["foo${VM_GAIN}"!="foo"]?"g(${VM_GAIN})":"")}) [pbx_config] 'docfu' => 1. Set(RTCFU=${IF($["${VMBOX}"!="novm"]?${RINGTIMER}:"")}) [pbx_config] 6. Set(RT=${IF($[$["${VMBOX}"!="novm"] | $["foo${CFUEXT}"!="foo"]]?${RINGTIMER}:"")}) [pbx_config] 10. GosubIf($[$["${DIALSTATUS}"="NOANSWER"] & $["fo...

...60 expires 120 May 26 07:34:52 NOTICE[2761]: pbx.c:1738 pbx_extension_helper: Cannot find extension context 'from-internal' May 26 07:35:23 NOTICE[2761]: pbx.c:1738 pbx_extension_helper: Cannot find extension context 'from-internal' my extension.conf is :- [globals] VM_PREFIX = * RINGTIMER = 15 REGTIME = 7:55-17:05 REGDAYS = mon-fri RECORDEXTEN = "" PARKNOTIFY = SIP/200 OUT_2 = IAX2/fwd OUT_1 = ZAP/g0 OUTPREFIX_2 = OUTMAXCHANS_2 = 1 OUTCID_2 = mithunafila<672648> OPERATOR = NULL = "" IN_OVERRIDE = forcereghours INCOMING = GRP-1 FAX_RX_EMAIL = fax@mydomain.co...

Hi All, Asterisk 1.4.12 on CentOS 5 Sorry for a question that I'm guessing is obvious to most of you. I'm trying to revamp my dialplan. When I first created it, I had something like: exten => s,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})}) exten => s,2,Dial(${rgMain},${RINGTIME},t) exten => s,3,VoiceMail(main at default) exten => s,103,VoiceMail(main at default)

Hello Team, Can you help me on this? I have attached here the queue logs of my asterisk. I've searching a parser for this. I do not know what are the meaning of that logs. Thank you so much. Your response is highly appreciated. Regards, Cristina -------------- next part -------------- An HTML attachment was scrubbed... URL:

My dial plan seems to work great - in that when I call extensions 1234 it connects to 1234. Strangely, after the call terminates (the other side hangs up first), Asterisk continues in the same context and then matches to extensions _. which causes an invalid extension error! Why does asterisk not leave the context (called internalmenu) after the remote hangup? Instead, it continues to the