search for: getpin

...888 over SIP trunk exten=8888,1,Answer exten=8888,n,Background(conf-getconfno) exten=8888,n,WaitExten(10) exten=8888,n,Hangup exten=_XXXXXX,1,Set(GLOBAL(CONF_ID)=${EXTEN}) exten=_XXXXXX,n,GoTo(rooms,${EXTEN},1) ; [rooms] exten=_XXXXXX,1,Set(CONF_ID=${EXTEN}) exten=_XXXXXX,n,Background(conf-getpin) exten=_XXXXXX,n,WaitExten(5) exten=_XXXXXX,n,Hangup exten=_1XXXXX,1,Goto(getpin,${EXTEN},1) exten=_2XXXXX,1,Goto(getpin,${EXTEN},1) exten=_3XXXXX,1,Goto(getpin,${EXTEN},1) exten=_4XXXXX,1,Goto(getpin,${EXTEN},1) exten=_5XXXXX,1,Goto(getpin,${EXTEN},1) exten=_6XXXXX,1,Goto(getpin,${EXTEN},1...

Hi there, I have some problems when using READ() statement in the dialplan to collect DTMF digits. I'm using the following within my extensions.conf to receive 6 digits exten => 9070,n,Read(CONFNO,conf-getpin,6) So far it works! The user getting the announcement and asterisk waits for 6 digits. User entered "102030", but asterisk gets "102033"!!! Executing [9070 at incoming:10] Read("SIP/1.1.1.1-08a3dfd0", "CONFNO|conf-getpin|6") in new stack Accepting a...

...t("SIP/9002-08f9feb8", "2") in new stack -- Executing MeetMe("SIP/9002-08f9feb8", "12345|p") in new stack == Parsing '/etc/asterisk/meetme.conf': Found -- Created MeetMe conference 1023 for conference '12345' -- Playing 'conf-getpin' (language 'en') -- Playing 'conf-onlyperson' (language 'en') -- Executing Playback("SIP/9001-08fb34d0", "conf-hasentered") in new stack -- Playing 'conf-hasentered' (language 'en') -- Executing Wait("SIP/9001-08f...

...t("SIP/9002-08f9feb8", "2") in new stack -- Executing MeetMe("SIP/9002-08f9feb8", "12345|p") in new stack == Parsing '/etc/asterisk/meetme.conf': Found -- Created MeetMe conference 1023 for conference '12345' -- Playing 'conf-getpin' (language 'en') -- Playing 'conf-onlyperson' (language 'en') -- Executing Playback("SIP/9001-08fb34d0", "conf-hasentered") in new stack -- Playing 'conf-hasentered' (language 'en') -- Executing Wait("SIP/9001-08f...

...hen listen to MoH. So-far so good. Now the 2nd person dials in. They enter the pin-code, and at that point, the MoH stops. Log shows this: -- Starting simple switch on 'Zap/1-1' -- Executing MeetMe("Zap/1-1", "400|iM") in new stack -- Playing 'conf-getpin' (language 'en') -- Recording -- Playing 'vm-rec-name' (language 'en') -- Stopped music on hold on SIP/101-0816b2a8 -- Playing 'beep' (language 'en') So it looks like it's possibly a deliberate action - but the effect on the 1st...

...@300/3", "wav|/tmp/test|m") in new stack -- Executing MeetMe("IAX2/300@300/3", "37455|APM") in new stack == Parsing '/etc/asterisk/meetme.conf': Found -- Created MeetMe conference 1023 for conference '37455' -- Playing 'conf-getpin' (language 'en') -- Playing 'conf-onlyperson' (language 'en') -- Started music on hold, class 'default', on IAX2/300@300/3 Aug 8 21:20:32 WARNING[19742]: app_meetme.c:962 conf_run: Unable to write frame to channel: Resource temporarily unavailable...

...app_meetme.c:2375 find_conf_realtime: No Zap channel available for conference, user introduction disabled [Mar 31 17:57:39] WARNING[22242]: app_meetme.c:2381 find_conf_realtime: No Zap channel available for conference, conference recording disabled -- <SIP/3601-c80b4520> Playing 'conf-getpin' (language 'en')

Hi all, I had a chance to use some call conferences that had some very neat functionalities: - When you call you are first asked for your name - When someone joins the conference a message "<name> is now joining the conference." is played. - When someone leaves the room a message "<name> has left to conference." is played. How can I set MeetMe/Asterisk to have

Due to changes at the office, I'm finally getting around to setting up an AA to deal with incoming calls. One of the big changes is that we're dropping the old alphanumeric pager and will just send pages to our phones. I've got the outbound greeting message working in a test context no problem right now, but I'm kind of stuck on how to capture a DTMF sequence from a user and

Yes. Tim Thompson http://www.amatechtel.com (806) 722-2227 -----Original Message----- From: CW_ASN - Gus [mailto:cw_asn@fibertel.com.ar] Sent: Wednesday, October 22, 2003 1:12 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Meetme Do you have ztdummy or zaptel device in your system? ----- Original Message ----- From: "Panny Malialis"

...strmbase: Get rid of the "pfnBreakConnect" field from BaseOutputPinFuncTable. setupapi: Handle an empty string for the source file in SetupQueueCopySection(). strmbase: Remove function pointers from the IEnumPinsImpl structure. wineqtdecoder: Fix implementation of QT_GetPin(). strmbase: Determine the pin count using the GetPin() callback. strmbase: Get rid of the "pfnGetPinCount" callback. winegstreamer: Use CONTAINING_RECORD instead of casting. strmbase: Use heap_alloc_zero() in enum_pins_create(). strmbase: Don't store the...

Anthony Scarpino wrote (elsewhere): > While writing up the man page.. I thought of a few things that I was > wondering if you considered.. > > Can an encrypted dataset (keytype=dataset) reside in a non-encrypted (no > kek defined) pool? I can see a case for and against allowing this when considering it purely at the feature level as users/admins see things. The admin can

Anthony Scarpino wrote (elsewhere): > While writing up the man page.. I thought of a few things that I was > wondering if you considered.. > > Can an encrypted dataset (keytype=dataset) reside in a non-encrypted (no > kek defined) pool? I can see a case for and against allowing this when considering it purely at the feature level as users/admins see things. The admin can

My wife's got an appropriate Southern England (Wimbledon) accent and I'm sure she would try her hand. Does anyone have a comprehensive list of the words that need to be said? Matt, do you have them if your wife's done a set for French users? Mark, if you have the kit maybe you could chop up the file? I write a utility to chop up and compress the wave file based on some of the C