search for: getconfno

...put in conference. But here's what happens when I dial 2663: -- Starting simple switch on 'DAHDI/1-1' -- Executing [2663 at internal:1] MeetMe("DAHDI/1-1", ",D") in new stack [2009-05-15 13:21:19] WARNING[2061]: file.c:641 ast_openstream_full: File conf-getconfno does not exist in any format [2009-05-15 13:21:19] WARNING[2061]: file.c:924 ast_streamfile: Unable to open conf-getconfno (format 0x4 (ulaw)): No such file or directory == Spawn extension (internal, 2663, 1) exited non-zero on 'DAHDI/1-1' -- Hungup 'DAHDI/1-1' conf-getco...

Is this me or what? -- Playing 'demo-congrats' -- Executing MeetMe("H323:996", "") in new stack -- Playing 'conf-getconfno' == Parsing '/etc/asterisk/meetme.conf': Found WARNING[17425]: File app_meetme.c, Line 151 (build_conf): Unable to open pseudo channel -- Playing 'conf-invalid' -- Playing 'conf-getconfno' -- Playing 'conf-getconfno' Thanks -- Jordan In a worl...

>-----Original Message----- >From: Larry Shields [mailto:LJ.Shields@Verizon.net] >Sent: Friday, August 27, 2004 12:20 PM >To: asterisk-users@lists.digium.com >Subject: [Asterisk-Users] No audio on PRI channel answered by Playback() orMeetMe() >If I assign the DID to ring extension SIP/2000 and then after time-out send >it to MeetMe() or Playback() it works and the caller

...hile the calling station is connected via PRI: -- Accepting call from '2502' to '2688' on channel 0/4, span 1 -- Executing Wait("Zap/4-1", "3") in new stack -- Executing MeetMe("Zap/4-1", "|Mps") in new stack -- Playing 'conf-getconfno' (language 'en') -- Playing 'conf-getconfno' (language 'en') -- Playing 'conf-getconfno' (language 'en') -- Executing Hangup("Zap/4-1", "") in new stack == Spawn extension (nec_pri, 2688, 3) exited non-zero on 'Zap/4-1...

...info.org/wiki-Asterisk+timer+ztdummy I get the following errors when calling the meetme number. Executing Wait("SIP/216.53.118.2-f41196e0", "1") in new stack -- Executing MeetMe("SIP/216.53.118.2-f41196e0", "|sicp") in new stack -- Playing 'conf-getconfno' (language 'en') == Parsing '/etc/asterisk/meetme.conf': Found Sep 19 13:51:22 WARNING[14066]: chan_zap.c:757 zt_open: Unable to open '/dev/zap/pseudo': No such file or directory Sep 19 13:51:22 ERROR[14066]: chan_zap.c:6687 chandup: Unable to dup channel: No such file...

...-- Executing Goto("SIP/5060-081e9db0", "pbx9|10|1") -- Goto (pbx9,10,1) -- Executing Answer("SIP/5060-081e9db0", "") -- Executing meetme("SIP/5060-081e9db0", "|iMs") -- <SIP/5060-081e9db0> Playing 'conf-getconfno' (language 'es') But can't get sound. If I quit ztdummy module meetme don't work, but I can get sound. Computer as Dell server. Any idea ? very thanks

...How does one handle this? When i'm in a conference, and I press the pound key, it does exit out nicely and continue down the dialplan, i have some NoOp on the next priority after the MeetMe command and it flows nicely from there. -- Accepting a maximum of 4 digits. -- Playing 'conf-getconfno' (language 'en') -- User entered '1234' -- Executing NoOp("SIP/2001-e4e7", "user entered conf no: 1234") -- Executing MeetMe("SIP/2001-e4e7", "1234|p") -- Playing 'conf-locked' (language 'en') == Spawn ext...

...2180 0 piix 8580 0 [permanent] ehci_hcd 24456 0 uhci_hcd 26256 0 rtc 10164 1 ztdummy usbcore 84740 4 ohci_hcd,ehci_hcd,uhci_hcd However, when I enter a meetme conference, I get this: -- Playing 'conf-getconfno' (language 'en') Mar 3 15:27:26 WARNING[23657]: channel.c:2535 ast_request: No channel type registered for 'zap' Mar 3 15:27:26 WARNING[23657]: app_meetme.c:461 build_conf: Unable to open pseudo channel - trying device -- Created MeetMe conference 1023 for conference '...

...conference: [Feb 5 17:46:13] WARNING[10725]: app_meetme.c:772 build_conf: Unable to open pseudo device [Feb 5 17:46:13] -- <SIP/sip.rd.touk.pl-b0006fc0> Playing 'conf-invalid' (language 'en') [Feb 5 17:46:17] -- <SIP/sip.rd.touk.pl-b0006fc0> Playing 'conf-getconfno' (language 'en') [Feb 5 17:46:26] WARNING[10725]: app_meetme.c:772 build_conf: Unable to open pseudo device [Feb 5 17:46:26] -- <SIP/sip.rd.touk.pl-b0006fc0> Playing 'conf-invalid' (language 'en') [Feb 5 17:46:29] -- <SIP/sip.rd.touk.pl-b0006fc0> P...

...extension. But look like following doesn't work. its just hangup if i type wrong number. how to fix this code.. ;Conference rooms/lines: exten => 7580,1,Goto(ivr-meetme,s,1) [ivr-meetme] include => meetme exten => s,1,Answer() exten => s,n,Wait(1) exten => s,n,Background(conf-getconfno) exten => s,n,WaitExten(20) exten => s,n,Hangup() exten => i,n,Playback(pbx-invalid) [meetme] exten => _89XX,1,MeetMe(${EXTEN},cMp) -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/atta...

...=> 1000,1,Answer exten => 1000,n,Meetme(|||d) Asterisk is complaing with: -- Executing Answer("IAX2/xxx.yyy.142.204:4569-2", "") in new stack -- Executing MeetMe("IAX2/xxx.yyy.142.204:4569-2", "|||d") in new stack -- Playing 'conf-getconfno' (language 'en') Warning, flexible rate not heavily tested! Oct 24 16:16:59 WARNING[1732]: channel.c:2597 ast_request: No channel type registered for 'zap' Oct 24 16:16:59 WARNING[1732]: app_meetme.c:465 build_conf: Unable to open pseudo channel - trying device -- Created Me...

Hi, In our office, we're slowly migrating from a cisco call manager set up to asterisk. Problem is management doesn't want to buy any other hardware ?as they had already invested a lot in cisco. The main cause of this is asterisk's added features like unique FAX number for everyone in the company (which will be the same as phone DID), Voice mail, Auto Answer etc yet we need thousands

...tell me where I went so wrong. Func_odbc.conf looks like this: [PIN] dsn=BRIDGE mode=multirow readsql=SELECT pin from users WHERE confid='${SQL_ESC(${CONF_ID})}' extensions.conf section: [infromhost] ;Host dials 8888 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...

I posted a problem earlier thinking it was due to a lack of sound card. Several members stated that you do not need a sound card to play audio to a PRI channel. I did some further testing and discovered that there is a problem with call progress tones or signaling on my PRI. I think that the reason I am not hearing audio from the MeetMe() or Playback() apps. is because the the calling side of

...ps. Just downloaded the latest stable 1.6.1.2. The app_realtime, which was perfectly brilliant and did exactly what I needed, is gone; replaced with func_realtime. The REALTIME function is unacceptable: ; Get the conference number from the user exten => s,n(readconfno),Read(USER_CONFNO,conf-getconfno,0,3,20) ; See if that conference exists in database exten => s,n,Set(CONFINFO="${REALTIME(meetme,confno,${USER_CONFNO})}") ; If conference doesn't exist, play error exten => s,n,GotoIf(${ISNULL(${CONFINFO})} ? 101) ; Retardedly string-manip out the conference setti...

I raised an issue with Sipgate because my Asterisk 1.6 plain refuses to recognize digits pressed on a keypad coming in from a Sipgate trunk. There answer was to set this: dtmfmode=rfc2833 in the general section of sip.conf This has made no difference. I've tried a range of settings (auto, rfc2833,info) but no matter what, it plain refuses to pick up key presses. Locally, if I call from an

Hi, I'm experimenting with the Meetme feature of Asterisk 1.2, exten => 2095,1,MeetMe(|Ds) This almost gives me what I want, where each employee can create their own on-the-fly conferences with a personal Conference Number and PIN. However, as the PIN is actually set by the first callee, then its subject to problems (first callee might enter the wrong PIN, and then no-one else can

...t;, "conf-hasentered") in new stack -- Playing 'conf-hasentered' (language 'en') -- Executing Wait("SIP/9001-08f8d7e0", "2") in new stack -- Executing CBMySQL("SIP/9001-08f8d7e0", "") in new stack -- Playing 'conf-getconfno' (language 'en') Oct 25 18:15:47 NOTICE[12281]: app_cbmysql.c:373 cb_exec: getConf: 1 -- Playing 'agent-pass' (language 'en') Oct 25 18:15:55 NOTICE[12281]: app_cbmysql.c:126 passQuery: Admin flags: Oct 25 18:15:55 NOTICE[12281]: app_cbmysql.c:127 passQuery: user fla...

We are developing an on-demand teleconferencing solution. We will be billing per-minute/per-user. I've successfully gotten Asterisk to write CDR data to a postgres database, but with the way I've got things setup right now the CDR does not have the dialed conference number. We need this information in order to be able to bill. As teleconferencing is the only application of the

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