search for: ypone

All, I upgraded to the [*] stable release branch. When I call into the box (confirmed on 2 installations) the caller no longer hears the ringing. The CLI confirms that extensions are being 'rung'. Whassup? Willy Willy Wouters ypOne Publishing

...=> s,1,DBPut(inuse/chan${ARG1}=1) exten => s,2,Dial(SIP/att-${ARG1}) however, I do not seem to be able to catch the event wich releases the channel in order to reset the DB variable. exten => h,1,DCPut(inuse/chan${ARG1}=0) ; this never gets executed Any ideas? Cheers, WW Willy Wouters ypOne Publishing

...using analog FXO, or VoicePulse or NuPhone. When I dial that same number trough the T1 / PRI interface however, I continually hear ringing, and then the call gets hungup. Any ideas anyone? It kinda annoys our users, since they like to *know* when they dial an invalid number. TIA, WW Willy Wouters ypOne Publishing

...d-calls] exten => 6111,1,Dial(SIP/3200) exten => 6111,2,Hangup ======================= When I dial extension 6111 at the a1 pbx, I see the Dial(IAX2/iax-a2/6111) beinf executed, but nothing happens at machine a2. Any comments? I must be missing something really simple. TIA WW Willy Wouters ypOne Publishing

...he data etc. Now HIS eyes glaze over. That's technical talk ... He's just there to follow up on the appointment and 'qualify' the customer to see if we are worthy of their cheap service. After I looked at their website, I can hear 'quack quack'. Cheers, WW Willy Wouters ypOne Publishing

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi. I'm getting the following error at random intervals on my TE410P with Asterisk CVS-03/30/04-11:49:01-CEST. I have two spans active, one connected to my Telco, the other to a Siemens PABX. Both spans display this behavior at random intervals. All calls are dropped when this happens. Spans are not necessarily in use when this happens.

...It says: your message has been saved. So HOW do I listen to my recording, make sure it sounds OK, and then CONFIRM that is what I want the mailsystem to use OR CANCEL out and leave things the way they are? The same goes, of course, for the unavailable and busy greetings TIA Willy Willy Wouters ypOne Publishing

...e generating the loop (fxs role) then there is no Telco to slave off, and the BOX should take its timing from the CARD hence select '0'. If this reasoning is correct, then when would one reasonably use '2' and even '3' etc. Sorry to be a pest about this. WW Willy Wouters ypOne Publishing

...t; _5.,1,Dial(IAX2/iax-a2:secret@a1.mystrx.com /${EXTEN}@inbound-calls So when you will dial anything beginning with 5, the call will be dialed in the context inbound-calls of a1.mystrx.com -----asterisk-users-admin@lists.digium.com a ?crit : ----- Pour: asterisk-users@lists.digium.com De: willy@yponeinc.com Envoy? par: asterisk-users-admin@lists.digium.com Date: 29-04-2004 22:59 Objet: [Asterisk-Users] IAX Example Needed Hi All! I have two [*]s, and both work OK as a simple local PBX. Now, I try to link them using IAX. Let's call those babies a1 and a2. From a1, I want to dial a phone conn...

I have one SIP extension that can't logon to voicemail. The log file says -- Incorrect password '3213' for user '4035' (context=other) even though the context in voicemail.cnf says 4035 => 3213,Bill Smith Thanks! Paul Mahler mail:pmahler@signate.com phone: 650.207.9855 fax: 877.408.0105 -------------- next part -------------- An HTML attachment was

Hi All I have a snom 200 phone here which works perfectly when using the handset to playback the voicemail messages etc. However when I play back the voice using the speakerphone it sounds choppy. Anyone had this problem before? Regards Mark

...ouse) serious call/voice deterioration due to channel verload. How does one stop this? I.e. it would be very desirable to specify channel capacity (say xx number of simultaneous calls allowed) and force an 'unavailable' if more is demanded. Any thoughts on this? Cheers, WW Willy Wouters ypOne Publishing

I have the voicemail setup working in that I get the MWI and it emails the message correctly. When I pressed the MWI button on my SNOM 200, it dials into the voicemail system and prompts me for a mailbox and password. I know there is a way to automatically connect directly into the mailbox via the extension.conf file, but I can not find the documentation I am looking for in reference to variables

I've been to the WIKI and I've searched the archives. Is anyone on the list successfully using iconnecthere behind NAT? I was, for over a year, and then I changed my "plan" with them. Now all my calls get intercepted immediately, "We're sorry, but your account is temporarily unavailable." Incoming calls work just fine. I contacted their so-called

Hello All, I am getting a set of errors when I boot Asterisk that I have not been able to solve. What is causing these error(s)? Asterisk boot output: ============== Asterisk CVS-04/10/04-21:44:51, Copyright (C) 1999-2001 Linux Support Services, Inc. Written by Mark Spencer <markster@linux-support.net> ========================================================================= [

When people call into my * box over the T1 interface, they get no ring tone. It rings the SIP phone and when the SIP user picks up, both parties can hear each other ok, its just the PSTN user calling in hears no ring. What could be causing this? I tried setting immediate to yes in zapata.conf, but that causes my DNIS and CallerID to stop being available. T100P with E & M Wink start

My PRI is being reset at least once a day with the following errors in the logs. zaptel, zapata, libpri, and asterisk are from CVS this morning.. This has been happening for weeks on all versions (including -stable). the T100P card appears to NOT be sharing an IRQ. xenon# cat /proc/interupts CPU0 0: 1203977 XT-PIC timer 1: 3 XT-PIC keyboard 2:

Firstly, let me just say I am new to asterisk and if anything I've said is covered in an FAQ or in previous posts I apologise but I have tried searching and I've attempted a few of the things I found but they didn't help. Has anybody got any experience using an X100P on an NTL phone line in the UK (I'm in an ex Cable & Wireless area if that makes any difference). The

Greetings, Below is part of the contents of my extensions.conf file. exten => s,1,Wait,1 ; Wait a second before answering. exten => s,2,Answer exten => s,3,ResponseTimeout,10 ; Set the amount of time the user ; has to make a selection. exten => s,4,DigitTimeout,5