similar to: Busy message

Greetings, This is probably some configuration issue, but for some reason my system has stopped playing a ringing sound when an extension is dialed. The phone rings but there is no ring sound in the ear piece. Gene Kochanowsky

Hi guys, What is different and the "context" to play unavail message and busy message? When a SIP connection is unregistered, voicemail will play unavail message, right? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040913/1a2d1c81/attachment.htm

Does anyone know of a device that will take an SS7 link and convert it to a PRI? -- respectfully, Joseph - (606) 477-2355 x140 ------=============

Using the following defnitions with a C7960: exten => 3001,1,Dial(SIP/3001,15,r) exten => 3001,2,Voicemail2(u3001) exten => 3001,102,Voicemail2(b3001) exten => 3001,103,Hangup If someone is on this phone (real conversation) and another call comes in, the second call goes through the 15 second timeout and is dropped into the 2-priority as "unavailable" (not the 102 busy as

I think I have it configured properly. ztcfg -vv shows it as channel 1 and zttool shows it as OK. But I can't dial out. When I try, it shows it arrive in teh right stack, but then issues the following errors: channel.c:1676 ast_request: No channel type registered for '{PSTN-1}' app_dial.c:554 dial_exec: Unable to create channel of type '{PSTN-1}' = = Everyone is busy at

Assume you have the messages button on your Cisco phone set to dial 3009. Here's an sample dialplan entry that will make the "DND" and "ToVM" and "Messages" button work as expected. This should work for both -stable and -head. exten => 3009,1,GoToIf($[X${RDNIS} != X]3009,4) exten => 3009,2,VoicemailMain() exten => 3009,3,Hangup exten =>

I saw several threads requesting an Asterisk forum to complement the email list. i.e. http://lists.digium.com/pipermail/asterisk-dev/2004-February/003103.html I recently created an Asterisk forum within TMC's popular VoIP forums for everyone to use. http://voip-forum.tmcnet.com/voip-forum/forum/forum_topics.asp?FID=15

I've managed to use GhoustScript (gs) to take a postscript file and convert it to tiffg3, but I CANNOT seem to make it merge multiple files. Here is the output from tiffinfo on the file that SG generates: fteTYGeh2v.tif: TIFF Directory at offset 0x8 Subfile Type: multi-page document (2 = 0x2) Image Width: 1728 Image Length: 1056 Resolution: 204, 96 pixels/inch Bits/Sample: 1

I'll do that, but I'm sure that it's the Polycom :) Caller ID shows up fine on my 7960... W/ the name on top and the number below that. -- Executing NoOp("SIP/614-3ede", "Caller*ID is Matthew Marlowe <6092521155>") in new stack When the phone rings, only 'Matthew Marlowe' would display. When I answer, both the Name & Number will show.

I'm getting the following when sending to a specific fax machine. Any ideas? File name is '/var/spool/asterisk/email2fax/7F2SOeYJiU.tif' Changed from phase 0 to 2 Slow carrier up Slow carrier down Slow carrier up <<< NSF: 20 00 00 11 80 00 8a 49 10 53 54 49 52 4c 49 4e 47 20 43 4f 56 49 4e 47 54 00 67 00 80 80 80 0c 01 02 NSF without final frame tag The remote is made by

Hi. I have installed a x100p (THE x100p for those who have seen my former post). Now I just want to connect a "normal" phone (not an IP phone) to the card and use it as a sip extension (I have a FWD account)... more clearly: I want to be able to pick up the phone and call any FWD user using my FWD account... receive the FWD calls in that phone, and also to be able to make normal

I replied to a post of mine a few days ago asking of anyone uses mailboxexists(). I haven't received any replies. Perhaps few use it or perhaps the reply was overlooked. I thought I'd post the question one last time before giving up on it for now... Thanks! -Michael

Hi everyone, Maybe someone could help me. I have Asterisk in production with TE410P connected to PSTN. When I call from internal phones, either voip or connected via other PRI trunk, to PSTN and if the called phone is busy I don't hear anything!?! I should hear tone indicating that called number is busy. I have played with busydetect and callprogress in zapata.conf, but I didn't find

Hi, Currently using voiptalk.org and the quality is getting really bad. I would like a second provider preferably in UK, anyone got any suggestions? Ta. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040421/3d91c7f6/attachment.htm

Hi, I suspected that I the analogue phone should have got a pass through signal when the power was off to the server, unfortunately it doesn't. I kept asking digium support about that but they didn't give me an answer. The problem is how do I identify whether the X100P is incompatibel with the network or faulty without possibly wasting another USD100??? Aaron On Sat, 2004-05-15, Eric

Hi Last week I installed Asterisk (release1) with digium t100p single span T1 (wct1xxp) board on Dell GX270 pc configured for PRI. Asterisk/t100p is currently the only user of the t1 line. All worked well for about a half a day, PSTN to SIP phones to non-SIP IP phones etc. Alas, since then I consistently get multitudes of blue alarms on all b-channels followed by a loss of d-channel: Aug 31

I have a x100p card that has call waiting on the line comming into it and then into *..... is there any way i can use call waiting on that line? Michael

It appears Asterisk can handle DTMF inband on only a limited selection of formats, of which G.729 is not one. The issue appears to be something involving "short data" -- whatever that is. (I'm inferring all this from looking at dsp.c in the vicinity of the error message I was getting, which pointed to line 1424.) What *is* "short data"? Is this really a show-stopper for

Hi all, I'm writing my graduation theses : analysis VO-IP protocols , and I cannot find any documents about Cisko Skinny Client Control Protocol. I have Cisco CallManager and some IP-phone and I'm sniffing traffic between that, but I don't understand, how this protocol works. Clearly i'm looking for description of SCCP commands and explanation some basic SCCP scenarios or what

Hi there, I just tried today's CVS: 4/23/2004 version and found a strange loise with music on hold. Basically, when on hold you hear very distorted music as if it was very loud. This is the exact same problem described last year at: http://lists.digium.com/pipermail/asterisk-users/2003-April/009735.html http://lists.digium.com/pipermail/asterisk-users/2003-May/011688.html No answers on