similar to: Extension ringing but no ringing sound.

When I dial a SIP phone which is specified in the sip.conf, but the phone is not connected, Asterisk gives the message "The user at Extension XXX is on the phone ...." Shouldn't the message be the unavailable message? Is there something wrong with my set up or is this a "bug" with Asterisk? Simon Brown

I am routing outgoing calls through a sip gateway. The calls go through no problem, however the ringing in the callers ear begins as soon as the last digit is dialed. This has two nasty side effects. First, the caller hears 1-2 more rings than the callee. Second, and more importantly, if the callee's line is busy, the caller continues to get hear ringing, even though the gateway has

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.

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

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

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

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 =>

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'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

hello all, I am having a trouble with Audio using h.323 channel... I am doing this Call comes into cisco 5300 and is sent to Asterisk, asterisk catch call with h.323 driver and send call to a SoftSwitch that routes the call, I can see log debug telling me, CALLED XXX, and then RINGING, and I can hear ring tones... but when call is answered, I DONT HEAR ANYTHING... I am using lastest ASTERISK

I have this line in my extensions.conf, exten => _393.,1,Dial(ZAP/${EXTEN:3},20,tr) when I make a zap call, it gives me two rings and then makes the zap call. Is there is a way I can make the call immediate? Kannaiyan

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

Is the CDR table the right table for billing? I did some tests and CDR records billing seconds for calls that where never picked up. Is this a bug in my system or is that the way CDR works? I called out on my X100T card. Best regards, Han Test data Duration 12 seconds 8 seconds billing time (never picked up my phone) Duration 111 seconds 108 seconds billing time (5 second but

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

Hi, Is anyone else having trouble with their numbers associated with Nufone.net? Our IAX registration with them started failing about two days ago and they have been unresponsive to both emails and phone calls. If one rings the number, one gets a "number is unreachable, please try again" recording... Michael Swan Neon Software, Inc.

When using a PRI, after the remote party hangs up, asterisk tries to spawn a call to the "h" extension. Is this normal behavior for a pri to try to call the "h" extension to try to clean things up? Call Comes In: -- Executing Dial("Zap/1-1", "SIP/16464436000@AST-237.65") in new stack -- Called 16464436000@AST-237.65 -- Accepting call from

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

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