similar to: cvs log archive

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

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

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

Hi guys, I am new to Asterisk. It looks great, but I have a couple of open questions such as - What softphones are recommended and provide Call Center level functionality? - What hardware (especially E1 (Euro S2M) class) are supported? I would like to search the list archives, but can't find a searchable archive. Can someone please point me into the right direction? Thanks. Regards,

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'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'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'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

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

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

Hey everyone, I want to run different lines directly to different extensions on two FXO analog interfaces. ie; Zap/1 goes to Ext. 101, Zap/2 goes to extensions 102 Does anyone know of a way to do this? Thanks! Chris

Hi, Has anyone got the X100P to work with an anlogue line in the Republic of Ireland? I have the X100P installed but zttool indicates a Red Alarm status on the card. It is on its own interrupt and I have tried different PCI slots but all to no avail. Are there any alternatives to the X100P that can work with asterisk and are likely to work in Ireland? Thanks, Aaron

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

I have a brand-spanking new TDM31B (3 FXS, 1 FXO) and when I start wcfxs (the only module that recognizes the card) from Zaptel 0.9.1 I get: Zapata Telephony Interface Registered on major 196 Freshmaker version: 63 Freshmaker passed register test Module 0: Installed -- AUTO FXS Module 1: Installed -- AUTO FXS Module 2: Installed -- AUTO FXS

i have been trying to get a newly liberated (from vonage) cisco ata-186 (sip ios v3.1) working properly with asterisk. my client is behind a linksys wrt-54g, which up to this point hasn't proven to be a problem (i have several sipura spa-2000's and polycom phones working just fine behind them). (i'm running cvs-head from yesterday). after looking at the various suggestions,

I've set up a very small * system for a small local paper. The system works great. Here's the issue: I have one of their phone's plugged into the phone port on the x100p and if the phone ring more than 2x then asterisk kicks in and doesn't recognize it as being picked up and starts playing the menu. Can i use wait or something to let the phone ring more and not start the menu?