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Please help me test my setup by dialing 800.564.0215 and listen to the queue for a bit. I have a quad port T1 with NFAS setup. I can dial-out but I cannot dial any 800 numbers (Global Crossing says I need LDS service and that will be a couple weeks) so I cant test it myself. I need at least 24 callers to feel comfortable enough that it is working properly. Thanks, Steve Totaro

Is it true that asterisk 1.2.6 does not use mpg123? I just installed asterisk 1.2.6 and while I do have music on hold (through format_mp3?) I do not have an mpg123 process running. I seem to be having serious audio issues when going through one of my providers (and just through that provider) when using mp3 for hold music, however when using wav files it is fine. The processor is only at about

I'm looking to bock a list of numbers users cant call. Is it possible to pull these from file specified in the dial plan, as apposed to mysql?

I need to avoid MOH on my asterisk box, so i need to have a ringing tone when attendant transfer is made, or a call is on hold.. Is there any way to do that. I did not see a simple way to do that. Regards

Hi, I'm wondering what I need to change to get the "swedish" type ring on a SIP-trunk. When I make an inbound call i still have the "US"-type of ring on my SIP trunks. I need help on changing this. However I've successfully changed this on the Zap interface for all inbound calls. Thanks in advance! Regards, Jan

Hi, I have a Sangoma 200A (I think that's the model #) analog 4 port card. It works great... however almost everytime after someone hangs up a call they were on.. the system rings the call back in, as though it were a new call coming in. When they pickup no one is there. Can anyone suggest why this is happening, and how I can make it stop?

There seems to be something about SL for queues since when the show queues CLI command is used, it give something like "SL:0.0% within 0s": pbx*CLI> show queues 1 has 3 calls (max unlimited) in 'rrmemory' strategy (243s holdtime), C:174, A:9, SL:0.0% within 0s Members: SIP/1242 (dynamic) has taken no calls yet SIP/1251 (dynamic) has taken 4 calls

Hi List; I understand that I have to compile zaptel but what about asterisk? Is it enough to extract it? Well, how I will run asterisk (without compilation and installation)? Any advise? Regards Bilal ____________________________________________________________________________________ Yahoo! Music Unlimited Access over 1 million songs. http://music.yahoo.com/unlimited

Hi all, Are there any issues to be concerned about when calls come in from PSTN to a PRI card and are forwarded back out the same PRI card? Anything different have to be enabled in zaptel.conf or zapata.conf or the Sangoma configs to make this work? What about using .call files that join two ZAP channels? Channel: ZAP/1/4081234567 MaxRetries: 0 RetryTime: 60 WaitTime: 60 Application:

hello my friends, when i make a genzaptelconf i get this message ******************** CAS signalling on span 2 conflicts with HDLC with FCS check on channel ******************* Any idea Please? I m installing zaptel 1.4 i checked in "http://bugs.digium.com/view.php?id=7860" that it's a bug but beacause i m a newbie in asterisk i can't undrestand what exactly mean Thank You

Not able to get the auto dialer part of asterisk to work with the zap channel. It works great with the sip channel. Here is the call file and the CLI output Call File Channel: ZAP/G1/6144994925 MaxRetries: 3 RetryTime: 40 WaitTime: 2 Context: amaxx Extension: 36652 Priority: 1 CLI Output Connected to Asterisk SVN-branch-1.4-r57207 currently running on VoIP-PBX (pid

I have a user who is off my system with IAX. When he calls and goes out my long distance provider my asterisk switch seems to be bridging the two calls. As a result I loose all accounting information. All I get is the call setup time (15 or 20 seconds). How can I either make asterisk not bridge the call, or keep correct tabs on the call accounting for me? Mar 23 21:55:24 VERBOSE[18185]

I have a server which I haven't installed that I have to maintain. This server uses MySQL, it has an asterisk database and in there some mysterious tables: stats_action, stats_agent, stats_callid, stats_config, stats_estados, stats_qstats, stats_queue, stats_queuexagent. I say mysterious because I don't have a clue about who is generating them. I did a grep for their names in /var,

Does anybody have a working Asterisk server with Unicall using MFCR2 in Brazil? Were having problems. It seems SPANDSP never detect the tones from the telco. Im using brazil protocol variant. Im having lots of problems to find out why spandsp seems to not detect the MF tones. We send the first digit, the telco says they receive it, and respond with the proper signal to ask for the next digit, we

Anyone have a suggestion on where I can get a decent new MB with 5v capable PCI slots. It seems like every decent server MB on the market has 3.3V slots only. Is diving into the junkbin my only choice if I can't afford to replace the 5v quad-T1 wildcard? Thanks Mark Farver

Hi I was given the task to try and build a VOIP solution to an office building with multiple tenants in all sizes and shapes. Some of them will require auto attendants and some will simply want direct lines to their phones. The question I have is: Can asterisk be configured to handle multiple auto attendants? TIA Paolo

There is plenty of information on the wiki for setting asterisk up for transferring calls both from the Dail() command, and features.conf. What really seems to be missing, is simply how do you actually perform the transfer? Blind transfers are pretty simple as you only have two obvious steps. How though do you do attended transfers? 1.) You have a call 2.) You dial *2 or whatever you have

I want to limit the number of simultaneous incoming calls that my IAX DID can accept to, say, 2. The IAX DID provider sets no limit. The code below does work, but when the limit is in effect, new callers hear a "call cannot be completed as dialed.." message instead of a busy signal. Maybe this is an issue with the provider, but I do not like this and want callers to hear a busy signal.

Helo List, Sorry I missed the rest of my email in my previous post. Please see below. I'm having an issue using the AND (&) operator evaluation in the code of my dialplan. The dial plan is coded to detect inbound DTMF digits from callers. key "1" is equivalent to "yes" and key "2" is equivalent to "no" in two diferent contexts in the dial plan.

Hello Everyone, I checked with zttool that sometimes after the machine boots the order of the boards is changed like this: ??? Alarms Span ??? OK Digium Wildcard TE110P T1/E1 Card 0 ??? OK Digium Wildcard TE110P