similar to: TO have ringing tone instead MOH

Displaying 20 results from an estimated 1000 matches similar to: "TO have ringing tone instead MOH"

2006 Mar 30
3
Please Help Test Quad PRI Using NFAS
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
2006 May 02
1
SIP trunk ring tone
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
2006 Mar 31
8
1.2.6 doesn't use mpg123?
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
2007 Mar 08
3
Boot order of 2 TE110P and 1 TDM400P in the same machine
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
2007 Apr 17
1
Transfercapability DIGITAL
Hi I have a requirement to bridge Digital ISDN call through an asterisk box but no matter what I setup in the dial plan the second leg of the zap bridge is always set to Transfer Capability of SPEECH, I wondered if any one has come across this and managed to fix it? Thanks in advance for your help Robb
2006 Mar 29
1
Blacklist out bound numbers from file
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?
2006 May 02
1
Sangoma Card Question
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?
2007 Jan 11
1
Queues Service Level
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
2007 Jan 11
2
Asterisk Compilation and Installation
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
2007 Jan 18
1
sangoma a102d + Asterisk 1.2.14 ... bridging together 2 call legs on same PRI?
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:
2007 Feb 14
1
CAS signalling on span 2 conflicts with HDLC with FCS check on channel 20
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
2007 Mar 07
1
auto dialer
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
2007 Apr 17
2
CDR datasets
Hello list, I have been working lately on a small CDR parsing utility, and would like to do some performance testing on it. I am looking for some - possibly large - real-life Asterisk CDR datasets to run some performance monitoring. Anybody's got some CDRs that can be shared? Thanks in advance, l. -- Loway Research - Home of QueueMetrics http://queuemetrics.com
2006 Mar 25
2
help on mfc/r2
Hello there! I've problem with setting up unicall / mfcR2. can't find proper notation for channel, trying unicall/1, unicall/1/1001, unicall/g1, unicall/g1/1000 and still having no luck. klaudia*CLI> !cp call /var/spool/asterisk/outgoing -- Attempting call on Unicall/1001 for application Dial(363) (Retry 1) Mar 25 09:29:34 NOTICE[19920]: channel.c:2429 __ast_request_and_dial:
2006 Mar 23
1
IAX Bridging and not recording CDR correctly
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]
2007 Feb 06
2
Mysterious tables starting with "stats_"
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,
2006 Apr 25
1
MFCR2 in Brazil, someone?
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
2007 Jan 12
3
5v capable motherboards
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
2006 Mar 30
2
multiple auto attendants
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
2006 Apr 07
2
Attended Transfer howto
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