similar to: Slow to get dialtone when going off hook - big problem for me :(

i need to know something in the zaptel configuration as it seems i can configure detecting the busy tone and hangup after number of busy tone counts, that was great but the problem is sometimes the pstn line has no dialtone and when i try to make call it continue dialing while not having a dialtone! while it should say "all lines are busy/congested" how can i configure that?? i already

Hi All... I have a maddening problem... I have Asterisk configured to pick up a line after 4 rings. I do this to allow my fax machine to pick up a particular distinctive ring pattern, so I don't have to pay for a dedicated fax line. If someone calls the line, lets it ring 3 times and then hangs up, Asterisk answers the line, and holds it off hook forever, constantly playing the

I'm wondering if the page application is broken in 1.2.5 The following: exten => 2001,1,Page(SIP/3254105) does strange stuff. The caller's phone immediately drops into the call, while the callee's phone is still ringing. I'd think it was a SIP messaging issue, except that the Dial() command is working fine, which makes me wonder if it's a bug in the Page appplication.

Anyone interested in this feature? I have a version 0.1 patch, which is currently against 1.2.25-bristuffed, but which should port trivially to almost any version. I am away until Tuesday 8th April, but if there is enough interest, I will open a "new-feature" ticket and upload the patch to the bugtracker so that more capable programmers can laugh at it ;-) It should work reasonably on

I know a few ppl have those CAC Adit 600's with t400 I can't seem to get my second span up on the T400 connected to the second spand on the adit (A:2) A:1 seems ok Can someone post they zaptel.conf span defintions And maybe a "print config" from the adit 600 cli I think my issue is timing srcs the coding, framing. bld out are all matched thx -------------- next part

Hi.. I am about to replace my aging Nortel Venture system with an Asterisk system and 6 Polycom IP 501 phones, and a couple sipura 841's for less used areas. We have 3 phone lines here. One is SBC, one Vonage, and one Voipjet... One hangup is that I can't figure out how to pass through a voicemail waiting indication from SBC. This is important because my wife and her family all

Hi Jean-Marc I am using speex in my voip app (wideband mode). I have noticed that when VAD kicks in, CPU considerably increases (bitrate decreases to 4 kbps). It takes about 7 ms to encode one frame versus 0.5 ms (for speech). This happens regardless preprocessor is used. Although the issue seems to be driver dependent, it occurs on XP, Vista and Mac OS X. Besides complexity and quality, this

Is there a way to generate an old-fashioned dial tone with Asterisk? I'm thinking of one that sounds like: http://www.seg.co.uk/telecomm/dialtone.wav -- Phil Reynolds o ____ mail: phil at tinsleyviaduct.com |L_ \ / Web: http://www.tinsleyviaduct.com/phil/ (_)- \/ Waltham 66, Emley Moor 69, Droitwich 79, Windows 95

I need to select a line from the Zap group channel using the SIP Phone (not FXO and not FXS ports). ignorepat does not work? Also, what is the method to let the second dial tone has another tone frequency? Regards Bilal ---------------- No, ignorepat is for FXS ports (FXS ports use FXO signaling). Also, ignorepat does not apply to SIP phones, because SIP phones provide their own dialtone,

Hello, I'm running * on a very basic configuration. I have a Wildcard TE405P with the first T1 connected to a PRI line and the remaining three to Adtran TA750 channel banks with FXS modules. I successfully configured everything to work with a couple of Swissvoice IP10S handsets (MGCP) and analog extensions connected to the channel banks. The problem I'm having is that when I pick up any

Hi all, Really excited to be using Asterisk and learning about VOIP and PBX's. I'm a complete beginner at telephony but have built and installed Asterisk 1.4.5 and read several of the Asterisk books online and have successfully connected to FWD with IAX2 and to GIZMO using SIP. Just purchased a Motorola Wildcard X100P and installed it into a clone PC running Fedora Core 6. Analog

Guys.. Im new to asterisk but is it possible to simulate a dialtone for example, in other PBX when you pick up the phone you can hear a certain dialup, which is the PBX dialtone, and when you hit 9, you can hear the PSTN dialtone, is this possible? __________________________________________________________________ Anton Krall

Hello, Here's what I'd like to do: call my Asterisk box from a phone, hangup after a few rings, then Asterisk calls me back and presents a dialtone, than I can dial any valid number in the context the call originated. I've done it with CAPI (thanks to the script on http://www.junghanns.net/asterisk/page14.html), I'd like to do it with H323. Problem is, how to present a

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

A question: is there any way to get * to answer certain DTMF sequences entered on an extension with a stutter tone? Long version: I would like to add features to my dialplan like "Caller ID Unblock" which work in the same way that the PSTN works: I pick up the phone, get a regular dialtone, press *82, and get a short stutter dialtone which confirms acceptance of the request, and then

> I'd like to create the following: a user picks up the phone > (gets a dial tone), dials '0' for an 'outside' line, gets a > second (different?) dialtone, and is able to enter an > external phone number. Klaus-Peter Junghanns has something like this on his page: http://83.137.99.170/jn/relaunch/asterisk/page19.html It didn't work for me correctly so I

say I needed to send a broadcast message that I wanted every user to hear when the pick up thier phone? can I "Play,message" on a line just before they get dialtone? or maybe after they dial before ring? how about a "ringdown" to a voicemail box and on end return them to thier line for the dialout? can * do ringdowns? when a user picks up an extension it automagically

Hi: I have a Sangoma A200 card installed in a server with two FXO modules and one FXS module. Analog sets connected to the FXS module have a "squeaky static" -- it's like static mixed with the sound of someone vigorously cleaning a window a few doors down. In other words, it's not a classic static noise, but it is noise, and it's distracting. Remote callers can hear this

Hello. All. I am a bit new to asterisk, started from half a month ago. I am setting up a home asterisk server with analog card. I am using asterisk 1.4.27. At the moment, I bought a X100P card and installed it on my computer. I used it to connect my home phone line. For the moment, it works fine when dial in. Soon I noticed when I dial out through it to my mobile, it can't hang up

Can anyone clue me in about these T400 T1 cards I see advertised? I hear they are Digium Clones. Is there some reason to avoid these? How do they compare to TE410P's for example. Bart