similar to: Sometimes yes - sometimes no (dialplan)

We come into this section of the dialplan: exten => 88670333333,1,Wait(1) exten => 88670333333,n,SayUnixTime exten => 88670333333,n,NoOp(If you know the extension ...) exten => 88670333333,n,Dial(${PHONE_6003}) The caller from the GK hears only ringing, not the time. The extension 6003 rings and I can pick up, but without any voice nor video. athome*CLI> -- Executing

I installed Asterisk as per instructions in the FAQ on the digium.com site. Double checked it. I also think they have a bug in the zapata.conf where the context should be incoming and not internal. 1) I hear no dialtone when I pickup the phone on the S100U. Asterisk sees the event and displays the message on the screen. I tried dialing but nothing happens. I hangup and * shows the hangup event.

I have three video phones here for testing: Extension 6003 is Eyebeam Extension 6004 is a hard phone (model 8770) Extension 6005 is a hard phone (model 8882) Can anybody have a look at my settings and the output I get from all kinds of dialings, please. The sip settings for all phones is (user / password different): [6003] type=friend username=6003 secret=pwd qualify=200 nat=yes host=dynamic

I'm all up for reading and looking round for people in the same boat to try and solve the issue together, but there appears to not be large community yet, just the asterisk mail lists. I got Asterisk working with X-Lite great now for internal calls and also calling land line numbers etc. The two problems i'm currently having are: 1. When someone calls in the phones ring 3 times then

Hey Everyone, I downloaded and installed the X-Lite softphone the other day (the lite version) and cannot seem to get it to work well. Don't get me wrong, it registers with my asterisk server and everything seems to work well, except the call quality really is horrible. I thought it may be the place I was trying it at (DSL) so I took it to the office and tried it right next to the asterisk

Hi, I had asterisk installed, ISDN-adapter, some x-lite software-phones and I can call betweens the softphone- and 'normal' phones during the ISDN-card. 2 questions now 1) Is it posible to create policies, so that some SIP-users can dial ALL numbers, and some SIP-users not are allowed to dial eg. 900xxx-numbers, 30xxxxxx (mobilphones), 40xxxx(long distance) and if possible on time

I have just started developing asterisk, and am trying to start simple. I have a X100P device and an S100U device. I am trying to use the examples provided, where I add a few lines to the /etc/zaptel.conf, /etc/asterisk/zapata.conf, and /etc/asterisk/extensions.conf so that I may connect an analog line to the X100P and an analog phone to the S100U. When I dial the analog line, it should ring

Aloha, I have a Cisco Gateway the is functioning as my SIP Gateway to the PSTN. I have an Asterisk box, RC2 with a for port FXS card providing dialtone for a Norstar Key System. I have it working so when you press a line key on the Norstar you get dial tone from the Asterisk box. The user has to dial '9' then they can dial there number which is sent to the Cisco GW via SIP and the call

Hi guys, sorry to be iterating this on the list once more, but I'm not able to get this stuff to work as I'd expect. So far, I've always managed to keep it out of NAT environments :-> My home LAN is NATed by a simple Draytek router. In the home LAN is an ATA186 with SIP. On the internet (public) is an Asterisk server. I have nat=yes in the sip.conf and the connectmode is set

I'm looking to expand my bandwidth for my Asterisk PBX. Why should I choose a T1 over DSL for my asterisk server? I found someone offering T1's for $290 a month + Loops or 3 Meg for $561 a month + Loops. Is this a good deal? Thanks Bart -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, Can anybody explain what these values exactly mean. As you all know its the dialtone value on an SPA3000 of linksys. 350@-19,440@-19;10(*/0/1+2). Can anybody help me how to write this code for a dialtone of frequency 425 which is continous. Thanks Dan -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

Do you have threewaycalling and transfer set in your zapata.conf? Here's mine (four TDM400's, seems to be working so far). I didn't do anything in my extensions.conf for any of these features (what confused me at first is the t and T options of the Dial application in extensions.conf are for transfers via the # key), when you flash you get another dialtone that works just like the

Alarms Span RED wildcard X101P Board1 OK wcusb/0 0 ive got my asterisk server up and running and working correctly, the first time after a reinstall and reboot everything was fine - i had both alarms OK and i could get the USB extension ringing when i ran the house number from my mobile. as soon as i tried again i got a red alarm on the wildcard board. now im using the sample

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,

Is there a possibility to send text based messages from/to a sip phone (text display) or to a video phone or from/to a messenger? bye Ronald

Matthew, can we use real-time also for H.323 phones? (h323_buddies) ??? bye Ronald

dear Hackers, i have a voicetronix Openswitch card, and i have been finding it very dificult to get it to work with asterisk. i intend to connect 8 ports to the PSTN and 4 as station ports. problem 1: On running asterisk all i get at first i get : event[9=>[11] station OFF hook] on vpb/1-12 even [12=>[11] loop drop on vpb/1-12 event [12=>[11] Tone detect:GRUNT event [2=>[11] Dial

First, please forgive me if this is a total newbie question, I've only just begun to scratch the surface of asterisk. I currently have a dialplan set up to let me dial a specific extension, authenticate the user, then have * dial a hard coded/programmed overseas number. What I would like to do is set up my dialplan to have an extension that offers up an outbound dialtone allowing the

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