similar to: No ringing using SIP or IAX phone, ringing using ZAP channel

Hi there Background: I want to add DDI and voicemail to users on an existing analogue pabx.. It does not support ISDN. I have 10 DDI numbers via IAX which I am having sent to my Asterisk box. I have 2 X100P cards connected to 2 analogue extension ports of my main legacy analogue pabx. I have set up voicemail for each of my DDI numbers, and when a call comes in for the person at pabx

Hello, almost any useful CLI command has an analogue on Asterisk Manager Interface, but I cannot find a way to get the list of subscriptions using AMI. Which is the command, if any? The CLI command is "sip show subscriptions" Leandro -------------- next part -------------- An HTML attachment was scrubbed... URL:

You can use "Command" command, and "sip show subscriptions" as a parameter -- Alex Epshteyn email: alex at thirdlane.com web: www.thirdlane.com phone +1 415.261.6601 ----- Original Message ----- > From: "Leandro Dardini" <ldardini at gmail.com> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at

hi all, i have an asterisk install with a digium 4 port fxo card and cisco 7960 sip phones -- running on a compaq Pentium III (Coppermine) at 800Mhz 256KB cache and 1GB of ram. when a call comes in on zap/1-1 for example, the delay between when zap sees the line going to ring state, and when the desktop telephone rings can be as long as 7000 milliseconds (or about 3 or 4 rings on an ear piece).

I am testing analog phone and fax machine plugged into Rhino Channel Bank which is connected to TE412P card. This site is in China. I am running RHEL 5, Asterisk 1.4.21.2, Zaptel 1.4.11 and libpri 1.4.4 I ran into a problem which is analog phone can hear dial tone and can make outgoing calls. Another phone (ether internal or external) can call the analog phone ***but the phone does not

So made my first forray into 1.4 and DAHDI and hit a problem. (Not convinced this is a DAHDI issue though...) Testing an analogue line and asterisk sees the caller ID being passed, but then fails to detect ringing. A plain old analogue phone plugged in rings just fine. Console output: == Starting post polarity CID detection on channel 4 -- Starting simple switch on

Hi all, can somebody explain me why i can't hear a ringing tone (alerting) if i'am going to connect to my destination end point? Is it basically so that i have to configure like: exten => xxx,1,Dial,ChanTec/number|timout|r Is it really nessesary to use the "r" option everytime if i want to indicate a ringing tone? This suggest a wrong call flow for the user ... Thanks for

Hi, Asterisk Version : 1.2.15 Card : TDM11B (1 x FXO , 1 x FXS) I have internal dialling working okay SIP->ZAP (analogue phone) and ZAP (analogue phone) -> SIP. The problem comes when I try and make a outbound call. Here is my extensions.conf :- Code: [incoming] exten => s,1,GoToIfTime(17:00-09:00\mon-fri\*\*?outofhours|s,1) exten => s,2,GoToIfTime(*\sat-sun\*\*?outofhours|s,1)

Hi All I have some grandstream phones registered to my asterisk and all internal, external, voicemail services etc are working very well. I am not sure that it is a problem more so an annoyance. If someone dials my extension number or external DDI while I am already in a call rather than skipping to the next priority in the dial plan for example voicemail the line continues to ring and while in

Hello, I've got very annoying behaviour from our asterisk PBX. We have 12 channels T1 e&m wink start for TDM and using iax softphones internally (iaxcomm, but tried firefly-thirdparty and discarded for bad sound quality). Slackware 9.1 w/ kernel 2.4.26+ digium TE110P card. In some cases when call is placed from softphone to TDM, system does not detect call answered on Zap channel and

I have Asterisk running with a combination of SIP and H323 clients. I am using the OH323 module instead of the H323 one. When the SIP clients ring each other, they can hear a ringing noise in the ear peice to let them know that the other parties phone is ringing. However, when the H323 client rings a SIP client, there is no ringing sound at all, although as soon as the called party picks up the

As soon I do a reload I see contant ringing like this on the CLI: -- Zap/14-1 is ringing -- Zap/23-1 is ringing -- Zap/22-1 is ringing -- Zap/20-1 is ringing -- Zap/19-1 is ringing -- Zap/14-1 is ringing -- Zap/23-1 is ringing -- Zap/22-1 is ringing -- Zap/20-1 is ringing -- Zap/19-1 is ringing This goes on continuously and no phones are ringing. I am using a digium T1 card and ADIT 600. Does

Hi Guys, Currently, I have a Sangoma A400 installed with 20 ZAP PSTN analogue lines. The first line is giving me problems due to rain (probably coroded line). My server using FreePBX dials out with g0 (group 0 which includes all 20 lines) and it happens that the bad line is the very first line. Can I simply put ; in zapata.conf like this to seclude the first zap line from getting calls in or

Hello everybody, I just buy a X101p clone and i'm new in asterisk. Here is my configuration : ISDN line ---- Alcatel ----PSTN ext 68-----Asterisk with X101p clone ------sip phone ext 200 - 203 ||| ISDN phones ext 60-67 >From sip phone to ext 60-67 it works. 9+extnumber >From sip phone to Land lines it works. 9+0+phone number >From ext

Hi. Does anyone know of any small SIP phones (and preferably have some experience of using them and happy to recommend them)? By 'small' I mean a single-piece phone, with dial buttons in the handset, so that it can be carried around easily in a laptop bag. Something like http://maplin.co.uk/images/Full/35493i0.jpg (which is unfortunately just a standard analogue telephone).

Peter - 40 phones and only 3 PSTN trunks?. I would recommend at least 2 BRIs for this. If you have ISDN you can also get DDI to the extensions.I would strongly recommend abandoning the analogue PSTN lines and using ISDN. The setup pain you will go through will be significantly less, combined with better audio, more features (like DDI numbers!) and much more robust connections. You should look

Hi, I cannot seem to establish what is causing my analogue line to be generating incoming calls, so I would like to do some debugging on my Zap channel. Can anyone confirm the syntax? I have tried; Debug channel Zap/2 Debug channel Zap/2-1 Debug channel zap/2 Debug channel zap/2-1 Debug channel zap 2 Debug channel zap 2-1 Debug channel zap 02 Debug channel 02 All of

I have a TDM card in a asterisk machine. I found that once I used it to call out, the call status changed to "connected" even the callee is still ring. How could asterisk distinguish the "ringing" and "connected" in zap channel? thanks.

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

Hi guys/girls, We are running a TDM04B card with Asterisk in a Linux box that has 15 GS102 extensions and 1 SNOM190 phone which we are using as an operator console. The FXO ports in the TDM04B are plugged directly into our telecoms provider's analogue lines. Something I've picked up with the SNOM is that sometimes when there are two active incoming calls via Zap channels and the first