similar to: Sip Channel Ring Detect

Hi Everyone, I have a SIP turnk which works fine with both inbound and outbound calling. However, the only issue is that there is no Ring Tone if someone calls us. The phones used are Aastra and Polycom connected to the PBX via VPN (SIP). I do get an outbound ring tone, so it's not that there is any media loss between the phones and the PBX. But when the DID is called there is dead silence

Hi, We are using MozIAX as a softphone with a USB headset and are making outbound calls using IAX with ulaw encoding to our voip provider. We're running asterisk 1.4 Users are complaining that the ring tone generated by asterisk is much louder than the voice call once connected. They are having to turn the volume down to avoid being deafened by the ring tone, but then have an unacceptably

Can anyone help point me in the right direction please? I'm based in the UK and I want to start using a Premium Rate number with Asterisk - I think the equivalent in the US would be a "900 number". Effectively the caller pays much more to call such a number than a normal national or local call. The problem with these is that I don't want Asterisk to actually signal to the

Is there a way to hide the callerid on analog line on outgoing calls. Any ideas whether it could be done through configuration, a script or hardware. Thanks; ____________________________________________________ Yahoo! Sports Rekindle the Rivalries. Sign up for Fantasy Football http://football.fantasysports.yahoo.com

Hello everybody, I have this problem where I can't get a ring tone when SIP devices dial an IAX2 route. I get the ring tone using IAX2 devices to dial the same route. The call completes normally in both cases... Facts: - Asterisk 1.0.9 - The Dial command is within an AGI. - ATA (grandstream) and firefly (SIP mode) would not give me the ring tone at all - Switching to a SIP route works ok -

I try to use a SIP trunk from a VOIP provider to make land to mobile calls. If I do these from a ZAP channel, using an analogue phone, after few seconds of silence (I don't like to generate fake [r]inging) I ear the ringing tone from the mobile operator along with any message the mobile operator decide to say me. If I try to use a SIP phone (or a IAX phone) attached to my asterisk box, I

Hi: Astcc is working fine, except for one thing. It doesn't give the called party enough time to answer the phone. If nobody picks up in two rings, astcc reports back no answer and hangs-up. The only instant NOANSWER "value" was mentioned in astcc.agi script is: elsif ($res eq "NOANSWER") { $res = &mystreamfile("astcc-noanswer");

Hi: I want to connect a wi-fi phone to my Asterisk box through a wi-fi AP so I can make voip calls. please send me your recomendation about what wi-fi phone I should be looking for. Anybody tried the HOP1502 Wi-Fi IP phone. Its listed price $39. Regards; Chawki ____________________________________________________ Yahoo! Sports Rekindle the Rivalries. Sign up for Fantasy Football

The answer is yes, I think, but I don't recall precisely how off the top of my head, and I'm walking out the door in a moment. The phone will hold more than a dozen distinct ring tones which you can create for yourself, and you can have asterisk direct it to use a ring tone independently of line appearance. The most direct way would be with the SIP Alert-Info field, but the phone itself

Hi: It's the first time I have this problem. No matter how I configure my two IAX machines the connection is rejected. "chan_iax2.c:5550 socket_read: Call rejected by ****: No authority found" iax server A: [saad_out] type=peer host=hostip username=username secret=secret disallow=all allow=gsm iax server B: [guest] type=user username=username secret=secret context=tele

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

Hi: Please advice me of a voip provider with reasonable reseller program. I was using voipjet and it has a lot of problems. Did anyone experienced asteriskout.com service? They have good prices. Regards; Chawki Hammoud ____________________________________________________ Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs

There was errors when I tried to start the script recommended by Andrew to boost bandwidth for voip http://www.mixdown.ca/~andrew/dump/rc.tc. This is the output I get : ./rc.tc start RTNETLINK answers: File exists RTNETLINK answers: File exists RTNETLINK answers: File exists RTNETLINK answers: File exists RTNETLINK answers: File exists RTNETLINK answers: File exists RTNETLINK answers: File

hi, in my test, i noticed that sip connection will hangup automaticlly when no speaks between the channel. about half a minute. is this the asterisk inner mechanism or is my configuration error? Thanks! messages on the cli as follow: -- SIP/1003-0000001d is ringing -- SIP/1003-0000001d answered SIP/1004-0000001c -- Stopped music on hold on SIP/1004-0000001c [Jan 18 10:08:42]

I'm banging my head last few days trying to get volume level on channel. So far, I tried using sox with no luck. sox file.wav -r 1 file.dat gave me volume levels but not before Asterisk finish call recording. What I really need is to detect volume level change during the call, say, 30% so I can trigger AMI action. Any idea?

This is my second attempt trying to get help and I am hoping someone can. When the musiconhold extension is matched, Asterisk attempts to execute musiconhold and stops right away, this is what I gets: Executing MusicOnHold("OSS/dsp", "") in new stack -- Started music on hold, class 'default', on OSS/dsp -- Stopped music on hold on OSS/dsp Is there a file that

Hi there, I am seeing some very interesting thing with the latest Zaptel 1.2.X, hope may be someone can shed some light on this. Normally, to dial via your Zaptel T1 card, you would do something like: ;Dial to PSTN exten => _9.,1,Dial(Zap/g1d/{EXTEN:1}) by not adding any option after the extension e.g. no "r" and no "m", Asterisk will pass thru the session media from the

Hi, folks: Okay, so here's an idea. I have a TDM-400 card with an FXO card in it connected to the PSTN and a Polycom IP 501 phone. Observe the following simple dialplan for illustration: > [incoming] > ; incoming calls from the FXO port are directed to this context from zapata.conf > > exten => s,1,Answer() > exten => s,2,Dial(SIP/polycom) And zapata.conf: >

Is there a way to differentiate between a SIP address which hasn't registered (but is within sip.conf) and one that's not there at all (i.e. not in sip.conf) using a straight dialplan. I'd like to differentiate actions depending the state of a SIP device and whether it's in my config or not (if that makes sense, basic automap of dial-in lines to sip phones, but if they've

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