similar to: No Incoming Ring Tone (Even with "r" option)

are you giving answer()? ..o-------------------------------------------------------o.. Brian Fertig Network/Systems Engineer IT Administrator Planet Telecom, Inc. Tampa,FL Office -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Juan Jose Comellas Sent: Friday, September 30, 2005 10:32 AM To: Asterisk Users

I am using Asterisk (Debian unstable packages) with an OH323 connection to my provider. Everything is working except for the generation of ringback tones when I receive inbound calls from the PSTN. My provider tells me that we're sending call progress indications and that because of this they're expecting us to generate the ringback tone. Does anybody know how to configure this in

I am trying to make an Inalp Smartnode 1200 (SIP-to-ISDN gateway) work with Asterisk. It works ... Partially. We are using the Inalp to connect ISDN phones, it basically acts like an ISDN ATA. First of all, when I make a SIP call to the unit with a simple Dial() command (no "r", so Asterisk shouldn't generate its ringback tone) I hear Asterisk's ringback tone anyway (I'm

Hi, I've noticed that sometimes instead of getting a regular ring tone when calling out on a Zap channel, I get this obnoxious loud noise which forces me to hang up. Is this a problem in the Zaptel driver? I seem to recall that ringback tones are generated by zaptel when dialing out from a SIP phone over a Zap trunk. Thanks.

Any ideas on why callers who call into my customer's SIP trunk are not hearing a ringback tone? I had this on one other asterisk system, and wound up needing to set progressinband=yes in the SIP trunk config. I have set this on the current system & restarted asterisk, but to no avail. I am using: AsteriskNOW distro Asterisk build is 1.6 from AsteriskNOW repository:

Hi everybody, Every time callers reach my FXO port, asterisk produces one ring tone just before it executes Answer(). How to remove this? I have commented "#define RINGBEGIN" on zconfig.h, but it does not help. Thanks in advance for your help. Cheers, Anto -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello all, Just starting to setup asterisk v 1.4.11 and need to run three distinct phone systems for three different companies. So far, I have inbound lines going to the appropriate dial plan within the extensions.conf file. I'm using exten => _X.,1,NoOp(FROM NUMBER: ${SIP_HEADER(TO):5:10}) to determine which number is being dialed by the caller and then using a gotoif to get to

Hi folks, (3rd time I am sending this message, none of the other appear to have made it through!) Using "OpenSSH_6.9p1 Ubuntu-2ubuntu0.1, OpenSSL 1.0.2d 9 Jul 2015" on the server, "OpenSSH_7.2p2, OpenSSL 1.0.2g 1 Mar 2016" on the client. I am trying to use sshtunnel with StreamLocal forwarding to enable me to connect back to the client's ssh port, without having to

Hi, I have PRI configured and up but when i am dialing outside i am not getting any ringback tone but my call is connected. following is my example SIP----------------->PRI ------------> mobile I have set progress=yes in chan_dahdi.conf but still not working if i call inbound from my mobile to internal extension ringing working please help me -S -------------- next

hi: When iam sending calls through sip a fake ringback tone is generated and then call status can't be viewed (if call is ringing,busy,offline) it just rings and rings. Can i disable this? Thanks in advance. _________________________________________________________________ Windows Live?: Keep your life in sync.

Hi, I use in Production : Asterisk 1.2.9.1 We Use Asterisk as a SIP Transit Server to record centrally all the calls. The call flow would be: incoming calls : PSTN -> GW -SIP-> Asterisk(Record) -SIP-> Softswitch -> IP Phone outgoing calls : IP Phone -> Softswitch -SIP-> Asterisk(Record) -SIP-> GW -> PSTN Dial plan in Asterisk is quite simple: [record] exten =>

Hi, The code definitely attempts to unlink any old listener beforehand (see misc.c:unix_listener()) so I don't understand why that isn't being called. You might try simulating your configuration using sshd's -T and -C to make sure the flag is correctly being set. Could chroot be interfering? Some platforms implement additional restrictions on devices and sockets inside chroot. -d

On Tue, 3 May 2016, Rogan Dawes wrote: > Hi Damien, > Thanks for the response! > > I tried moving the StreamLocalBindUnlink directive outside of the Match > rule, and it worked. But that doesn't explain why the Match was not > correctly setting the directive: > > This is running on an alternate port with -ddd: > > debug3: checking match for 'User

Good morning, i've noticed many times that there are IVRs that play a ring tone just before bridging me to an agent. My asterisk does not behave like this but i've always wanted to. I'm now playing with 1.6.2.9 and i've read in queue's doc: http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue R ? stops moh and rings once an agent is ringing (Asterisk Trunk) (in

I'm having the following problem: I call into my Asterisk box (RedHat Linux 9.0, 1 Digium X100P card) to access voicemail. After dialing the appropriate extension I get voicemail, am presented with the user's unavailable message, and can leave a message normally. The problem comes when I press "#" to end the recording, at which point I am told "Your message has been

Have a problem which can't find solution to on WIKI.. Trying to get * to use UK based indication tones. i.e. british ring, dial tone, busy signal. Have changed the indications.conf file to default to UK. However this seems to have no affect. What am i missing. Am using 1.0.3 stable. Many thanks Andrew. ---------------------- indications.conf [general] country=uk [uk] description =

Hi. Has anyone had any success in implementing a callback or ringback function in Asterisk? I've had a look at the callback-voicemail example on voip-info.org http://www.voip-info.org/wiki/view/Asterisk+tips+callback However it won't quite work for me. I need it for local SIP users which most of them don't have voicemail. If one SIP user calls another SIP user and the second user is

Hi All I am trying to simplify a dialplan for a few thousand users. Would what I have below work? If someone dials exten 710001 would it go through answer and then to the macro to try dialing the SIP phone thats registered on 710001 and then onto voicemail if no answer or not signed on? exten => 71XXXX,1,Answer() exten => 71XXXX,2,Macro(71macro,${EXTEN}) exten => 71XXXX,3,Hangup()

On Sun, May 8, 2016 at 9:04 PM, Markus Friedl <mfriedl at gmail.com> wrote: > I have an ugly patch for that feature that requires protocol modification. Why does it require a protocol modification? Couldn't the client request regular forwarded-tcpip from the server then decode SOCKS entirely within the client? -- Darren Tucker (dtucker at zip.com.au) GPG key 8FF4FA69 / D9A3 86E9

Hi! Part of extensions.conf: exten => 985,1,Dial(SIP/0317998985&H323/00702221448 at Avaya,20) exten => 985,2,Goto(985-${DIALSTATUS},1) exten => 985-BUSY,1,VoiceMail(0317998985 at inputinterior.se,b) exten => 985-BUSY,2,PlayBack(vm-goodbye) exten => 985-BUSY,3,HangUp() exten => 985-NOANSWER,1,VoiceMail(0317998985 at inputinterior.se,u) exten =>