similar to: announcement using Dial

Hi, I just setup my * with digium. I started testing voicemail first between atas, and i am not sure why it is not prompting me any when the call is not answered or if busy. i only get continuous ringback and the following message: asterisk*CLI> -- Executing Dial("SIP/6882332-1697", "SIP/5104112978|20|tr") in new stack -- Called 5104112978 --

----- Original Message ----- From: "Jess Magnaye" <jess@arretni.com> To: <wipe_out@users.sourceforge.net> Sent: Tuesday, January 06, 2004 3:19 PM Subject: Re: [Asterisk-Users] Pls confirm > Is the format "allow=g723.1" in sip.conf valid? > > somehow i cannot get it working to do g723 passthru. also, i've read that > doing g723 will disable

How can I capture a "-1" result on a Dial command? Basically, I have the following setup, and I want to be able to process the audio file after the outbound call has been done regardless how how it ends. No matter how the call ends, I can't get "macro-record-stop" to run. Any help would be great. -Eric <from extensions.conf> [macro-dialanalog] exten =>

Can someone on the list confirm if Asterisk can do g723 or g729? when connecting to provider? or it is only supporting g711? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040106/1d6c78cb/attachment.htm

I am just wondering if this is normal. I have my * running for a week now and I'm still testing its interoperability with other voip provider (in sip using codecs other than g711). yesterday, i changed my linux's (RH9). and since the new ip i assigned is located on a different site, i have to shut it down and move it physically. after that, i cannot run my * anymore. i am getting this

Hi all First I want to wish for everone a happy new year... Well... I have run asterisk 1.4.16.1 in a server. I have this IVR, in extensions.conf: [ura] ;exten => s, 1, Wait,1 exten => s, 1, Answer() exten => s, n, Noop() exten => s, n(debug),DumpChan() exten => s, n, Set(LANGUAGE()=pt_BR) exten => s, n, Set(CALLFILENAME=/var/spool/asterisk/monitor/entrada/) exten => s,

I have the following dialplan: ; calls prefix by '8' are recorded exten = _8[01]./_251,1,Set(something=shortened) exten = _8[01]./_251,n,Set(WAV=filename) exten = _8[01]./_251,n,Monitor(wav,${WAV},mb) exten = _8[01]./_251,n,Dial(mISDN/2/${EXTEN:1},,g) exten = _8[01]./_251,n,System(send-recorded-conversation ${WAV}.wav ${EXTEN:1} emailaddr) exten = _8[01]./_251,n,Hangup() The idea is that

Hello; I have a problem with voicemail and my asterisk 1.2.1 on a OS X Mac Pro intel box. When I try to record a message from an incoming call or a greeting message from internal phone using voicemail, It's like something is not doing well. I can heard anything, only a distorsion sound that is equal to lenght of the message left. First I thoug that could be something with format=gsm|wav.

> Thanks for the reply. > > 1. My VAD is turned off (00140014), and it didn't help for that cut-off. I > am not sure if OutboundProxy has to be configured to have it working fine. > Or this just happened to me? What is your ATA's software? > > 2. I tried dtmfmode=inband on sip.conf, and dtmfmode=rfc2833. None worked. > As per ATA, it is by default using rfc2833.

On Thursday, January 22, 2004 9:55 PM, Jess Magnaye [SMTP:jess@arretni.com] wrote: > in telco world, there's like 64kbps per channel and voice can be > carried on a 16kbps channel. is it possible to configure asterisk to > make 4 extensions (ATAs example), to call out using single FXO port > at the same time? if that is possible, then is it also possible to > make t1-pri to

Anybody on the list who implemented Cisco ATA + * + Cisco 2600? I cannot get my calls from ATA to terminate to the Cisco gateway via *. I am not sure if it is my hardware problem. I'm getting the following "codec negotiation problem" from Cisco. 23:39:08: Unexpected VoIPCodec Type :g729br8 23:39:08: Unexpected VoIPCodec Type :gsmefr I appreciate any help I can get. Thanks.

Hollo I'm new, My name is Ariel Arelovich I've encountered the following problem trying to sue metaflac. I used the following command: c:\Archivos de programa\FLAC>metaflac --set-tag-from-file="CUESHEET=sola.cue" so la.flac sola.flac: ERROR: file 'sola.cue' for 'CUESHEET' tag value has embedded NULs And I've got that error. See I have the flac that is

I have a question on how to handle the "h" routines. I have noticed that if the call is hung up by the side that originated the call, the "h" routine is not extendable via a macro, or at least I have been unable to do it. My tests have included only SIP->SIP calls. If the originating side hangs up first: The macro is called from "exten =>

hmm... your cut-and-paste looks like ascii, but is the cuesheet in utf-8? --- Ariel Arelovich <aarelovich@gmail.com> wrote: > Hollo I'm new, My name is Ariel Arelovich > > I've encountered the following problem trying to sue metaflac. I used > the > following command: > > c:\Archivos de programa\FLAC>metaflac >

I have an application that requires I be able to dial into an asterisk box, then from there dial out to another user through a PSTN. I'd like to be able to both 1) record this call and 2) let another user dial in using something like ChanSpy to listen to the conversation. I can get this working by executing an auto-dial script to connect one end of a call to an outside Asterisk box which

I'm trying to get this zap dial to work. I want to send DNIS and ANI to other system (ZAP/g2) at answer, while the caller hears ring (RBT). I hear the RBT, but callerid and exten is not sent to other T1 - The ZAP/g2 T1 is standard D4, SF, E&M Wink Start. - At ZAP/g2 wink, asterisk should send DTMF "*ANI*DNIS*" exten => _XXXX,1,NoOp,${CALLERID} exten =>

Hi! Trying to figure out what I am doing wrong... 1 std SIP phone (0317998975) registered to an Asterisk (SVN-trunk-r234256) 1 Cell phone 00733025975 attached through H323. extensions.conf exten => 975,1,Goto(975-${DEVICE_STATE(SIP/0317998975)},1) exten => 975-INUSE,1,VoiceMail(0317998975 at inputinterior.se,bs) exten => 975-INUSE,2,Hangup() exten =>

Hello, Limiting the call duration with the L-option of the Dial()-command is working fine, however the announcement is not played. Dialplan : exten => _367,n,Set(LIMIT_PLAYAUDIO_CALLER=yes) exten => _367,n,Dial(SIP/test6,,L(11000,5000,5000)) The call lasts for 11 seconds, but 5 minutes before time runs out an announcement should come. I hear no announcement, not on caller-side nor on

Network Configurations Block D, Surrey Park, Barham Road, Westville, 3610 Helpdesk: (086) 163-8266 Tel: (031) 266-1563 Fax: (031) 266-4206 Hi people. I'm hoping someone has come across this problem with version 1.2.14 In my dial plan I call various SIP phones using the following little macro: exten => s,1,Set(TIMEOUT(absolute)=14400) exten =>

I can smile now. I made my * work with my Cisco. Finally. First problem was Ethernet on my Linux. After installing * on a different machine, I got rid of that "icmp udp unreachable" error. My next problem was the call stays on on Cisco gateway, but the ATA drops it. I figured out it was my mistake on dialplan in extensions.conf --- (it took me a day to notice it.. damn!). my config