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...gt; 6882332,103,Voicemail,b6882332 exten => 5104112978,103,Voicemail,b5104112978 exten => 1111,103,Voicemail,b1111 ;get message exten => 1234,1,VoicemailMain(6882332); exten => 9999,1,VoicemailMain(1111); voicemail.conf --------------------- [default] 6882332 => 6882332,test1,jess@arretni.com 5104112978 => 5104112978,test2, jess@arretni.com 9011 => 9011,Asterisk,jess@arretni.com 1111 => 1111,Nada,jess@arretni.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040105/793f5339/atta...

----- 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 g...

...but nothing helped. I was thinking the * is ONLY > recognizing the DTMF if there is telco board installed. Is it? > > > ----- Original Message ----- > From: "Philipp von Klitzing" <klitzing@pool.informatik.rwth-aachen.de> > To: "Jess Magnaye" <jess@arretni.com> > Sent: Tuesday, December 23, 2003 12:36 PM > Subject: Re: [Asterisk-Users] Fw: Questions and finding > > > > Hi! > > > > > 1.) First test > > > - ATA1 calls to ATA2. When voicemail starts playing, it just cuts-off > > > after 5-10seconds (...

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

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 poss...

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.

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

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

Hello to the list again. I have my ATA behind NAT connecting to * then calls are fwd to Cisco 2600. Calls are completing, I just cannot figure out why I don't hear any ALERTING signals from the 2600 (ringback, fast busy, SIT, etc). Audio works fine though. I'm using G711ulaw. And I don't want ATA or * to provide the false ringback so I took out ('r') in my Dial command.

I would just like to follow-up on the ringback problem I'm getting from *. As I've said in my previous post, I am not hearing the "real ringback" from the Cisco gateway terminating my call. I don't want to provide false ringback from * (r option of dial), because it'll still give me ringback even if I am suppose to hear announcement or fastbusy. Below is captured ISDN

IF I want to play sound files, 1.) what format should it be? (*.au or *.wav) 2.) where should it reside? 3.) what syntax should I follow? Is exten=>_.,102,Dial(SIP/${EXTEN}@ip,1,tHA(sound.au)) correct? I tried this and it doesn't work. Thanks, -------------- next part -------------- An HTML attachment was scrubbed... URL:

it just came to my mind, and i haven't done any researches yet if somebody tried this one with asterisk.. :) well just in case somebody or someone on the list aware, i appreciate any advise. 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

I installed * to primarily test its voicemail feature. I installed it on a server WITHOUT any telco board (i.e., digium). Installation looks ok, however I am having problems. MY SETUP: 2xATAs are configured to use * as GkorProxy Asterisk is registered to my SER SIP/RTP Proxy 1.) First test - ATA1 calls to ATA2. When voicemail starts playing, it just cuts-off after 5-10seconds (consistently).

Here's my sip debug output. anybody knows why Cisco sent * is CANCEL msg? Can someone tell me what ATA version are they using? Maybe this is also another issue.. I am using v2.16. This is using G711ulaw. Sip read: > SIP/2.0 100 Trying Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK13fbce3e From: "Jess" <sip:6882332@mydomain.com>;tag=as6818ebfb To:

Is Asterisk capable of handling video conference? I am wondering if there is anybody in the list who tried it with NetMeeting(s). If it is possible, is the * required to register in the GK for this purpose? or making it as h323gw only is enough. -------------- next part -------------- An HTML attachment was scrubbed... URL: