search for: magnaye

----- 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...

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, 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 --

...; by setting Audiomode, 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 > > &...

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

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

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,...

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.

Hello, I am trying to see what I need to do SIP to H323 using G.729. I have Oh323 and SIP working with G711 fine. If I have a SIP client configured to use G729 and H323 client also to G729, how many license should I need to buy from Digium. Many thanks SW

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:

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: