[asterisk-users] SIP REFER Message, over NAT

Hi people, I have a few SPA-942 around, all of them work fine except one. The one behind NAT.. In every phone you can: * Pickup a Call on one of the line buttons, * Create a new call on another button * Press xferLx to join those to calls. This works everywhere except on

[Asterisk-Users] Building a PBX on spain, europe

Hi there, We have to change our PBX to a new one and was thinking about building it with asterisk. The things I need it to do are: Use a few basic ISDN (European) lines for voice, Have a few clients running ip phones (sip, h323.) Make it deliver calls to apropiate extension depending on

[Asterisk-Users] Problem with SIP 407

I’m in trouble with SIP. I’ve got a SIP FXS gateway from www.micronet.info (SP5002/S) and traed to register to asterisk, It seems to autentícate but sniffing the net it shows a 407 proxy authen required error message and I cannot make any outgoing calls from that gateway. That’s what I have:

RE: [Asterisk-Users] Problem with SIP 407

On this cuts note that the gateway has username 'Republica', you could see some reference to Republica2 which corresponds to a second line on the gateway that I have disabled. Thanks for your help! That's SIP debug when dialling '9' (9 would do Goto(s,1)) === *CLI *CLI 11

RE: [Asterisk-Users] Problem with SIP 407

I have commented de Republica2 in sip.conf 'cause if I uncomment it neither Republica and Republica2 register (maybe because they're on the same gateway?) Well, inspite it register well when I try tocall any extension It plays 'busy' tone immediately after Asterisk takes the calls I thought it

[Asterisk-Users] Flash application on H323

I've read about a 'flash' application for Zaptel that could allow to make Flash(somewhere) SendDMTF(remaining digits) Is there anyway to implement that with H.323 ? and SIP ? That's Dial(H.323FXO) SendDMTF(remaining digits) The main issue is that Dial doesn't

[Asterisk-Users] SIP over NAT

Assuming that getting H323 to work over NAT is almost really hard… What is about having both SIP clients venid different NAT’s ¿ is it posible or as hard as H.323? Thanks! Marc. ___ Asterisk-Users mailing list [EMAIL PROTECTED]

[Asterisk-Users] Microsoft Portrait 2.2

Has anyone suceded using Microsoft Portrait with Asterisk? I've seen only one thread about that on the list and it's a bit old, maybe someone has succeeded lately.. It successfully signsin onto asterisk but I could make no calls with it, I can only see these: * Got SIP response 481 Call

RE: [Asterisk-Users] Re: Execute command in shell

- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de James H. Cloos Jr. Enviado el: viernes, 06 de febrero de 2004 9:58 Para: [EMAIL PROTECTED] Asunto: [Asterisk-Users] Re: Execute command in shell Marc == Marc Fargas [EMAIL PROTECTED] writes: Marc I've seen its possible to use

[Asterisk-Users] Execute command in shell

Is it posible to make Asterisk execute a command on extensions.conf during a call ¿ (That's to transfer H323 call by telnetting the gatekeeper so Asterisk doesn't seem to like transferring h.323 ) Thanks! Marc ___ Asterisk-Users mailing list

RE: [Asterisk-Users] Execute command in shell

I've seen its possible to use the System applications, but what about passing arguments to the command ? Thanks for your help! -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Marc Fargas Enviado el: jueves, 05 de febrero de 2004 13:37 Para: [EMAIL

RE: [Asterisk-Users] Can audio streams go client to cleint with IAX?

Is it possible to make audio streams go client to client with H.323 ? (both client being H323) Thanks! -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de T. Chan Enviado el: lunes, 02 de febrero de 2004 23:56 Para: [EMAIL PROTECTED] Asunto: RE:

RE: [Asterisk-Users] Can audio streams go client to cleint with IAX?

speaking, unless you're using an rtp proxy, the rtp audio should go client--client. H323 does the call setup and teardown and such, but the audio stream is usually direct. Jeremy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marc Fargas Sent: Monday

RE: [Asterisk-Users] Can audio streams go client to cleint with IAX?

possible as is asterisk currently capable of it, then no. It wouldn't be too hard, look in channels/h323/ast_h323.cpp to see where Jeremy started on it. You just have to get the openh323 lib to initiate the transfer. - Original Message - From: Marc Fargas [EMAIL PROTECTED] To: [EMAIL

RE: [Asterisk-Users] Can audio streams go client to cleint with IAX?

Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] Can audio streams go client to cleint with IAX? what clients are you using? They probably won't support SIP - Original Message - From: Marc Fargas [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, February 03, 2004 11:18 AM Subject

[Asterisk-Users] Transferring H.323 Call

application and both get in touch trhough asterisk, but using Call Asterisk stays on the middle and the sound quality gets poor. Is there any way to ‘transfer’ the call so Asterisk doesn’t stay in the middle ?? I use OpenH323GK (www.gnugk.org) as the gatekeeper. Thanks a lot.    Marc Fargas PS: SORRY

[Asterisk-Users] Transferring H.323 Call

Asterisk doesnt stay in the middle ?? I use OpenH323GK (www.gnugk.org) as the gatekeeper. Thanks a lot. Marc Fargas