RE: [Asterisk-Users] Most Reliable Proxy Server?

Thank you.Simon Brown [EMAIL PROTECTED] wrote: You could try these: voiptalk - www.voiptalk.org sipgate - www.sipgate.de Simon From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ron McMillinSent: Thursday, 15 April 2004 15:29To: [EMAIL PROTECTED]Subject: [Asterisk-Users] Most

[Asterisk-Users] Most Reliable Proxy Server?

Hi all, Do you know if there's any free public SIP proxy server that is more reliable that FWD and Iptel? Thanks Ron

Re: [Asterisk-Users] Dial Outside SIP address from AGI

Thank you. This explains it.Nathaniel Powning [EMAIL PROTECTED] wrote: On Mon, 12 Apr 2004, Ron McMillin wrote: Is it possible to dial an OUTSIDE SIP address while inside AGI application? For example, within extension context, I could use [from-sip] exten = 7723,1,Dial(SIP/[EMAIL PROTECTED

[Asterisk-Users] Dial Outside SIP address from AGI

Hi all, Is it possible to dial anOUTSIDESIP address while inside AGI application? For example, within extension context, I could use [from-sip] exten = 7723,1,Dial(SIP/[EMAIL PROTECTED]) and this works whereas when I'm inside agi app, $AGI-exec('Dial',"SIP/[EMAIL PROTECTED]") and thisDOESN'T

[Asterisk-Users] SoundCard and Voice Quality

Hi all, If I'm just using Asterisk as PBX and calls going through between ouside lines and inside extensions, (not using any softphone running on the asterisk pc), does what soundcard I use affect voice quality at all? Do I have to get a full duplex soundcard? Thanks Ron

[Asterisk-Users] Agi and bridging problem when codecs differ

Hi all, I have encountered this problem: if the caller isconnected to the callee using Dial() commandcalled from extensions in extensions.conf, there is no problem. But if the same caller and callee are connected using an AGI-exec('Dial'...), the line is disconnected when asnwer. There's a

Re: [Asterisk-Users] two-stage dialing

Hi, This is not gonna work, is it? Is there such thing as Dial_but_not_connect_? I am trying to do the same thingbut don't know how to accomplish this. If you've or anyone here figured out, please let me know. Thank you very much, Ron [EMAIL PROTECTED] wrote: I am trying implement two-stage

[Asterisk-Users] FXSFXO Port Converter Problem

I would like tocall into my asterisk box and depends on extension, will call to different ATA (which has different SIP addresses). ATA will ring FXSFXO port converter and will allow me to dial a PSTN line that the FXSFXO porter converter is hooked up to. The problem is DTMF detection for the relay

[Asterisk-Users] DTMF Detection Problem

Hi, My set up is like this Asterisk---SipuraATA-AnalogPhone When I'm calling into asterisk from a cell phone, there's no dtmf detection problem as asterisk can detect correct extensions that I press. But when the phone is further connected to the AnalogPhone thru the ATA, the dtmf signal is

[Asterisk-Users] X100P fails to detect user hung up

I am using the wildcard X100P with *. PSTN line comes in to the FXO port of this card. Everything works fine most of the time. However, occasionally Asterisk doesn't seem to be able to detect the user has hung up and therefore tie up the line for quite a long time. Does anyone know if there's

[Asterisk-Users] add g.729 license

Hi, I already have one g.729 license on *. Could anyone tell me if I want to add a few more, can I just buy these online and follow their installation instruction, and * will add these addtional licenses? Or this will invalidate my current license? thanks ron