[asterisk-users] SIP to IAX2 with delayed echo

2008-11-20 Thread c james
A Polycom 550 and a IAX client (Mozphone and ZoIPer were used) are having a conversation. Call quality is reported as good except for an echo with a 3 second delay. Most of my searches are saying echo happens only on the PSTN piece, but there isn't one here. Can someone point me in the right

Re: [asterisk-users] SIP to IAX2 with delayed echo

2008-11-20 Thread c james
sure there aren't any system resource constraints such as high CPU usage or memory usage... :-) Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 - c james [EMAIL PROTECTED] wrote: A Polycom 550 and a IAX client (Mozphone and ZoIPer were used) are having

Re: [asterisk-users] SIP to IAX2 with delayed echo

2008-11-20 Thread c james
Steve Totaro wrote: Just for sh1t$ and giggles, try sip to sip and drop the IAX piece. IAX2 is not all it is cracked up to be. Also, do a ping to see latency, 200ms is pretty much my standard. Coming from outside the network, setting up for a couple rounds of NATting isn't going to work

Re: [asterisk-users] SIP to IAX2 with delayed echo

2008-11-20 Thread c james
Drew Gibson wrote: c james wrote: A Polycom 550 and a IAX client (Mozphone and ZoIPer were used) are having a conversation. Call quality is reported as good except for an echo with a 3 second delay. Most of my searches are saying echo happens only on the PSTN piece, but there isn't one

Re: [asterisk-users] SIP to IAX2 with delayed echo

2008-11-20 Thread c james
Steve Totaro wrote: On Thu, Nov 20, 2008 at 1:13 PM, c james [EMAIL PROTECTED] wrote: Steve Totaro wrote: Just for sh1t$ and giggles, try sip to sip and drop the IAX piece. IAX2 is not all it is cracked up to be. Also, do a ping to see latency, 200ms is pretty much my standard. Coming

Re: [asterisk-users] Voicemail IMAP ./configure error

2008-11-13 Thread c james
Mark Michelson wrote: c james wrote: Mark Michelson wrote: c james wrote: Mark Michelson wrote: c james wrote: I have c-client installed on a 64bit system running Gentoo. I am trying to run configure so I can test the IMAP voicemail functionality. But asterisk-1.4.22 # ./configure

Re: [asterisk-users] Voicemail IMAP ./configure error

2008-11-11 Thread c james
Mark Michelson wrote: c james wrote: Mark Michelson wrote: c james wrote: I have c-client installed on a 64bit system running Gentoo. I am trying to run configure so I can test the IMAP voicemail functionality. But asterisk-1.4.22 # ./configure --with-imap=/usr/include/imap just gives me

[asterisk-users] Voicemail IMAP ./configure error

2008-11-10 Thread c james
I have c-client installed on a 64bit system running Gentoo. I am trying to run configure so I can test the IMAP voicemail functionality. But asterisk-1.4.22 # ./configure --with-imap=/usr/include/imap just gives me the following error. checking for gnutls_bye in -lgnutls... no checking for UW

Re: [asterisk-users] Voicemail IMAP ./configure error

2008-11-10 Thread c james
Mark Michelson wrote: c james wrote: I have c-client installed on a 64bit system running Gentoo. I am trying to run configure so I can test the IMAP voicemail functionality. But asterisk-1.4.22 # ./configure --with-imap=/usr/include/imap just gives me the following error. checking

Re: [asterisk-users] Voicemail on PRI

2008-08-07 Thread c james
Yann Derichard wrote: Hi, I am trying to install a Voicemail on PRI after a redirection on an away or a busy (a normal call which is redirected to voicemail in fact) but I can't find the function in Asterisk which allow me using the phone number of the callee (because I have only the

[asterisk-users] Magnetic door locks

2008-07-17 Thread c james
I have an opportunity to interface asterisk with a security system to open their magnetic door locks. The security system needs a dry contact close upon activation to signal the door. Has anyone done this before? ___ -- Bandwidth and Colocation

Re: [asterisk-users] Astricon question: four or five tracks?

2008-06-13 Thread c james
John Todd wrote: Is it too much to have 5 talk tracks at Astricon? Do the extra tracks. With a recording to review at night or online that nullifies the problem of picking. Really, with most presentations having slides all you need is fair video but excellent audio. How quick could this be

[asterisk-users] Really destroying SIP dialog

2008-06-12 Thread c james
I am trying to work in the console, figuring why it exits, but about 75% is always taken up with Really destroying SIP dialog '' Method: OPTIONS Can anyone point me where I can stop this without turning down the debugging/verbose on the entire console.

Re: [asterisk-users] redfone fonebridge2

2008-06-10 Thread c james
Bill Michaelson wrote: I'm looking for reports of recent experience with redfone fonebridge2 (with echo can) TDMoE gizmos. Anybody? Good? Bad? We use it and it works without any problems. Tech support was helpful, documentation was not. Thumbs up.

Re: [asterisk-users] Start call from asterisk

2007-10-26 Thread c james
Suity Zsolt wrote: Can you give me a hint, how can I start a call from asterisk with some (php, bash, etc) script? I need to start two calls and bride it together. http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out ___

[asterisk-users] Astricon 2007

2007-07-29 Thread c james
Saw the website but can't find a schedule or even an email address to contact someone. Anyone know more about Astricon? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update