Re: [asterisk-users] Early media and IAX2

2010-09-04 Thread Russ Dill
On Tue, Aug 31, 2010 at 8:11 PM, Matt Riddell li...@venturevoip.com wrote: On 28/08/10 10:18 AM, Russ Dill wrote: My IAX2 trunk provider, Teliax, seems to be forcing early media. Early media is cool and all, but my Asterisk install doesn't seem to be fully supporting it. My initial setting was

Re: [asterisk-users] Wanted: UK-specific hardware recommendations (FXOand FXS)

2010-09-04 Thread --[ UxBoD ]--
- Original Message - Roger Burton West wrote: I want to hook one of them to the PSTN. Given that I am in the UK, what is a reasonably easily-available device to provide an FXO interface from a Linux box, with a minimum of faffing around with drivers? Just one line is needed,

[asterisk-users] Snom phones recommended firmware

2010-09-04 Thread John Taylor
We're using firmware 7.3.30 on an installation of Snom 300 phones. Should we stick with it, or do the newer firmwares have better support for Asterisk? Thanks John -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] Freepbx + Asterisk problem - NEED HELP

2010-09-04 Thread Mehmet Kuzulugil
Sorry for unnecessary requests. I have read the book and it is related to the old version. Now I am studying the reference book included in the sources... Asterisk Reference Information Version 1.6.2.11 # make pdf and /usr/src/asterisk-1.6.2.6/doc/tex/asterisk.pdf is there for pdf people. On 2

Re: [asterisk-users] Snom phones recommended firmware

2010-09-04 Thread Philipp von Klitzing
Hi! We're using firmware 7.3.30 on an installation of Snom 300 phones. Should we stick with it, or do the newer firmwares have better support for Asterisk? So what is it that you are missing that firmware 8 does offer? 7.3.30 is rather stable and therefore a good choice. Actually I would

[asterisk-users] Manuplating Queue

2010-09-04 Thread Timothy Smith
Hi, I am implimenting a solution for a radio station where by calls are first received by an attendant, who interviews the caller and then places the call in a queue along with some information about the caller. The radio presenter can then choose which call to pick up depending on those in the

Re: [asterisk-users] Manuplating Queue

2010-09-04 Thread Hoggins!
Hello, We use call parking for this feature. It might not be the best solution, but it works quite well. We tweaked the parking values a little bit so that the parked callers don't timeout too quickly. The receptionist fills up a dynamic list that the presenter can consult, and knows which

[asterisk-users] fast busy out?

2010-09-04 Thread Thomas Perron
why does this not work? i simply want to hear the recorded message exten = s,1,Answer() ;exten = s,n,Record(zipcodegutter1.gsm) ;zcg1 exten = s,n,Playback(zipcodegutter1) exten = s,n,Dial(SIP/c01s/159,120,A,(demo-thanks)) --

[asterisk-users] Global Outage?

2010-09-04 Thread Matt Desbiens
Is anyone else using Vitelity right now and having an issue with a global outage of sorts? Potral/WWW arent accessible and it would appear through monitoring that the outbound is flapipng like mad. The outbound can be rerouted, I know, but inbound is a huge problem right now. [Sep 4 10:26:13]

Re: [asterisk-users] Manuplating Queue

2010-09-04 Thread Timothy Smith
Thank you Hoggins! I am going to try it out and let you know. Regards, Tim On Sat, Sep 4, 2010 at 4:02 PM, Hoggins! fucks...@wheres5.com wrote: Hello, We use call parking for this feature. It might not be the best solution, but it works quite well. We tweaked the parking values a little bit

[asterisk-users] Vitelity offline?

2010-09-04 Thread Roger Marquis
Vitelity seems to be offline to both IP and voice traffic. Is there any place to find out what their status is? Roger Marquis -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us

Re: [asterisk-users] fast busy out?

2010-09-04 Thread Ondrej Škopek
I assume thtat you've already recorded the message, and the out commented the Record app.. try adding .gsm to the playback, to ensure that * doesn't look for other formats.. and according to http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Record you should use file:format in the Record

Re: [asterisk-users] fast busy out?

2010-09-04 Thread Ondrej Škopek
* then 2010/9/4 Ondrej Škopek skopekond...@gmail.com I assume thtat you've already recorded the message, and the out commented the Record app.. try adding .gsm to the playback, to ensure that * doesn't look for other formats.. and according to

Re: [asterisk-users] fast busy out?

2010-09-04 Thread Anton Raharja
On 09/04/2010 08:40 PM, Thomas Perron wrote: why does this not work? i simply want to hear the recorded message exten = s,1,Answer() ;exten = s,n,Record(zipcodegutter1.gsm) ;zcg1 exten = s,n,Playback(zipcodegutter1) exten =

Re: [asterisk-users] Vitelity offline?

2010-09-04 Thread Kyle Kienapfel
Looks like they have twitter. Its good that you mentioned them in the subject unlike the guy who wrote an hour and a half ago with subject Global Outage? http://twitter.com/vitelity http://twitter.com/vitelityWe are currently experiencing network difficulty on Vitelity's core router. We are

Re: [asterisk-users] Vitelity offline?

2010-09-04 Thread Matt Desbiens
Not that I'm aware of short of our direct contact. It would appear from the traceroutes that i've done this morning, that this appears to be a big part of the issue Tracing route to portal.vitelity.net [64.74.178.100] 1074 ms74 ms75 ms

Re: [asterisk-users] fast busy out?

2010-09-04 Thread Ondrej Škopek
no I am not sorry, and please reply to this list, and not to me directly.. On Sat, Sep 4, 2010 at 6:16 PM, Thomas Perron thomas.per...@gmail.comwrote: thank you for your note on the Asterisk users group list Are you in Scandanavia somewhere? Cheers Tom -- -- Ondrej Škopek --

Re: [asterisk-users] Vitelity offline?

2010-09-04 Thread Roderick A. Anderson
Roger Marquis wrote: Vitelity seems to be offline to both IP and voice traffic. Is there any place to find out what their status is? 09:30 PDT, Inland Northwest (Spokane, WA; Hayden, ID). I went directly to their website -- http://www.vitelity.net. Then called my business number and got

Re: [asterisk-users] Vitelity offline?

2010-09-04 Thread Matt Desbiens
Just a heads up. It would appear that Vitelity is back online and processing calls and the portal is back up and running. On Sat, Sep 4, 2010 at 12:14 PM, Matt Desbiens desbie...@gmail.com wrote: Not that I'm aware of short of our direct contact. It would appear from the traceroutes that

Re: [asterisk-users] fast busy out?

2010-09-04 Thread Thomas Perron
Thank for your the tip Ondrej. Here is what worked on my CentOS box. exten = s,1,Answer() exten = s,n,Wait(2) exten = s,n,Record(zipcodegutter%d:gsm) exten = s,n,Wait(2) exten = s,n,Playback(${RECORDED_FILE}) exten = s,n,Wait(2) exten = s,n,Hangup() 2010/9/4 Ondrej Škopek

Re: [asterisk-users] How to finish an AGI

2010-09-04 Thread Edwin Quijada
IMHO, is more easy in Perl that in dialplan but if for you work .. *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-Soporte PostgreSQL *-www.jqmicrosistemas.com *-809-849-8087

[asterisk-users] Possible malformed G729B - SID (VAD/DTX) frames from carrier endpoint ?

2010-09-04 Thread Vikram Ragukumar
Hello, We are in the process of debugging a voice quality issue for a client of ours that is a VoIP services provider. The client uses a softphone that runs on a pjsip stack. When placing a call using the softphone, it negotiates the use of G729 codec with the remote endpoint (ptime = 20ms). The

Re: [asterisk-users] How to tell if there is a transfer from CDR?

2010-09-04 Thread C F
Last time I analyzed this (I believe back in 1.2) there was no way of telling. However a blind transfered call would generate 2 CDR recoreds: 1. For the part of the call with the transferrer and transfered. 2. For the part of the call with the transferee and transfered. The call duration for the

Re: [asterisk-users] Possible malformed G729B - SID (VAD/DTX) frames from carrier endpoint ?

2010-09-04 Thread Steve Underwood
On 09/05/2010 04:08 AM, Vikram Ragukumar wrote: Hello, We are in the process of debugging a voice quality issue for a client of ours that is a VoIP services provider. The client uses a softphone that runs on a pjsip stack. When placing a call using the softphone, it negotiates the use of