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
- 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,
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
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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
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
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
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
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))
--
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]
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
Vitelity seems to be offline to both IP and voice traffic. Is there any
place to find out what their status is?
Roger Marquis
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New to Asterisk? Join us
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
* 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
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 =
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
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
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
--
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
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
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
IMHO, is more easy in Perl that in dialplan but if for you work ..
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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
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
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
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