Thanks for the confirmation!
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I just purchased credits with voipbuster to use it with asterisk. I have not been having any problems, however I'm curious if they have a 1 hour limit on calls?? I have used it to join bridge calls for work, and i got dropped at the 1 hour mark. has anyone else experienced this? its not really
Have you found any information yet about this? I am looking for good and affordable phones that can use DNS myself, but not for failover, simply for ease of use by some non-computer savvy family members. So far, I am afraid I'm going to be limited to USB/software phones. I would greatly appreciate
I had the same problem, and have beenworking with Sherwood. he suggested that I try dtmf=inband, which has been working great for me.
On 8/27/05, John covici [EMAIL PROTECTED] wrote:
I am using viatalk as my voip provider and they use dtmf=rfc2833, butasterisk is not seeing any of the dtmf. I am
On 8/27/05, John covici [EMAIL PROTECTED] wrote:
I was thinking of that, but I was hoping I would not have to do thatbecause I had a previous provider and all kinds of trouble going the
other way -- how's your outbound dtmf working?
outbound so far, changing to inband made my company's VM system
Is anyone else with ViaTalk experiencing an outage right now? My DID
has been down since 5AM (8/20). Asterisk is unable to re-register or
connect for outbound calls. I have also tried calling support and
their number gives a fast busy.
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command but I couldn't do sip
show peer num and couldn't show channels, etc This is very
disconcerting
We've overall had little to now major issues with it running on our
switch
--Original Message-
-From: [EMAIL PROTECTED]
-[mailto:[EMAIL PROTECTED] On Behalf Of
-Justin
I too have been having inbound dtmf problems with VP Connect using
iax2 for inbound. When I switched to sip, and added the
relaxdtmf=yes, all 10 inbound test calls I did seemed to work fine for
dtmf. I'm going to leave my config set up to use sip for inbound VP
Connect calls for a while and see
Hmm. How did you do that because I went to the support site, filed a
'question' and i've had no response..
On 4/26/05, Spencer Nassar [EMAIL PROTECTED] wrote:
I saw the same thing. Filed a support ticket with Voice Pulse Connect
and they cleared it up within an hour.
SN
On Apr 22,
so how do we get this fixed, its happing to my one and only DID as well...
On 4/22/05, Me [EMAIL PROTECTED] wrote:
I had the same problem with another provider whom I got no response from as
usual..
We had 5 or 6 numbers that worked fine and one that just quit sending DTMF.
-
that seems to have done it!! _ALERT_INFO works great for the current
CVS, although it didn't seem to work in 1.0.5.. Thanks for the help!!
On Sat, 19 Mar 2005 16:52:26 +1300, Matt Riddell
[EMAIL PROTECTED] wrote:
You could try _ALERT_INFO instead of ALERT_INFO...
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Cheers,
Matt
i would recommend renaming or deleting /usr/lib/asterisk/modules, i
beat my head on the wall for an hour or so with this when upgrading
asterisk and trying to downrev back to 1.0.5 when i was having
problems with the latest cvs (which turned out to be a simple config
mod). so if you upgrade and
I would appreciate any help in locating the latest firmware for the
cisco 12sp+ phone. I currently have D2.04. I've searched the digium
lists, google and yahoo with out any luck so far.
Thank you for your time!
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I've done the unspeakable. I took a working setup of 1.0.5 and
upgraded with the latest cvs. with 1.0.5 distinctive ring worked
great. with the latest cvs, it doesn't seem to work with my sipura
2000 (the only thing i have to test it with). I can see in console
that its sending the info, but
I did figure out how to get 1.0.5 back up and running.
/usr/lib/asterisk/modules had extra modules from cvs that 1.0.5 didn't
like, and since modules.conf has autoload=yes, i was breaking myself..
thought I would share in case some other noob has the same problem.
I would still like to know if
i'm trying to set up a 12sp+ and here's what I'm getting in asterisk console:
-- Starting Skinny session from 192.168.0.181
Device SEP00308062B5CD is attempting to register
Mar 16 19:06:35 ERROR[26875]: chan_skinny.c:1856 handle_message:
Rejecting Device SEP00308062B5CD: Device not found
Mar
OK. maybe i needed to fully stop/start asterisk...now i get
registered, but the phone says Program Update on the screen?
-- Device 'SEP00308062B5CD' successfuly registered
Requesting capabilities
Version Request
Received CapabilitiesRes
Mar 16 19:39:21 WARNING[27156]: chan_skinny.c:2301
well, answered my own question again: changed the version string as
you see below, comes right up... I would like to get a newer firmware
though!
; Typical config for 12SP+
[SEP00308062B5CD]
device=SEP00308062B5CD
version=P002D204; Thanks critch
;version=P002F204 ; Thanks critch
**# is the sequence to get into programing mode where you can setup
the network info. when in this mode use * for . and use # to
finish that setting (enter key if you will)
goto voip-info.org and do a search on 12sp for more details. i got my
12sp running in a couple hours after doing some
I am having problem with voipjet and g.711 (ulaw) as well. I tried
ilbc with no luck. basically my outbound call connects, i can hear
them talk, but they can't hear me.
i am not getting errors in console with either ulaw or ilbc, just no
audio to the called party.
it worked great yesterday,
well now i'm confused, because its working again. so i'm not sure if
it really had a problem or not.
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there might be an easier way, but i changed the asterisk source code
before i compiled to reference the zap devices under /dev in their new
place. i think everything you need to change is under the apps dir in
the source tree
On Wed, 2 Mar 2005 20:02:33 +0500, Rizwan Chaudhry [EMAIL PROTECTED]
I'm really trying to understand this. I have a Sipura 2000, brother
MFC, and SpanDSP set up on asterisk.
because asterisk softfax was not working well, i set up faxes to goto
line2 on the sipura. this is working fine, i have only tested a few
short faxes, but already my completion rate was 100%
Sorry for the cross post, but I'm still trying to find a Seoul DID. I
received an email from LiveVoip.com that said they have service in
South Korea, but when I called them they said they didn't offer such
service.
If you have the capability to offer a DID please let me know what your
pricing
$5 DID? with who if you don't mind me asking?
On Fri, 18 Feb 2005 13:56:12 -0600, Jay Milk [EMAIL PROTECTED] wrote:
Yes, it's doable, had this running for several months here. However,
you'll need to get a softphone for $10/month from them, and they'll
provide the sip-credentials on their
?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Justin Richards
Sent: Wednesday, February 16, 2005 4:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] fax with asterisk
I'm getting a lot
Hmm I don't know what to think now. I just tested again, using a
Cannon Multi-function laser fax/printer (G3) I got very few fake imput
errors, and the fax went through almost perfect. (PSTN to *)
Using a Brother MFC 3820CN that is attached via IAXy on private LAN, I
get tons of fake imput's
I'll give it a shot and see how it works!!
On Thu, 17 Feb 2005 10:37:16 -0700, Tony da Costa
[EMAIL PROTECTED] wrote:
For anyone playing around with IAXy(S100i) devices, I am making the
following available:
Windows IAXy Provision v1.00
This is a from-the-ground-up development of a means of
Have you tried canreinvite=no? I have a like config and my phones on
the private lan work fine in both directions. i don't even have the
canreinvite directive in my config...
On Thu, 17 Feb 2005 16:54:49 -0600, Matthew Boehm [EMAIL PROTECTED] wrote:
Can anyone give an example of the
Worked for me, actually even better than before with the linux
provisioning app. for some reason, i could never get the alternate
server setting to work before (I want to put my real IP as the first,
then my internal IP as the alternate so I can travel with it easier).
maybe i had done something
I'm getting a lot of this too :-( my fax stuff worked great under 1.0
but after upgrading to 1.0.5 i've been broken..
Fax3Decode2D: Warning, (FakeInput): Premature EOL at scanline 496 (got
912, expected 1728).
Fax3Decode2D: (FakeInput): Bad code word at scanline 497 (x 470).
Fax3Decode2D:
working just how I want it to..
On Fri, 14 Jan 2005 08:04:57 -0700, Ken Godee [EMAIL PROTECTED] wrote:
Justin Richards wrote:
I have not used any M$ products, but it works with shoutcast like this:
default = quietmp3:/var/lib/asterisk/mohmp3-empty,http://host.com:8000/
basically, create
On Wed, 5 Jan 2005 16:50:12 -0700 (MST), Dan Adams
[EMAIL PROTECTED] wrote:
I was wondering, does anyone know if it is possible to have a stream of
audio coming from a Microsoft compressed audio stream fed to the caller if
they are placed on hold and if so how might this be done?
I have not
First, please forgive me if this is a total newbie question, I've only
just begun to scratch the surface of asterisk.
I currently have a dialplan set up to let me dial a specific
extension, authenticate the user, then have * dial a hard
coded/programmed overseas number. What I would like to do
, Justin Richards wrote:
I currently have a dialplan set up to let me dial a specific
extension, authenticate the user, then have * dial a hard
coded/programmed overseas number. What I would like to do is set up
my dialplan to have an extension that offers up an outbound dialtone
allowing
On Tue, 4 Jan 2005 09:04:37 -0800, Dallas Jones
[EMAIL PROTECTED] wrote:
Can anyone out there explain how to successfully integrate my Voice Pulse
Connect account into this config so the clients can make outbound calls? I
tried using the sample configs provided by VoicePulse (after making
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