Hi all,
I'm actually using Asterisk 1.4.26 and Skype For Asterisk (last
version).
Everything was going fine, but yesterday I've got this messege when
I've tried to restart asterisk (and skypeforasterisk):
[Dec 28 15:18:08] NOTICE[2420] core.cpp: Found license
'S4A-XXX' providing 5
Hi,
Does using a different codec affect the volume of the voice?
i was testing g711 and g729, voice seems to be softer on g729 compared
to g711. sorry not really familiar on how codecs work.
regards
Ron
___
-- Bandwidth and Colocation Provided by
Hi,
I'm having trouble with dialing out on analog lines. Asterisk can't seem to
detect answers.
I have two zap groups.
Group 1 is connected to an external analog PSTN provider. This group seems to
work fine, especially with answeronpolarityswitch.
Group 2 is a group of GSM gateways, ie.
Hi Carlos,
It's simply not possible due to a firmware limitation when general SIP and not
Aastra proprietary mode (not enougth memory capacity).
Don't lack your time by searching a non exisiting solution.
Best Regards,
Francois
-Message d'origine-
De :
Hi Daniel,
Are you using a demo/beta version of Skype for Asterisk ?
If yes, this status is normal, the demo/beta program is terminated from a
while.
I am using the real commercial (not free) and not getting that message.
Best Regards,
Francois
-Message d'origine-
De :
Hi,
no I'm using the real commercial once.
I've installed it in November 2009.
Regards,
daniel
F6HQZ ha scritto:
Hi Daniel,
Are you using a demo/beta version of Skype for Asterisk ?
If yes, this status is normal, the demo/beta program is terminated
from a while.
I am using the
Hello everyone.
I'm trying to set up a SIP DID on a customer, which uses T.38 for faxing. Voice
is working great, but I never configured anything using T.38 in Asterisk so I'm
kinda lost.
On my googling I found out that would be best letting the Linksys SPA8000 (for
those that don't know,
Hello,
I've asterisk (asterisk 1.6.0.6) box with two network interfaces (two
public IP: IP1 and IP2). SIP binds on 0.0.0.0 . Is it possible configure SIP
peer/user to receive/send traffic from one of these IP? For example one
client sends/receives traffic from IP1, other client send/receives
Hi,
I would contact customer support then.
My licenses expire all 2029 (20 years after buying).
regards,
Philipp
Daniel Grotti wrote:
Hi,
no I'm using the real commercial once.
I've installed it in November 2009.
Regards,
daniel
F6HQZ ha scritto:
Hi Daniel,
Are you using a
On Mon, 28 Dec 2009, Tim Nelson wrote:
- Leif Neland le...@neland.dk wrote:
I want some cheap ip-phones with auto-answer, to work as paging system
at dinnertime.
Options, please.
Leif
I've had great luck using the BT201 phones from Grandstream for this
purpose. In fact, this is
If asterisk enters the answered state at any point in the call, then the call
disposition becomes answered.
Thank you and have a nice day,
Anthony Francis
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Szasz Szabolcs
Sent: Tuesday,
Hello,
We're trying to receive G.711 (aLaw) faxes on the asterisk and convert
them to tif. With T.38, we have several issues, so we are trying to use
G.711, since the gateway is located in the same LAN, so there's no
bandwidth/packet-lose issue.
We also use on the same Asterisk Real-Time
Hi,
My license expires at 2029, but this isn't a license problem I think.
I've downloaded my S4A from the following link:
http://downloads.digium.com/pub/telephony/skypeforasterisk/asterisk-1.4/x86-32/skypeforasterisk-1.4_1.0.6-x86_32.tar.gz
As Digium documentation says.
Regards,
daniel
Cyprus VoIP wrote:
My question:
Is the following syntax for disabling T.38 support correct?
vm*CLI -- Executing Set(SIP/Proxy-, t38pt_udptl=no)
vm*CLI -- Executing Set(SIP/Proxy-, SIP_CODEC=aLaw)
vm*CLI -- Executing Answer(SIP/Proxy-, )
I have no idea
Daniel Grotti wrote:
Hi,
My license expires at 2029, but this isn't a license problem I think.
I've downloaded my S4A from the following link:
http://downloads.digium.com/pub/telephony/skypeforasterisk/asterisk-1.4/x86-32/skypeforasterisk-1.4_1.0.6-x86_32.tar.gz
As Digium documentation
Hi men,
I am sure this is the demo version, not the correct actual licensed one.
Fro mthe CLI, enter that :
fax show version
My Asterisk reply that :
Fax For Asterisk Components:
Applications: 1.6.1_1.0.15
Digium Fax T.38 Driver: 1.6.1_1.0.11 (optimized for c3_2_32)
Leif Neland wrote:
I can't believe anyone would use RJ-11 any more. You can multi-purpose
RJ-45 jacks to work with POTS lines. Run everything down to a central
panel and send pots over the jacks that you need to. That way if you
decide you need/want to go IP in the future, you're all set.
I have no idea where you got the idea that such a thing is possible...
it's not. sip.conf settings for SIP endpoints are not channel variables,
and cannot be modified from the dialplan unless the CHANNEL() dialplan
function has been specifically extended to support them.
I was actually HOPING
we tested asterisk 1.6.2.0, found that
when call from one sip_channel to another sip_channel ,
--
exten = _X.,1,Noop()
exten = _X.,n,Dial(SIP/${EXTEN},50,TtgM)
--
Hi Sucan,
A2Billing doesn't have a mailing list but you may ask your specific question
on A2billing Forum or maybe even here. This may be of intrest to you if you
have an installation question:
A2Billing automated install script :
http://a2billing2asterisk.googlepages.com
You should set the ddwhome variable with the Set function or declare it
on the global context. Try the Dial command with the dial string
directly, before using the variable.
Fro debugging purposes you should set debug and verbose at least to 10
and check the logs.
Regards,
Juan
James A. Shigley
A2billing forum has a lot of information and questions are answered
very fast. Try searching on the forum before posting, cause the answer
may be there already.
forum.asterisk2billing.org/
Regards,
Juan
Bruce Nik wrote:
Hi Sucan,
A2Billing doesn't have a mailing list but you may ask
It depends on what flavor of Asterisk and trunks (SIP/Zapata/DAHDI) you are
using.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joe Freeman
Sent: Monday, December 28, 2009 8:14 PM
To:
Now, referring to the error above, I see (in voip-info.org) that
t38passthrough is an R/O variable and not an R/W, but in any case, I got
0 as a result, so it should have been OK, and it's not, as ReceiveFAX
still sends a T.38 reINVITE. If I can't modify it, what should I do?
For the
Cyprus VoIP wrote:
Now, referring to the error above, I see (in voip-info.org) that
t38passthrough is an R/O variable and not an R/W, but in any case, I got
0 as a result, so it should have been OK, and it's not, as ReceiveFAX
still sends a T.38 reINVITE. If I can't modify it, what should
At the moment, 1.6.0.20 realtime with Dahdi 2.2, TDM is a TE420, but
that won't be customer facing.
Thanks-
Joe
Danny Nicholas wrote:
It depends on what flavor of Asterisk and trunks (SIP/Zapata/DAHDI) you are
using.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Hi,
I'm looking for an application to show all the calls received/made
including (this is very important!) transferred calls because I need to
track all the time spent on the phone by all my employees.
There is a list here but they are too many to try them all:
Daniel-
no I'm using the real commercial once.
I've installed it in November 2009.
Did you have the demo version installed before the commercial version? I.e.
install the commercial over the top of
the demo version?
-Jeff
F6HQZ ha scritto:
Hi Daniel,
Are you using a demo/beta version
On Monday 28 December 2009 23:49:13 Tilghman Lesher wrote:
On Monday 28 December 2009 18:09:15 JR Richardson wrote:
I turned on console debug to see the actual mysql queries and to my
surprise and concern, I see every query for an extension priority
repeated 3 or more times prior to
On 12/29/2009 1:01 AM, Jeremy Kister wrote:
e.g., in the first call, below, the channel name is
SIP/vgw1-0075 -- the second call (on the same FXO port after a
soft hangup on the CLI) is SIP/vgw1-0077
How can I extract this information in the dialplan so that I can use
the
Most of the asterisk-dev members read this discussion (In My Experience).
${EXTEN} in the case you state would be SIP/vgw1-0075.
Perhaps this link would be helpful
http://www.voip-info.org/wiki/view/Asterisk+Detailed+Variable+List
-Original Message-
From:
On 12/29/2009 3:23 PM, Danny Nicholas wrote:
Most of the asterisk-dev members read this discussion (In My Experience).
${EXTEN} in the case you state would be SIP/vgw1-0075.
Perhaps this link would be helpful
http://www.voip-info.org/wiki/view/Asterisk+Detailed+Variable+List
Thanks for
You could do a System(core show channels) and grep out 911 and kill
everything else; probably easier as an AGI call that a dialplan function,
but both can be done.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf
On 12/29/2009 3:54 PM, Danny Nicholas wrote:
You could do a System(core show channels) and grep out 911 and kill
everything else; probably easier as an AGI call that a dialplan function,
but both can be done.
great idea; thanks!
--
Jeremy Kister
http://jeremy.kister.net./
Un-top-posting...
On 12/29/2009 1:01 AM, Jeremy Kister wrote:
e.g., in the first call, below, the channel name is SIP/vgw1-0075
-- the second call (on the same FXO port after a soft hangup on the
CLI) is SIP/vgw1-0077
How can I extract this information in the dialplan so that I
I am running Asterisk V 1.4.22
Twice during the last two days the Context Switches on our box has gone from
about 7K to 80K in 2.5 hours. The load average would spike to 17, drop to
0.35 then spike again.
When connecting to the console 'core show channels' will list the channels
but not total
On 12/29/2009 3:54 PM, Danny Nicholas wrote:
You could do a System(core show channels) and grep out 911 and kill
everything else; probably easier as an AGI call that a dialplan function,
but both can be done.
my end result just feels ugly. the loop is due to the fact that I
have more than
Before I start I am a Panasonic certified dealer AND I have installed
over 100 Asterisk systems that are in production.
That said for your application use Panasonic, DONT use Asterisk.
Use the Panasonic KX-TDA50G. Supports up to around 50 ports.
In addition to Analog and their proprietary Digital
On Mon, Dec 28, 2009 at 5:42 PM, John Novack
jnov...@stromberg-carlson.org wrote:
Rick Huebner wrote:
My brother-in-law is finishing up his McMansion and I've done all of the
low voltage wiring and am starting the trimout. We are batting around
what to do for a phone system and I'm torn
On Mon, Dec 28, 2009 at 11:45 PM, Doug d...@natel.net wrote:
At 16:13 12/28/2009, Rick Huebner wrote:
My brother-in-law is finishing up his McMansion and I've done all of the
low voltage wiring and am starting the trimout. We are batting around
what to do for a phone system and I'm torn
On Tue, Dec 29, 2009 at 11:30:21PM -0500, C F wrote:
Before I start I am a Panasonic certified dealer AND I have installed
over 100 Asterisk systems that are in production.
That said for your application use Panasonic, DONT use Asterisk.
Use the Panasonic KX-TDA50G. Supports up to around 50
Hi all,
I'm trying to use a wrt54gl router running dd-wrt as a provisioning server for
a remote installation.
I've got dhcp working and I have provisioning files ready to go. I understand
that I need to set bootp option 66 to point to the tftp/ftp/http server. In
fact, I have this working
Hi Sir,
We have integrated Skype with Asterisk (skype user id:-
rexesbposolutions). Each call which is coming to skype account is
getting transfered to Asterisk Queue. It has following two cases:
case 1: When we call from normal skype account to skype account
(rexesbposolutions), everything is
On Wed, Sep 9, 2009 at 4:02 AM, Jeff LaCoursiere j...@jeff.net wrote:
On Wed, 9 Sep 2009, hadi motamedi wrote:
Thank you for your message . But I tried to find it on my server , as the
followings :
#find / -name sip.cfg -print
But it didn't return any result . Can you please let me
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