Hello Hadi
In beginning i also face this problem . I solved it by converting to SLN
format.
You also try to convert it to sln format.
this link might help you
http://www.voip-info.org/tiki-index.php?page=Convert+WAV+audio+files+for+use+in+Asterisk
yeah it can :)
On Sat, Sep 26, 2009 at 11:30 AM, hadi motamedi motamed...@gmail.comwrote:
Thank you for your reply . Excuse me , you mean the Asterisk can play SLN
files ? Can you please confirm ?
On Sat, Sep 26, 2009 at 6:57 AM, ABBAS SHAKEEL
shakeel.abbas@gmail.com wrote:
Hello
Dear All
Can you please do me favor and let me know if there is an facility in
Asterisk server that can be used to have remote access to the server ?
Please be informed that we have installed commissioned our Asterisk server
at remote site with DECT telephony service provisioning for our
2009/9/26 hadi motamedi motamed...@gmail.com:
I need to convert the original *.wav sound files (their file attribute is
reported as WAVE audio mono 8000 Hz) into 32 bit 44 KHz for better voice
quality .
That's useless. You can do that of course, but even if you reencode
the file, the quality
Use
Audocity Software
Ravindra kumar
On Sat, Sep 26, 2009 at 11:14 AM, hadi motamedi motamed...@gmail.comwrote:
Dear All
Can you please do me favor and let me know how can I convert *.wav files
into 32 bit 44 KHz ? Please be informed that I have specific sound files in
*.wav format that I
use
Asterisk now software. You can access by IP.
On Sat, Sep 26, 2009 at 2:11 PM, hadi motamedi motamed...@gmail.com wrote:
Dear All
Can you please do me favor and let me know if there is an facility in
Asterisk server that can be used to have remote access to the server ?
Please be informed
hadi motamedi wrote:
Dear All
Can you please do me favor and let me know if there is an facility in
Asterisk server that can be used to have remote access to the server ?
Please be informed that we have installed commissioned our Asterisk
server at remote site with DECT telephony service
Sorry My Question was not very clear.
Asterisk System that is placed some where on local LAN (suppose in office
A) A sip(or any other whose softphone is available) phone Client that is
out side this local network (suppose at office B).
now if I want the asterisk server to be avaiable for
Hello Hadi
While playing files extension is not specified. Remove the extension and
Enjoy
On Sat, Sep 26, 2009 at 3:13 PM, ravi kumar ravi...@gmail.com wrote:
Use
Audocity Software
Ravindra kumar
On Sat, Sep 26, 2009 at 11:14 AM, hadi motamedi motamed...@gmail.comwrote:
Dear All
Can
Thank you for your reply . But I am seeking for PPPoE remote access that
fits my case here . Can you please let me know if there is any solution in
this regard ? (like PPPD)
On Sat, Sep 26, 2009 at 12:16 PM, Michiel van Baak mich...@vanbaak.infowrote:
On 09:41, Sat 26 Sep 09, hadi motamedi
On Sat, 26 Sep 2009, hadi motamedi wrote:
Thank you for your reply . But I am seeking for PPPoE remote access that
fits my case here . Can you please let me know if there is any solution in
this regard ? (like PPPD)
It would be really cool if iaxmodem would actually answer an incoming
Don't put a SIP server behind destination NAT. Just don't.
ABBAS SHAKEEL wrote:
Sorry My Question was not very clear.
Asterisk System that is placed some where on local LAN (suppose in
office A) A sip(or any other whose softphone is available) phone
Client that is out side this
are using some kind of router?
On Sat, Sep 26, 2009 at 8:20 AM, ABBAS SHAKEEL
shakeel.abbas@gmail.comwrote:
Hello
I Recently completed an IVR application with Asterisk.
Now we are moving towards VOIP. Please give a direction how to move
forward.
What i have studied so far
I am
On 09:41, Sat 26 Sep 09, hadi motamedi wrote:
Dear All
Can you please do me favor and let me know if there is an facility in
Asterisk server that can be used to have remote access to the server ?
Please be informed that we have installed commissioned our Asterisk
server
at
Abbas Shakeel wrote:
I Recently completed an IVR application with Asterisk.
Now we are moving towards VOIP. Please give a direction how to move forward.
Depends on what your goals are.
What i have studied so far
I am confused with NAT issues. As i can have many SIP peers on local LAN it
Does anyone have info or starter points on how to take emails from an
external POP3 or IMAP server and cause them to be voiced by Asterisk?
It is our e-mail server, so we can do anything to it. My question is
concept or products required to get asterisk to do the job. Text-to-voice
converter?
Hi
Is it possible to read the full name of the Voice Mail extension from
voicemail.conf using VMauthenticate command ? as everytime I call
VMauthenticate and try to feed in my password - it always returns VM_NAME as
empty string . Alternatively let me know if there is any other way to read
Thank you for your reply . Can you please let me know if there is an
facility to provide PPP over E1 as my Asterisk has ISDN PRI link outwards ?
I mean if any facility inside Asterisk can provide PPP over E1 for remote
access via ISDN PRI link ?
On Sat, Sep 26, 2009 at 11:18 AM, ravi kumar
Hello
I Recently completed an IVR application with Asterisk.
Now we are moving towards VOIP. Please give a direction how to move forward.
What i have studied so far
I am confused with NAT issues. As i can have many SIP peers on local LAN it
works but from internet it donts. We need to do
Thanks Alex
By just avoiding this will solve this problem?
On Sat, Sep 26, 2009 at 9:47 PM, Alex Balashov abalas...@evaristesys.comwrote:
Don't put a SIP server behind destination NAT. Just don't.
ABBAS SHAKEEL wrote:
Sorry My Question was not very clear.
Asterisk System that is
When did that happen? Added to libpri, someone beat me to it.
What you may have seen is my recent minimal implementation as a patch to
1.6.1 - 1.6.2 https://issues.asterisk.org/view.php?id=15604, which is
working, but deprecated.
The task list to get it done properly for trunk Asterisk 1.6.3
Thank you very much for your confirmation . Excuse me , the format needs to
be like the followings ?
exten = s-NOANSWER,n,playback(FR1.sln)
Can you please do me favor and confirm if the above is correct ?
On Sat, Sep 26, 2009 at 7:42 AM, ABBAS SHAKEEL
shakeel.abbas@gmail.comwrote:
A
On 09:41, Sat 26 Sep 09, hadi motamedi wrote:
Dear All
Can you please do me favor and let me know if there is an facility in
Asterisk server that can be used to have remote access to the server ?
Please be informed that we have installed commissioned our Asterisk server
at remote site with
A good way is to give try
On Sat, Sep 26, 2009 at 11:41 AM, ABBAS SHAKEEL shakeel.abbas@gmail.com
wrote:
yeah it can :)
On Sat, Sep 26, 2009 at 11:30 AM, hadi motamedi motamed...@gmail.comwrote:
Thank you for your reply . Excuse me , you mean the Asterisk can play SLN
files ? Can you
On Fri, Sep 25, 2009 at 01:47:04PM -0400, Stephen Brown wrote:
Sure thing, this is if I hang up before it hits voicemail:
This does not include debug-level information . In the CLI, set:
core set debug 5
Then in logger.conf make sure you have a log file that also gets verbose
and debug. If
Brian Camp wrote:
What unit is dtmftimeout measured in?
In samples, 1/8000 of a second each, or 125 us if you prefer.
The sample configuration is provided below. Does it mean...
; The amount of time a DTMF digit with no 'end' marker should be
; allowed to continue (in 'samples', 1/8000 of
Hi,
I've built an Asterisk HA cluster by means of heartbeat and drbd. The
following folders are stored on shared storage and referred to by means of
symbolic links:
/etc/asterisk
/var/lib/asterisk
/usr/lib/asterisk
/var/spool/asterisk
/var/log/asterisk
I was under the impression that phone
If not, it will solve many other problems you would otherwise have.
ABBAS SHAKEEL wrote:
Thanks Alex
By just avoiding this will solve this problem?
On Sat, Sep 26, 2009 at 9:47 PM, Alex Balashov
abalas...@evaristesys.com mailto:abalas...@evaristesys.com wrote:
Don't put a SIP
Contact bindings for AORs/registrations are stored in AstDB, but the
state of a peer as being registered is stored in runtime memory.
I agree that this is kind of silly.
Bart Coninckx wrote:
Hi,
I've built an Asterisk HA cluster by means of heartbeat and drbd. The
following folders are
On Sat, 2009-09-26 at 21:54 +0500, ABBAS SHAKEEL wrote:
Thanks Alex
By just avoiding this will solve this problem?
No,
Just moving the asterisk-server before the firewall won;t do any good.
because in that situation the firewall is in between asterisk and your
LOCAL sip-clients: you
On 26/09/09 19:42, Hans Witvliet wrote:
snip /
What you can do (perhaps not the best solution...) is having one
asterisk server behind your firewall, serving all your local
sip-clients. And another at the other side of the firewall, only for
serving remote clients. And have both systems
On Sat, 26 Sep 2009, Alan Lord (News) wrote:
Hmmm, has anyone tried SIP over a VPN?
We are thinking of testing this but haven't yet...
Al
I have a client with Sonicwall VPNs. Asterisk is at head office on
internal LAN, six external locations all have Linksys 2102 ATAs and
Polycom
I use SIP over OpenVPN incessantly. Works great.
Jeff LaCoursiere wrote:
On Sat, 26 Sep 2009, Alan Lord (News) wrote:
Hmmm, has anyone tried SIP over a VPN?
We are thinking of testing this but haven't yet...
Al
I have a client with Sonicwall VPNs. Asterisk is at head office on
Depending on the latency, wrapping the UDP stream into a TCP-based tunnel
can be good -- if the VPN tunnel occasionally drops a packet, the tunnel
will re-transmit the UDP packet. Of course, if the (one-way) latency is too
high, the re-transmitted payload will arrive outside the jitter buffer and
On Sat, 2009-09-26 at 19:32 +, Jeff LaCoursiere wrote:
On Sat, 26 Sep 2009, Alan Lord (News) wrote:
Hmmm, has anyone tried SIP over a VPN?
We are thinking of testing this but haven't yet...
Al
I have a client with Sonicwall VPNs. Asterisk is at head office on
internal
On Sat, 26 Sep 2009, John A. Sullivan III wrote:
snip
We are using SIP over both IPSec and SSL VPNs very successfully with
access controls in the tunnel ingress via the ISCS network security
management project (http://iscs.sourceforge.net). There are a couple of
issues.
I'm not sure what
On Sat, 2009-09-26 at 22:09 +, Jeff LaCoursiere wrote:
On Sat, 26 Sep 2009, John A. Sullivan III wrote:
snip
We are using SIP over both IPSec and SSL VPNs very successfully with
access controls in the tunnel ingress via the ISCS network security
management project
On Sat, 2009-09-26 at 20:07 +0100, Alan Lord (News) wrote:
On 26/09/09 19:42, Hans Witvliet wrote:
snip /
What you can do (perhaps not the best solution...) is having one
asterisk server behind your firewall, serving all your local
sip-clients. And another at the other side of the
On Sat, 2009-09-26 at 19:32 +, Jeff LaCoursiere wrote:
On Sat, 26 Sep 2009, Alan Lord (News) wrote:
Hmmm, has anyone tried SIP over a VPN?
We are thinking of testing this but haven't yet...
Al
I have a client with Sonicwall VPNs. Asterisk is at head office on
internal
On Sat, 2009-09-26 at 22:09 +, Jeff LaCoursiere wrote:
On Sat, 26 Sep 2009, John A. Sullivan III wrote:
snip
We are using SIP over both IPSec and SSL VPNs very successfully with
access controls in the tunnel ingress via the ISCS network security
management project
Hi,
Is there a way to know for how long an agent is talking on the queue call?
(without keeping a timer myself... just asking asterisk)
thanks,
Gabriel
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 -
In my quest to actually send a fax, I'm now stuck trying to send the
confirm.
First I send the fax:
-- Executing [s...@outbound-fax:2] System(Console/dsp, env echo
-e Channel:DAHDI/g0/12036378447\\nContext:fax-tx\\nExtension:
s\\nPriority: 1\\n
Last week I did a Microsoft VPM from one XP computer to another via Verizon
broadband wireless.
SIP worked ok, but BLF on a Grand Stream 2010 didn't work.
In addition to the VPN the phone was behind a NAT router. The phone was
already set up behind the NAT Router, the only difference was to
Resending is not a waste if the re-transmitted packet can arrive within the
jitter buffer window. Practically speaking, though, since UDP packets are
generally not retransmitted (unless it's within some kind of TCP-based
tunnel), it's a moot point.
Frank
-Original Message-
From:
As with many applications using UDP as a transport, most UDP-based
application-layer VPN schemes (such as OpenVPN) do have some sort of
rudimentary backward acknowledgment and reliability layers implemented
on top of UDP. They're just a lot more lightweight, primitive, and
generally much
u don't change the ${uniquefile} for the second System/Originate
try to add a string to the ${uniquefile} ...
eg
${uniquefile}0
Martin
On Sat, Sep 26, 2009 at 8:05 PM, sean darcy seandar...@gmail.com wrote:
In my quest to actually send a fax, I'm now stuck trying to send the
confirm.
On Sat, Sep 26, 2009 at 12:47 PM, Alex Balashov
abalas...@evaristesys.com wrote:
Don't put a SIP server behind destination NAT. Just don't.
Why not? Mind to explain?
ABBAS SHAKEEL wrote:
Sorry My Question was not very clear.
Asterisk System that is placed some where on local LAN (suppose
Thanks Alex and Hans.
The Discussion was Really helpful.
Now a days i am using VPN for far SIP clients. But I will try what Hans
Suggested and let you know if i found any problem.
On Sun, Sep 27, 2009 at 9:34 AM, C F shma...@gmail.com wrote:
On Sat, Sep 26, 2009 at 12:47 PM, Alex Balashov
Isn't an SSL based tunnel all TCP?
Not in the case of OpenVPN. I'm not sure about the commercial
offerings.
Correct. My recollection is that OpenSSL uses TCP for the setup
and management of the tunnel (e.g. authentication and key
exchange) and uses UDP to carry the actual payload... each
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