Hi Ronald
I took a bit of interest in your problem as I'm going to have to be
doing the same thing in a few weeks.
oenais is in the yum repositories so you can install from there if using
redhat/centos based OS
It is also in apt repositories if you're using a debian based OS
Let me know how
iptables -L -n | grep icmp gives you the same on both machines?
Is it possible that the other public IP is behind a main firewall,
provided by your ISP? I know our hosting provider has this. They filter all
traffic through their main router, and after that locally with iptables.
On Tue, Jan 3,
I already tried what u posted didnt work
but thanx for the reply :)
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Sammy Govind
[govoi...@gmail.com]
Sent: Wednesday, January 04, 2012 11:32 PM
Hi,
This is not strictly an asterisk questions, but... ive got a client with an old
digital pbx phone systems connected to an isdn30e line.
I've been shown a sip gateway that can connect to asterisk on one side, and
also has an ISDN30e socket that the old phone system can connect to. But it's
All screwing up with Asterisk is supposed to be documented in the relevant
UPGRADE*.txt files. Have you checked them?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph
Sent: Thursday, January 05, 2012
On Friday 06 January 2012, Dan Journo wrote:
Hi,
This is not strictly an asterisk questions, but... ive got a client with an
old digital pbx phone systems connected to an isdn30e line.
I've been shown a sip gateway that can connect to asterisk on one side, and
also has an ISDN30e socket
David Backeberg dbackeb...@gmail.com writes:
Thanks for clearing that up. I was getting all excited that I could
flash the PAP2T; I've always used regular voice tones over SIP with
the PAP2Ts.
SPA-2102 supports T.38. If you ignore the WAN-port, it is practically a
PAP2T. The only time you
On Fri, Jan 6, 2012 at 8:16 AM, Eric Wieling ewiel...@nyigc.com wrote:
All screwing up with Asterisk is supposed to be documented in the
relevant UPGRADE*.txt files. Have you checked them?
is supposed to be but does NOT happen. There are many examples of
regressions introduced after many
Correct, but the changes to the insecure= item is clearly documented in the
UPGRADE-1.6.txt file
* SIP: The old insecure options, deprecated in 1.4, have been removed.
insecure=very should be changed to insecure=port,invite
insecure=yes should be changed to insecure=port
Be aware that some
Yes, I already declared 'use lib
/home/asterisk/lib/lib64/perl5/5.8.8/x86_64-linux-thread-multi/;' in my
AGI. When I execute the script as a user Asterisk, i.e. perl -wc test.pl in
return I'm getting OK and no error messages and script is running fine when
I try to run in shell.
Even though I
Here's one more thing to try - do a su - nobody then run the agi - once it
runs under su - nobody there's no reason it should run in Asterisk.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ahmed Munir
Sent: Friday, January 06, 2012
El día 6 de enero de 2012 06:00, Roland aster...@rolandow.com escribió:
iptables -L -n | grep icmp gives you the same on both machines?
Yes.
ACCEPT icmp -- 0.0.0.0/00.0.0.0/0
Is it possible that the other public IP is behind a main firewall,
the ISP say to dont have any
I found this on another post and cleaned it up - might help
#!/usr/local/bin/perl
use strict;
use IO::Socket;
my $target = shift; #192.168.0.255;
my $target_port = 4569;
socket(PING, PF_INET, SOCK_DGRAM, getprotobyname(udp));
# Build Packet ...
# Names from ethereal filter of registration
Hi
I'm experimenting with using a port other than 5060 on one of our
asterisk servers.
Does anyone know how to change the target port on a Snom phone.
I have tried adding :new port number to the end of the registrar but
this doesn't work.
Advanced - SIP/RTP - Network identity(port) is something
Add a BEGIN {...} block prior to the use statements and in there redirect
STDERR to a file. This will aloow you to capture compilation errors You
should also add some debugging statements at key points in the script.
Then run the script and review the file to see what errors it generated.
--
Interestingly enough, they just list port 5060 in the Asterisk
interoperability guide:
http://wiki.snom.com/Interoperability/PBX/Asterisk
Could that mean it is a fixed setting? (crappy)
*José Pablo Méndez
*
On Fri, Jan 6, 2012 at 10:17 AM, Ishfaq Malik i...@pack-net.co.uk wrote:
Hi
Hi
I want to play streaming music from an internet lp like
http://114.23.245.234:9000 . there will be maximum of 75 callers at a time.
Is combination of SetMusicon Hold and WaitMusiconHold command the best
option ? All callers would be calling on PRI lines. The streaming source
would be from
Hello,
Reading through the Wiki:
Asterisk supports the ability to write dialplan instructions in the Lua
programming language. This method can be used as an alternative to or in
combination with extensions.conf and/or AEL. PBX lua allows users to use
the full power of lua to develop telephony
Kevin,
I rolled back clean my virtual asterisk testbed and instead of libspandsp2
and libspandsp-dev, I installed libtiff4 - libtiff4-dev and
downloaded/compiled spandsp 0.0.6, then I fixed the libraries with
echo /usr/local/lib /etc/ld.so.conf.d/spandsp.conf
cuz it wasn't able to load the
On 01/06/2012 12:54 PM, José Pablo Méndez Soto wrote:
Kevin,
I rolled back clean my virtual asterisk testbed and instead of
libspandsp2 and libspandsp-dev, I installed libtiff4 - libtiff4-dev and
downloaded/compiled spandsp 0.0.6, then I fixed the libraries with
echo /usr/local/lib
Ah ok,
I got the incredible idea to go look into the make menuselect for a
res_fax_spandsp option after reading this:
http://lists.digium.com/pipermail/asterisk-dev/2010-September/046344.html
I found it in 1.8, now you say it doesn't come with gateway support.
Thanks for the clarification
On 01/06/2012 01:02 PM, José Pablo Méndez Soto wrote:
Ah ok,
I got the incredible idea to go look into the make menuselect for a
res_fax_spandsp option after reading this:
http://lists.digium.com/pipermail/asterisk-dev/2010-September/046344.html
I found it in 1.8, now you say it doesn't come
On Fri, 6 Jan 2012, José Pablo Méndez Soto wrote:
My question is, what is the benefit of using Lua?
I've never used Lua, but I also have a curiosity about it.
A couple of years ago, I wrote my first dialplan in AEL. Some bits were
clumsy, minor syntax errors caused major parts of my
I wouldn't jump to a whole different language just to have an elegant
script plan. There must be another reason why Lua is being so widely
implemented than elegance and execution performance.
Anyone?
*José Pablo Méndez
*
On Fri, Jan 6, 2012 at 1:12 PM, Steve Edwards
On 01/05/12 21:24, Joseph wrote:
On 01/05/12 22:12, Bruce B wrote:
but not it is not working again.
I wish they stop screwing up with that Asterisk, they keep
introducing new version and more bugs :-/
Wish not granted !!! :-) You will be the guinea pig to new features !!!
Same
AFAIK insecure=very has been replaced by insecure=port,invite. Also, you
might want to put wait(2) or progress after answer in the dialplan to allow
CID to process.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf
On Fri, 6 Jan 2012, José Pablo Méndez Soto wrote:
I wouldn't jump to a whole different language just to have an elegant
script plan.
I would and probably will.
Maybe it's just me, but I look at a dialplan filled with database access,
conditional gotos and priority labels and my eyes just
Not to purposely open a flame war here, but is lua preferable to ael?
Aren't we better off just properly documenting our original conf files since
new changes often introduce bad opportunities?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
On Fri, 6 Jan 2012, Danny Nicholas wrote:
Not to purposely open a flame war here, but is lua preferable to ael?
Aren't we better off just properly documenting our original conf files
since new changes often introduce bad opportunities?
One of the advantages of Lua is that it is embedded in
On Fri, Jan 6, 2012 at 10:01 AM, Benny Amorsen benny+use...@amorsen.dkwrote:
David Backeberg dbackeb...@gmail.com writes:
Thanks for clearing that up. I was getting all excited that I could
flash the PAP2T; I've always used regular voice tones over SIP with
the PAP2Ts.
SPA-2102 supports
Just installed asterisknow 1.6. I can access freepbx. I need to test
system on my LAN. Which softphone is best to use? I'm running ubuntu
on Dell optiplex G260 desktop at home. I'm hoping to setup basic IP PBX
for incoming/outgoing calls. No video.
Tom
--
On Sat, Jan 7, 2012 at 12:00 AM, Tom Poe tom...@meltel.net wrote:
Just installed asterisknow 1.6. I can access freepbx. I need to test
system on my LAN. Which softphone is best to use? I'm running ubuntu on
Dell optiplex G260 desktop at home. I'm hoping to setup basic IP PBX for
On 01/06/2012 10:30 AM, Ron Bergin wrote:
Add a BEGIN {...} block prior to the use statements and in there redirect
STDERR to a file. This will aloow you to capture compilation errors You
should also add some debugging statements at key points in the script.
Then run the script and review
On 01/06/12 14:44, Danny Nicholas wrote:
AFAIK insecure=very has been replaced by insecure=port,invite. Also, you
might want to put wait(2) or progress after answer in the dialplan to allow
CID to process.
I've tried putting even wait(7) it didn't help.
The problem is I'm getting this error:
Check your sip.conf and users.conf - my guess is that the pstn-1270 is an
assigned value that you need to remove or comment out.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph
Sent: Friday, January 06,
On 01/06/2012 04:03 PM, Ross Cameron wrote:
On Sat, Jan 7, 2012 at 12:00 AM, Tom Poe tom...@meltel.net
mailto:tom...@meltel.net wrote:
Just installed asterisknow 1.6. I can access freepbx. I need to
test system on my LAN. Which softphone is best to use? I'm
running ubuntu on
On Fri, 6 Jan 2012, Danny Nicholas wrote:
AFAIK insecure=very has been replaced by insecure=port,invite. Also,
you might want to put wait(2) or progress after answer in the dialplan
to allow CID to process.
Is this an issue with SIP? I thought either it was in the INVITE or it
wasn't.
--
Ok so its not a cosmetic thing only. I eases your administration. Do a
point for performance.
Now, what about my questions regarding extending the systems caps by
building things asterisk could not build by itself. does it hold true?
On Jan 6, 2012 3:28 PM, Steve Edwards
On 01/06/12 16:49, Danny Nicholas wrote:
Check your sip.conf and users.conf - my guess is that the pstn-1270 is an
assigned value that you need to remove or comment out.
What do you mean assigned value
My user.conf:
[general]
fullname = New User
userbase = 6000
hasvoicemail = yes
vmsecret =
Putting in a Wait(n) is only (sometimes) needed to wait for the CallerID NAME
on PRI or BRI.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Friday, January 06, 2012 6:06 PM
To: Asterisk
On 01/06/12 18:15, Eric Wieling wrote:
Putting in a Wait(n) is only (sometimes) needed to wait for the CallerID NAME
on PRI or BRI.
Putting wait(5) in my dial plan doesn't work as I'm getting this error message:
WARNING[2344]: chan_sip.c:13930 check_auth: username mismatch, have 11, digest
As I said, the Wait is only useful for PRI and BRI. It is TOTALLY USELESS for
SIP and IAX.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph
Sent: Friday, January 06, 2012 6:35 PM
To: Asterisk Users
On 01/06/12 16:35, Joseph wrote:
On 01/06/12 18:15, Eric Wieling wrote:
Putting in a Wait(n) is only (sometimes) needed to wait for the CallerID NAME
on PRI or BRI.
Putting wait(5) in my dial plan doesn't work as I'm getting this error message:
WARNING[2344]: chan_sip.c:13930 check_auth:
Does sox have more features on a Debian system than RHEL? Is that why it
won't work on RHEL?
Cheers,
On Wed, Jan 4, 2012 at 6:42 PM, Lefteris Zafiris zaf@gmail.com wrote:
Fresh code is out! The use of sox can be now optionally enabled by the
user if the system has a recent version of the
On Fri, 6 Jan 2012 20:46:14 -0500
Bruce B bruceb...@gmail.com wrote:
Does sox have more features on a Debian system than RHEL? Is that why
it won't work on RHEL?
RHEL's 5 version of sox is really old and outdated. The command syntax
and the switches are totally different compared to recent
On Saturday 07 Jan 2012, Tom Poe wrote:
Just installed asterisknow 1.6. I can access freepbx. I need to
test system on my LAN. Which softphone is best to use? I'm running
ubuntu on Dell optiplex G260 desktop at home. I'm hoping to setup
basic IP PBX for incoming/outgoing calls. No video.
Thanks.
I have been testing Aastra phones with SIP and had great results. I am
testing my cell phone now and sometimes get -1 for id, status, utterance,
and confidence. What does that mean?
Cheers
On Fri, Jan 6, 2012 at 9:40 PM, Lefteris Zafiris zaf@gmail.com wrote:
On Fri, 6 Jan 2012
NVM. I explored the code and see the logic. I had sox = 1 so it was failing
on RHEL.
To report, my cell phone from a PRI gets same confidence level just like
SIP. Building my control app now. Should make my life much easier while
driving. Thanks again :-)
-Bruce
On Fri, Jan 6, 2012 at 10:50 PM,
I'm trying Asterisk 10.0 (as 8.x is not passing PSTN CallerID) and Asterisk
10.0 is no better.
I'm still getting:
WARNING[12295]: chan_sip.c:14446 check_auth: username mismatch, have 11, digest has
pstn-1270
NOTICE[12295]: chan_sip.c:22769 handle_request_invite: Failed to authenticate device
Added two new features to the script: Timeout value and speechdata type.
*exten = s,n,agi(speech-recog.agi,en-US,3000,phoneNumb)*
- Will listen for 3 seconds and sanitize return as a single number without
any spaces in between. This helps when one reads phone number in format
415-554-2323 and
50 matches
Mail list logo
