Doug:
for what it's worth I am having the exact same nightmare. Not sure exactly
when it started but I believe it was a change in 1.8.8.1 / 1.8.9.0-rc1 (I am
running 1.8.9rc1). I also have Polycom (335, 550, 650) and blind transfers
are broken. All legs of the call are dropped when the xfer is
On 01/07/2012 09:34 AM, Bruce B wrote:
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
Hello
I just read this article about an Asterisk server that got hacked to
make free international calls through an ITSP:
www.rowetel.com/blog/?p=2210
I have a couple of questions:
1. Am I correct in understanding that SIP ALG on a router makes it
easier to host an Asterisk server on a
Hi,
I have an Opensips server dispatching to 3 Asterisk servers. I would
like to assign public IPs to all of these servers and avoid NAT
altogether - phones will also have public IPs. The way I set this in the
lab, all the SIP traffic goes thru the SIP proxy (Opensips) and RTP
goes
On 01/06/2012 05:00 PM, 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.
Tom
On Sat, 07 Jan 2012 09:27:29 -0500, sean darcy seandar...@gmail.com
wrote:
But what really made us choose linphone was you use it on android/iphone.
That has been a huge plus. As a bonus, you can use any degegistered
smartphone - that is, one not hooked up to the cellular network,only
wireless
- Original Message -
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
On Sat, Jan 7, 2012 at 5:19 AM, Luke Hamburg l...@solvent-llc.com wrote:
Doug:
for what it's worth I am having the exact same nightmare. Not sure exactly
when it started but I believe it was a change in 1.8.8.1 / 1.8.9.0-rc1 (I
am
running 1.8.9rc1). I also have Polycom (335, 550, 650) and
On Fri, 2012-01-06 at 16:00 -0600, 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
On 01/07/12 08:50, Tim Nelson wrote:
- Original Message -
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]:
In article 20120107163819.gc3...@syscon7.inet,
Joseph syscon...@gmail.com wrote:
I'm not sure this is the case:
Asterisk-1.4.39
[home_server]
type=friend
host=dynamic
secret=123456
context=extensions
disallow=all
allow=ulaw
allow=alaw
requirecalltoken=no
Asterisk-1.8.7
Error message on asterisk-1.4.39
chan_iax2.c:9541 socket_process: Rejected connect attempt from
192.168.141.8, requested/capability 0x2/0x703 incompatible with our
capability 0xc.
According to this log, server 192.168.141.8 has codecs defined as 0xc
(ulaw and alaw), which matches your
On Sat, Jan 7, 2012 at 9:34 AM, Gilles codecompl...@free.fr wrote:
On Sat, 07 Jan 2012 09:27:29 -0500, sean darcy seandar...@gmail.com
wrote:
But what really made us choose linphone was you use it on android/iphone.
That has been a huge plus. As a bonus, you can use any degegistered
smartphone
On Fri, 6 Jan 2012, Dale Noll wrote:
I found the following lines to be helpful.
$ENV{TNS_ADMIN}=/usr/lib/oracle/11.2/client/;
$ENV{ORACLE_HOME}=/usr/lib/oracle/11.2/client/;
$ENV{LD_LIBRARY_PATH}=/usr/lib/oracle/11.2/client/lib/;
I think a 'better practice' would be to put the 'stuff
On Sat, 7 Jan 2012 12:34:44 -0500, Sean Darcy seandar...@gmail.com
wrote:
Yes, I did mean de-registered. I meant a phone that no longer has the
ability to use the cellular network - only wifi. For instance, we have
a couple of Droids that used to be on Verizon. They work just fine as
sip-phones
Chances are the incoming call is not matching anything in iax.conf. turn on
iax debug, try a call, post the results. Maybe someone familiar with IAX can
help you.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
On 01/07/12 17:13, Tony Mountifield wrote:
In article 20120107163819.gc3...@syscon7.inet,
Joseph syscon...@gmail.com wrote:
I'm not sure this is the case:
Asterisk-1.4.39
[home_server]
type=friend
host=dynamic
secret=123456
context=extensions
disallow=all
allow=ulaw
allow=alaw
This means you are allowing guest calls. A VERY bad thing.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph
Sent: Saturday, January 07, 2012 1:27 PM
To: Asterisk Users Mailing List - Non-Commercial
On 01/07/12 13:27, Eric Wieling wrote:
This means you are allowing guest calls. A VERY bad thing.
Doesn't it pertain to codes only?
in my [guest] section I have:
;[guest]
;type=user
;context=default
;callerid=Guest IAX User
so it is disabled, isn't it?
--
Joseph
-Original
The codecs and contexts defined in [general] apply to unauthenticated calls.
If the incoming call matched the entry in sip.conf or iax.conf then the codecs
in that entry would be used.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Could be the phone firmware. I'm not sure. I'll probably get it resolved next
week post back how it goes.
-
Doug Mortensen
Sent via DroidX2 on Verizon Wireless™
-Original message-
From: Ryan Wagoner rswago...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
Oh crap. I just reread the previous post realized I'm not alone. Hallelujah!
I'll post back more info soon.
-
Doug Mortensen
Sent via DroidX2 on Verizon Wireless™
-Original message-
From: Ryan Wagoner rswago...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
On 01/07/12 13:42, Eric Wieling wrote:
The codecs and contexts defined in [general] apply to unauthenticated calls.
If the incoming call matched the entry in sip.conf or iax.conf then the codecs
in that entry would be used.
I just change in iax.conf in [general] section:
from:
allow=all
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