Aloha,
We are looking to roll a solution that will have the following network layout:
ISDN-PRI -- Asterisk -- T.38 -- ATA -- Fax
Does version 1.8 with the Digium fax driver have this capability? I
like 1.8 because it is a long term support version.
What ATA's are people using?
Any working
On 01/04/2012 07:51 AM, Bruce B wrote:
And with recent version 14.3.2 I get:
/usr/local/bin/sox FAIL formats: no handler for file extension `flac'
-- speech-recog.agi: /usr/local/bin/sox failed: 512
-- SIP/-002eAGI Script speech-recog.agi completed, returning 0
Regards,
Hi,
Does anybody know if RAMI (Ruby Ami) is still functional?
And is this still compatible with asterisk 1.8
Best Regards,
Arjan Kroon
Mobillion BV
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Please help me..
Best Regards,
*Jayesh Labade*
e-mail: jayesh.lab...@gmail.com
On Wed, Jan 4, 2012 at 12:51 PM, Jayesh Labade jayesh.lab...@gmail.comwrote:
Hello Experts,
I have pasted my issue in http://pastebin.com/zBGVmdcY
I Cant able to Originate call from SIp trunk..I got this [Jan
Hi,
Give the complete details about the asterisk version, and SIP trunk conf
details
On Wed, Jan 4, 2012 at 3:07 PM, Jayesh Labade jayesh.lab...@gmail.comwrote:
Please help me..
Best Regards,
*Jayesh Labade*
e-mail: jayesh.lab...@gmail.com
On Wed, Jan 4, 2012 at 12:51 PM, Jayesh
Hi,
I am using asterisk ver 1.8.8.1.
My SIP trunk conf details are below..
[general]
context=default ; Default context for incoming calls
realm=192.168.1.55
allowguest=yes
realmauth=yes
send_rpid=pai
register = test02:test02@192.168.1.55
[test02]
type=peer
nat=no
Hi there
Happy New Year
I have a new install of asterisk 1.8.8.1 on ubuntu server
3.0.0-14-server #23-Ubuntu SMP Mon Nov 21 20:49:05 UTC 2011 x86_64 x86_64
x86_64 GNU/Linux
It has a Sangoma A200 card and I thought should be fairly standard but I have a
new error when trying to start asterisk
On Wednesday 04 January 2012, Duncan Turnbull wrote:
Hi there
Happy New Year
I have a new install of asterisk 1.8.8.1 on ubuntu server
3.0.0-14-server #23-Ubuntu SMP Mon Nov 21 20:49:05 UTC 2011 x86_64 x86_64
x86_64 GNU/Linux
It has a Sangoma A200 card and I thought should be fairly
I'm using the Linksys PAP2T and the Grandstream with SpanDSP and tx_fax and
rx_fax on multiple installations with no problems.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Darnell
Sent: Wednesday, 4
Hi checked your debug like.
Did you check that your SIP device ir registered with server ?
if yes then dial below command from CLI
*originate sip/test02 application dial*
On Wed, Jan 4, 2012 at 3:33 PM, Jayesh Labade jayesh.lab...@gmail.comwrote:
Hi,
I am using asterisk ver 1.8.8.1.
My
On 4/01/2012, at 11:47 PM, A J Stiles wrote:
For what it's worth, I once tried installing Asterisk on an old VIA C7 box;
and it turns out that this processor, while detecting as an i686, doesn't
implement the full i686 instruction set -- and Asterisk is trying to use one
of the
Hi virendra,
Dialed same command.. I got below output
ast18*CLI originate sip/test02 application dial
== Using SIP RTP CoS mark 5
[Jan 4 14:13:07] NOTICE[29823]: chan_sip.c:19718 handle_response_invite:
Failed to authenticate on INVITE to 'Anonymous
Hi All,
I am trying to set call codec at extension.conf but it doesnt work ... its like
my command doesnt change anything
exten=6500,1,Answer
exten=6500,2,Playback(welcome)
exten=6500,3,SIPAddHeader(email:${SIP_HEADER(email)})
I loaded the latest 1.6 which gets slightly further and a core dump shows this,
but its past my ability to interpret
# gdb -se asterisk -c core | tee /tmp/backtrace.txt
GNU gdb (Ubuntu/Linaro 7.3-0ubuntu2) 7.3-2011.08
Copyright (C) 2011 Free Software Foundation, Inc.
License GPLv3+: GNU GPL
this looks great - is there any chance of coverting the googletts.agi
to use flac as well ?
Julian
On 4 January 2012 09:06, Lefteris Zafiris zaf@gmail.com wrote:
On 01/04/2012 07:51 AM, Bruce B wrote:
And with recent version 14.3.2 I get:
/usr/local/bin/sox FAIL formats: no handler for
On 01/04/2012 04:07 PM, Julian Lyndon-Smith wrote:
this looks great - is there any chance of coverting the googletts.agi
to use flac as well ?
Julian
In googletts.agi we get the voice data from google in mp3 and we convert
it in a format that asterisk can read and playback (slin). If we
the only reason is that I didn't want to have to install sox. Lazy.
that's all ;) Just another piece of software to find and install
running on amazon ec2, is the best thing to download the source and
compile sox ?
Thanks
Julian
On 4 January 2012 14:18, Lefteris Zafiris zaf@gmail.com
On 01/04/2012 04:24 PM, Julian Lyndon-Smith wrote:
the only reason is that I didn't want to have to install sox. Lazy.
that's all ;) Just another piece of software to find and install
running on amazon ec2, is the best thing to download the source and
compile sox ?
Thanks
It should be
nope :(
On 4 January 2012 14:29, Lefteris Zafiris zaf@gmail.com wrote:
On 01/04/2012 04:24 PM, Julian Lyndon-Smith wrote:
the only reason is that I didn't want to have to install sox. Lazy.
that's all ;) Just another piece of software to find and install
running on amazon ec2, is the
Hello,
Which QSIG (ECMA or ISO) variant and profiles does asterisk support ?
(I could not find this info inside chan_dahdi.conf)
Regards
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to
Hey,
There is a new kid in town if you want to code in ruby. Use
adhearsionhttps://github.com/adhearsion/adhearsion/wiki,
it's a framework to make voice apps.
On Wed, Jan 4, 2012 at 2:49 PM, Arjan Kroon | Mobillion
arjan.kr...@mobillion.nl wrote:
Hi,
Does anybody know if RAMI (Ruby Ami) is
Hi all,
I'm trying to run an AGI in PERL which uses the module DBD-Oracle.
Currently my AGI is working fine in my two servers but not in my other four
servers. When I tried execute an AGI (as a user asterisk) in command line
it works fine (even I also declare environmental variables in user
We did get this fixed. Turns out that my tech didn't reboot the phone after
disabling the vlan configuration. He's new and still learning.
Thank you for your time and suggestions.
On Monday 02 January 2012 6:04:49 pm Jim DeVito wrote:
Agreed. Check the switch for some kind of port security.
Providing which version of Asterisk you are using might be helpful.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 12:09 PM
To: Asterisk Users Mailing List -
anyhelp guys?
I tried a lot of stuff but it doesnt work the Codec for audio call only
cannt be set...how I can set the call type video/audio at dail plan?
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On
1.6 and 1.8 ... I tried changing stuff on both
when I make audio call from my client which supports both audio and video its
sent to the other client as video call .I tried settings the
SIP_CODED_INBOUND and outbound also ... but no luck
From:
1.6 does not support setting the outbound codec.1.8 uses different
variables to set the outbound codec. See UGRADE.txt in the Asterisk source for
the 1.8 information,.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
I tried also in asterisk 1.8 setting outbound variable but didnt work also
https://wiki.asterisk.org/wiki/display/AST/chan_sip+Channel+Variables
check the above ... I changed it and tried but still I get a video call
From:
My guess is that you should set the codec either before SIPADDHEADER or
before ANSWER.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 11:25 AM
To: Asterisk Users
The module probably isn't readable/executeable from Asterisk
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ahmed Munir
Sent: Wednesday, January 04, 2012 10:45 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] asterisk
how can u give me a command?!..
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
[da...@debsinc.com]
Sent: Wednesday, January 04, 2012 11:29 AM
To: 'Asterisk Users Mailing List -
Move line 5 up to line 3 or line 1 (P.S. - the 1-6 numbering scheme is
cumbersome; 1-n-n-n-n-n is more practical).
Like this
exten=6500,1,Answer
exten=6500,n,Playback(welcome)
exten=6500,n,set(SIP_CODEC=gsm) -- this is not changed
exten=6500,n,SIPAddHeader(email:${SIP_HEADER(email)})
didnt work also :(
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
[da...@debsinc.com]
Sent: Wednesday, January 04, 2012 11:39 AM
To: 'Asterisk Users Mailing List - Non-Commercial
CLI output from call?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 11:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
-- Executing [6500@DLPN_DialPlan1:1] Set(SIP/6000-, SIP_CODEC=gsm
) in new stack
-- Executing [6500@DLPN_DialPlan1:2] Set(SIP/6000-, SIP_CODEC_INB
OUND=gsm) in new stack
-- Executing [6500@DLPN_DialPlan1:3] Set(SIP/6000-, SIP_CODEC_OUT
BOUND=gsm) in new stack
You are fighting a losing battle - you can't control the other end
Ignoring ${SIP_CODEC} variable because it is not shared by both ends.
You can probably do a SIP SET DEBUG ON and see what codecs are available on
the other end.
-Original Message-
From:
I am the other end most codecs are available
now my problem is when I make audio call using one side its converted to video
call request (since my other end has also all codecs)
my app clients can do Audio and Video call,
now the Video call is ok
but the Audio part get converted to
Any suggestion will be great
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
[fkha...@iconnecths.com]
Sent: Wednesday, January 04, 2012 11:55 AM
To: Asterisk Users Mailing List -
Please post the sip.conf entries for 6000 and 6500.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 11:56 AM
To: Asterisk Users Mailing List - Non-Commercial
there is nothing in sip.conf about what u asked
but 6500 is a queue with following info
[6500]
fullname = testing
strategy = rrmemory
timeout = 15
wrapuptime = 15
autofill = no
autopause = no
joinempty = yes
leavewhenempty = no
reportholdtime = no
maxlen = 0
musicclass = test
member = SIP/6251
On Wednesday 04 January 2012, Duncan Turnbull wrote:
I loaded the latest 1.6 which gets slightly further and a core dump shows
this, but its past my ability to interpret
# gdb -se asterisk -c core | tee /tmp/backtrace.txt
GNU gdb (Ubuntu/Linaro 7.3-0ubuntu2) 7.3-2011.08
Copyright (C) 2011
Is this freeware, or a module which you can include in your ruby code?
Or is it a complete framework?
On 04 Jan 2012, at 5:31 PM, gokulnath wrote:
Hey,
There is a new kid in town if you want to code in ruby. Use
adhearsionhttps://github.com/adhearsion/adhearsion/wiki, it's a framework to
make
What about the allow/disallow lines in sip.conf?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 12:04 PM
To: Asterisk Users Mailing List - Non-Commercial
allow=all
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
[da...@debsinc.com]
Sent: Wednesday, January 04, 2012 12:09 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Hi,
I installed the modules in asterisk user home directory with read and
excitable permissions for asterisk but still my AGI not working.
Please provide me other advise to resolve this issue.
Date: Wed, 4 Jan 2012 11:30:34 -0600
From: Danny Nicholas da...@debsinc.com
Subject: Re:
Both sides?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 12:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set
yup and video support is yes
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
[da...@debsinc.com]
Sent: Wednesday, January 04, 2012 12:15 PM
To: 'Asterisk Users Mailing List -
What are the permissions on the module you are trying to run? (ls -l
/var/lib/asterisk/agi-bin/module)
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ahmed Munir
Sent: Wednesday, January 04, 2012 12:15 PM
To:
Note to self: Never release anything asterisk related without testing
on RHEL/Centos 5
Thank you for reporting this. I have replaced sox with flac and it seems
to work now on older platforms too (tested on Centos 5 with asterisk 1.4).
You can get the updated code here:
Does anyone know what languages are supported?
-Original Message-
From: Bruce B bruceb...@gmail.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Wed, 4 Jan 2012 13:25:18
To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
Reply-To: Asterisk
Wow - nice! A few quick questions:
1. How long can the recording be for translation?
2. Any limitation on how much text the return (transcribed) variable can hold?
3. Any commercial / terms of use limitations?
From: asterisk-users-boun...@lists.digium.com
On Wed, 4 Jan 2012, Arjan Kroon | Mobillion wrote:
Is this freeware, or a module which you can include in your ruby code?Or
is it a complete framework?
Is this list faster than Google?
--
Thanks in advance,
-
Steve
On Wed, 4 Jan 2012, A J Stiles wrote:
If you stick a /* harmless comment */ in this file and re-save it, this will
give the file a new modification time. Then run `make` again. It will
recompile just localtime.c (this being the only source file that has changed
since the last time make was
On Wed, Jan 4, 2012 at 8:47 PM, Michelle Dupuis mdup...@ocg.ca wrote:
Wow - nice! A few quick questions:
1. How long can the recording be for translation?
At the moment the recording timeout is set at 15sec. I haven't tested
yet the max
length of voice data ta google accepts (all this voice
On Wed, Jan 4, 2012 at 8:27 PM, isr...@gmail.com wrote:
Does anyone know what languages are supported?
For sure english and spanish, since its undocumented i don't have a
complete list
yet.
Lefteris Zafiris
--
Un-top-posting...
On Wed, 4 Jan 2012, Ahmed Munir wrote:
I'm trying to run an AGI in PERL which uses the module DBD-Oracle.
Currently my AGI is working fine in my two servers but not in my other
four servers. When I tried execute an AGI (as a user asterisk) in
command line it works fine
Works beautifully. Amazing job Lefteris. Thanks.
The best result I got in probability was 0.9725632 by saying, hello. I
think there is some non-phonetic logic built-in as well. I tried, 1, 2 and
I got 0.86534226 in accuracy. While I tried 1, 2, 3, 4, 5 I got,
0.97256315. Probably Google
Hello,
I see the following error in the logs
[Jan 4 11:37:35] NOTICE[21]: chan_sip.c:15388 check_user_full: From address
missing 'sip:', using it anyway
Does anybody know how to stop this error? It does not seem to be affecting
performance on the Asterisk 1.8.4.1 running on Centos Linux 2.6. I
On 1/4/2012 2:26 PM, Lefteris Zafiris wrote:
Works beautifully. Amazing job Lefteris. Thanks.
The best result I got in probability was 0.9725632 by saying, hello. I
think there is some non-phonetic logic built-in as well. I tried, 1, 2 and
I got 0.86534226 in accuracy. While I tried 1, 2, 3,
wow i just tried in hebrew and i'll say just 1 word WOW
On Wed, Jan 4, 2012 at 9:48 PM, sean darcy seandar...@gmail.com wrote:
On 1/4/2012 2:26 PM, Lefteris Zafiris wrote:
Works beautifully. Amazing job Lefteris. Thanks.
The best result I got in probability was 0.9725632 by saying, hello.
On 1/4/2012 4:37 AM, Jayesh Labade wrote:
Please help me..
Best Regards,
*Jayesh Labade*
e-mail: jayesh.lab...@gmail.com mailto:jayesh.lab...@gmail.com
On Wed, Jan 4, 2012 at 12:51 PM, Jayesh Labade jayesh.lab...@gmail.com
mailto:jayesh.lab...@gmail.com wrote:
Hello Experts,
I have
On Wed, 04 Jan 2012 14:48:22 -0500
sean darcy seandar...@gmail.com wrote:
This is really spectacular. Thanks.
I'm running Fedora 15, so I can use flac or sox. Any reason to prefer
one over the other?
sean
We have to convert the voice data to flac format before sending them to
google,
On 5/01/2012, at 8:06 AM, Steve Edwards wrote:
On Wed, 4 Jan 2012, A J Stiles wrote:
If you stick a /* harmless comment */ in this file and re-save it, this will
give the file a new modification time. Then run `make` again. It will
recompile just localtime.c (this being the only source
Hi!
Hello! I wanted to know if you have experienced problems installing both a
Sangoma and a Digium card in the same Server.
Thnks a lot!
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk?
The Asterisk Development Team is pleased to announce the first
release of DAHDI-Linux 2.6.0 and DAHDI-Tools 2.6.0.
2.6.0 is a feature release which:
- Adds support for the TE820 8-span card to the wct4xxp driver.
- Decrease load time of analog cards supported by the wctdm24xxp
driver.
On 01/01/2012 04:17 PM, cov...@ccs.covici.com wrote:
Hi. I am using asterisk 1.8 and everything was working fine when I was
at svn 342661. I then upgraded to vrsion 349339 and discovered the
following problem -- one of the end points is a freeswitch box which
offers a number of codecs,
On 01/03/2012 10:03 AM, Patrick Lists wrote:
On 03-01-12 16:24, Danny Nicholas wrote:
Hello List,
I work in an environment where I have to request IPTABLES changes rather
than doing them myself. Is there a way to utilize my SSH (port 22) to
get a functional (and with good sound) Asterisk
On 01/04/2012 12:25 AM, Matt Darnell wrote:
Aloha,
We are looking to roll a solution that will have the following network layout:
ISDN-PRI-- Asterisk-- T.38-- ATA-- Fax
Does version 1.8 with the Digium fax driver have this capability? I
like 1.8 because it is a long term support version.
DT == Duncan Turnbull dun...@e-simple.co.nz writes:
DT I have a new install of asterisk 1.8.8.1 on ubuntu server
DT 3.0.0-14-server #23-Ubuntu SMP Mon Nov 21 20:49:05 UTC 2011 x86_64 x86_64
x86_64 GNU/Linux
DT The only errors I can see are limited - I also stopped wan router and dahdi
and I
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 program (won't work in
RHEL/Centos 5)
This is done by editing the script and setting the variable 'use_sox'.
When sox is used the audio gets normalized, low frequency noise (100Hz)
On 5/01/2012, at 12:21 PM, James Cloos wrote:
DT == Duncan Turnbull dun...@e-simple.co.nz writes:
DT I have a new install of asterisk 1.8.8.1 on ubuntu server
DT 3.0.0-14-server #23-Ubuntu SMP Mon Nov 21 20:49:05 UTC 2011 x86_64 x86_64
x86_64 GNU/Linux
DT The only errors I can see are
Kevin P. Fleming kpflem...@digium.com wrote:
On 01/01/2012 04:17 PM, cov...@ccs.covici.com wrote:
Hi. I am using asterisk 1.8 and everything was working fine when I was
at svn 342661. I then upgraded to vrsion 349339 and discovered the
following problem -- one of the end points is a
hello:
i think it can be done, please refer this link:
http://wiki.sangoma.com/Asterisk-FAQ#Digium
Best regards,
James.zhu
Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri-SIP).
website: www.voipviews.com
Date: Wed, 4 Jan 2012 18:47:28 -0200
From:
my config:
hardphone - pstn gateway - asterisk - pstn gateway - hardphone
i am using asterisk 1.4.xx
w option is Dial is for recording. how does it related to ringtone?
pls advise.
從︰ Carlos Rojas crt.ro...@gmail.com
收件人︰ Qqblog Qqblog qqb...@ymail.com;
On Wed, Jan 4, 2012 at 1:02 AM, David Klaverstyn
da...@klaverstyn.com.au wrote:
I'm using the Linksys PAP2T and the Grandstream with SpanDSP and tx_fax and
rx_fax on multiple installations with no problems.
David,
Are you running 10.0 or 1.8?
Glad to know that the PAP2T has a solid T.38
I'm using 1.8, but also have 1.4 and 1.2 installs using the same configuration.
Regards
David.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Darnell
Sent: Thursday, 5 January 2012 1:55 PM
To: Asterisk
Hi Kevin,
Thanks for your suggestion.
On the website of OpenAIS, it seems that it is not supported anymore and
their download links (FTP and SVN) are broken (been trying it for about a
month now). Is it still possible to use OpenAIS method? The other solution
on the wiki is using XMPP which is
Hi,
Sorry for late reply. Hope you've already found out something about it.
What version of asterisk you are using, that function for choosing
inbound/outbound call leg codecs is for newer versions of asterisk.
See these pages:
http://www.voip-info.org/wiki/view/Asterisk+variables
Can anybody please reply on this?
Regards,
Kamlesh
From: kamlesh_...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Tue, 27 Dec 2011 09:49:21 +
Subject: Re: [asterisk-users] DIALSTATUS Values
Hello,
After investing some time, I could come to know the reason for not
Hi,
one reason for having that robotic voice could be improper codecs others
include low CPU processing power, memory not free etc. I once had the same
kind of issue with VAD(voice activity detection) turned ON from my service
providers equipment so my asterisk was performing poorly with VAD.
Hi,
The server or client application that is sending you sip packets is missing
the sip: string in from header. You should have it sorted out because if
that header goes to some external equipment the call may fail because of
this.
Regards,
Sammy
On Thu, Jan 5, 2012 at 12:44 AM, motty.cruz
This works fine for me,
$dialstatus = $agi-get_variable(DIALSTATUS);
$cdr['dialstatus'] = $dialstatus['data'];
Try as it is, I believe it's because of concatenation.
Regards,
Zohair Raza
On Fri, Dec 2, 2011 at 4:27 PM, Tony Mountifield t...@softins.co.uk
Are you talking about having an SSH tunnel and route your SIP traffic
through it !!?
On Thu, Jan 5, 2012 at 4:20 AM, Kevin P. Fleming kpflem...@digium.comwrote:
On 01/03/2012 10:03 AM, Patrick Lists wrote:
On 03-01-12 16:24, Danny Nicholas wrote:
Hello List,
I work in an environment where
Hello,
Trying to set up res_fax_spandsp. Based on
https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway I wrote this in
my extensions.conf:
exten = 306,1,NoOp(Fax transmission)
same = n,Set(FAXOPT(gateway)=yes)
same = n,Dial(DAHDI/3)-FXS port to fax machine
same
I guess this is a permissions issue.
Make sure your agi script has execute permissions (755) and it belongs
to asterisk:asterisk .
for that you need:
chmod 755 /var/lib/asterisk/agi-bin/agi-script-name.agi chown
asterisk:asterisk /var/lib/asterisk/agi-bin/agi-script-name.agi
Regards,
LL
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