moving the
'pass' rule to below your 'block' rules will allow any connections
originating from networks listed in your table and also exists
in the table, will be blocked.
Larry.
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On 2/04/2020 5:39 AM, Larry Moore wrote:
On 2/04/2020 5:28 AM, Mark Boyce wrote:
On 1 Apr 2020, at 22:14, Greg Troxel <mailto:g...@lexort.com>> wrote:
I think you need to use tcpdump and turn up firewall debugging.
sngrep is your friend …My bet is UDP vs TCP on firewall rules
ists when the table entry is updated.
Does your script also issue a command to kill existing states from that
host after it has updated the table, e.g. pfctl -k 45.143.220.235
Larry.
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hich will include the TSI, is all part of the image (Tagline in
HylaFAX) and not stored separately on the receiving terminal.
Cheers,
Larry.
On 18/12/2016 6:20 PM, Yves wrote:
Hi,
thanks for your answer. Unfortunately this is, what I already know. I
was wondering, why it is possible to se
([Header Position])
Larry.
On 18/12/2016 4:30 AM, Yves wrote:
Hi,
I am using asterisk 11.8 in combination with spandsp to send and
receive T38 Faxes. All works fine, but I do not know
how to get the remoteheader from the fax I receive.
When I send a fax, there are Faxopts to set
Hi,
If the 'fax show version' command doesn't work on your system it will
also indicate you don't have the 'res_fax' module installed, thus you
won't be able to take advantage of the T.38 gateway functionality, did
you try the command?
You're indicating you are having problems with IAX
Cheers,
Larry.
On 15/11/2016 8:09 PM, tux john wrote:
Hi. Since I am messing a lot with it without seeing the end of, may I
ask if there is any solid guide for that please?
On 13/11/2016, 07:42 Larry Moore <lmo...@starwon.com.au> wrote:
Some additional information which may he
});
Set(FAXRXPATH=/var/spool/asterisk/fax/received);
Set(FAXRXFILE=fax-${CALLERID(number)}-${UNIQUEID});
NoOp( RECEIVING FAX : ${FAXRXFILE} );
ReceiveFAX(${FAXRXPATH}/${FAXRXFILE}.tif,f);
NoOp( Subroutine Retu
of Asterisk 11:
Set(FAXOPT(gateway)=yes)
I have it working in my installation however I have incoming voice calls
too hence I use 'faxdetect' to direct the call to the 'fax' extension.
Cheers,
Larry.
On 12/11/2016 5:24 AM, tux john wrote:
hi. i am using asterisk 11.24.1 in my raspberry. i
On my Asterisk 11 system I have the following in extensions.ael for
chan_sip.
8001=> {
Set(SIP_CODEC=alaw);
//Dial(MulticastRTP/linksys/224.168.168.168:34567/224.168.168.168:6061);
Page(MulticastRTP/linksys/224.168.168.168:34567/224.168.168.168:6061,q,5);
Could it be in the [general] section you should have;
accept_outofcall_message=yes
Your line appears to be missing the 'p' in accept and an extraneous 's'
in message.
Larry.
On 21/09/2015 2:48 PM, Emil Ohlsson wrote:
Hi,
I'm having trouble configuring Asterisk to respond to an incoming
On 29/07/2015 12:13 PM, Jean-Denis Girard wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Thanks for your reply Larry.
Le 27/07/2015 01:22, Larry Moore a écrit :
I think the 488 Not acceptable here is occurring because the channel
connecting through is not T.38 capable
but maybe you can make something
from this that helps.
Please note, I don't have the old set up to test so I can't be certain
of the above configurations.
Cheers,
Larry.
On 27/07/2015 11:15 AM, Jean-Denis Girard wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi list,
2 weks ago I asked
jbenable=no
faxdetect=no
directmedia=no
callbackextension=receive
t38pt_usertpsource=yes
encryption=no
Note, in this example I am using 'callbackextension' instead of
'register =', refer to the default sip.conf for further information.
Larry
an extension on your Asterisk server
which will receive the fax e.g. using ReceiveFAX() and see if your
connection problem persists?
Larry.
On 15/12/2014 5:10 AM, Larry Moore wrote:
On 14/12/2014 7:24 PM, Recursive wrote:
On 11.12.2014 11:53, Larry Moore wrote:
You may very well find getting T
!
You should also include information relating to your SIP configuration
(with appropriate obfuscations) for the connection to peer
27XgY8YwfI2S9NAg as well as what T.38 options you have set in the
general section of sip.conf.
Larry
On 23/10/2014 4:57 PM, Larry Moore wrote:
On 23/10/2014 5:43 AM, Leandro Dardini wrote:
Hello,
I am struggling to have a SPA504G to auto answer (for intercom/paging).
I have tried the following SIP headers (not all together), but without
luck:
SIPAddHeader(Call-Info:\;answer-after=0
.
It's been a while since I played with my SPA8800, with the test I
performed it did work, that was inbound and outbound calls.
Larry.
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On 17/11/2014 7:34 PM, Larry Moore wrote:
On 17/11/2014 6:42 PM, Olivier wrote:
Hello,
If I'm not mistaken, it is not possible to get T.38 on a SPA3102 FXO
port (it is possible with the FXS port).
Do you know, by experience preferably, if this is possible with an
SPA8800 FXO port
On 17/11/2014 9:24 PM, Olivier wrote:
2014-11-17 12:55 GMT+01:00 Larry Moorelmo...@omninet.net.au:
On 17/11/2014 7:34 PM, Larry Moore wrote:
On 17/11/2014 6:42 PM, Olivier wrote:
Hello,
If I'm not mistaken, it is not possible to get T.38 on a SPA3102 FXO
port (it is possible
On 23/10/2014 4:57 PM, Larry Moore wrote:
On 23/10/2014 5:43 AM, Leandro Dardini wrote:
Hello,
I am struggling to have a SPA504G to auto answer (for intercom/paging).
I have tried the following SIP headers (not all together), but without
luck:
SIPAddHeader(Call-Info:\;answer-after=0
On 24/10/2014 12:49 AM, Tim Nelson wrote:
- Original Message -
On 23/10/2014 10:07 PM, Larry Moore wrote:
On 22/10/2014 11:23 AM, Tim Nelson wrote:
Greetings-
Working with the T.38 gateway functionality that is sparsely
documented
[1], I'm attempting to get the following
On 25/10/2014 11:43 PM, Larry Moore wrote:
On 24/10/2014 12:47 AM, Tim Nelson wrote:
- Original Message -
On 22/10/2014 11:23 AM, Tim Nelson wrote:
Greetings-
Working with the T.38 gateway functionality that is sparsely
documented
[1], I'm attempting to get the following
On 24/10/2014 12:47 AM, Tim Nelson wrote:
- Original Message -
On 22/10/2014 11:23 AM, Tim Nelson wrote:
Greetings-
Working with the T.38 gateway functionality that is sparsely
documented
[1], I'm attempting to get the following functional:
What type of endpoint are you using
it all configured much like I have listed hear and it still
doesn't work then you need to check the firewall configuration on your
Asterisk system and ensure it is allowing outbound Multicast traffic.
Larry.
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On 23/10/2014 4:57 PM, Larry Moore wrote:
On 23/10/2014 5:43 AM, Leandro Dardini wrote:
Hello,
I am struggling to have a SPA504G to auto answer (for intercom/paging).
I have tried the following SIP headers (not all together), but without
luck:
SIPAddHeader(Call-Info:\;answer-after=0
On 23/10/2014 6:41 PM, Larry Moore wrote:
snip
Listing from my Asterisk:
'8000' = 1. Set(SIP_CODEC=alaw)
2. Dial(MulticastRTP/linksys/224.168.168.168:45678/224.168.168.168:6061)
3. Hangup()
snip
Hmm, my SPA525G doesn't auto-answer a page however my SPA92X do.
snip
Just upgraded
On 23/10/2014 3:55 AM, Tim Nelson wrote:
- Original Message -
Greetings-
Working with the T.38 gateway functionality that is sparsely
documented [1], I'm attempting to get the following functional:
Asterisk calling system - Asterisk system in T.38 Gateway Mode (box
in
On 22/10/2014 11:23 AM, Tim Nelson wrote:
Greetings-
Working with the T.38 gateway functionality that is sparsely documented
[1], I'm attempting to get the following functional:
What type of endpoint are you using which is originating the call and is
it T.38 capable?
Larry
On 23/10/2014 10:07 PM, Larry Moore wrote:
On 22/10/2014 11:23 AM, Tim Nelson wrote:
Greetings-
Working with the T.38 gateway functionality that is sparsely documented
[1], I'm attempting to get the following functional:
What type of endpoint are you using which is originating the call
On 24/10/2014 12:49 AM, Tim Nelson wrote:
- Original Message -
On 23/10/2014 10:07 PM, Larry Moore wrote:
On 22/10/2014 11:23 AM, Tim Nelson wrote:
Greetings-
Working with the T.38 gateway functionality that is sparsely
documented
[1], I'm attempting to get the following
directing the call to the SPA8800
Larry.
On 14/10/2014 8:59 PM, Frederic Van Espen wrote:
Hello list,
We're currently facing some issues concerning T.38 gateway faxing.
This is a device used almost exclusively for receiving faxes. Calls
are incoming to asterisk on a SIP trunk (sangoma
I don't know if this may help
https://www.escaux.com/docs/DRD_T38Support_AdminGuide.html assuming you
can enable T.38 on the Sangoma Netborder, then you can turn off the
faxgateway option.
Larry.
On 15/10/2014 3:19 PM, Larry Moore wrote:
It's been a while since I played with a Cisco SPA8800
On 9/10/2014 9:28 PM, Mitul Limbani wrote:
Oops its qualify= n not notify=
Also check if your asterisk sip server I available with ports on the
public ip that your phone is trying to register from 3G nw.
In your devices sip configuration set;
qualify=no
nat=yes
in Bria;
Settings -
to
process it in the 'h' extension.
You may want to refer to this thread
http://lists.digium.com/pipermail/asterisk-users/2013-June/279625.html,
note the afax2email script will e-mail the received tiff fax image if
the PDF conversion fails.
Larry
= yes,fec,maxdatagram=400 ; Enables T.38 with FEC error
correction and overrides
; ; the other endpoint's provided
value to assume we can
; ; send 400 byte T.38 FAX packets
to it.
;
Larry
=Alert-Autoanswer);
SIPAddHeader(Call-Info:\;Answer-After=0);
SIPAddHeader(P-Auto-Answer: normal);
SIPAddHeader(Answer-Mode: Auto);
Larry.
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New to Asterisk
doesn't help are;
faxdetect=cng
t38pt_udptl=no
Larry.
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?
Larry.
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asterisk-users mailing list
On 3/02/2014 7:38 PM, Jakob-Matthias Böttger wrote:
as He is describing it he should actually provide t.38. but i don't know
why it is not working thus im now getting
Feb 3 12:32:55] WARNING[9942][C-0004]: chan_sip.c:10353 process_sdp:
Failed to initialize UDPTL, declining image stream
[Feb
On 3/02/2014 8:42 PM, Jakob-Matthias Böttger wrote:
Am 03.02.2014 13:20, schrieb Jakob-Matthias Böttger:
Am 03.02.2014 12:56, schrieb Larry Moore:
On 3/02/2014 7:38 PM, Jakob-Matthias Böttger wrote:
as He is describing it he should actually provide t.38. but i don't
know
why it is not working
Hello,
Perhaps you need to have directmedia=no set for the channel, the call
doesn't appear to have been answered hence asterisk won't be able to
hear any tones to determine for itself if the call is an incoming fax.
Larry.
On 21/01/2014 6:51 PM, Jakob-Matthias Böttger wrote:
Hello
Sorry, I missed the line showing the call had been answered.
On 22/01/2014 8:11 AM, Larry Moore wrote:
Hello,
Perhaps you need to have directmedia=no set for the channel, the call
doesn't appear to have been answered hence asterisk won't be able to
hear any tones to determine for itself
Have you checked your localnet=, deny=, permit=, contactdeny=
contactpermit= settings?
My 2c worth.
On 20/01/2014 10:51 AM, David Cunningham wrote:
Hi,
We have a Kamailio and Asterisk cluster, both machines being on a real
103.x IP address and also on a 172.x OpenVPN address.
The problem
Is Kamalio running on the same system as Asterisk?
On 21/01/2014 2:41 PM, David Cunningham wrote:
Hi Larry,
Thanks for the reply. We have all of those settings left out of our
sip.conf, so this should allow everything, right?
On 21 January 2014 17:38, Larry Moore lmo...@omninet.net.au
it only has a the T38
information for the call thus no audio (g729) information is in the SIP
message.
I don't believe the original poster is attempting to receive a
facsimile, instead use voice calls.
Larry.
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_
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a=T38FaxMaxDatagram:612
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy
How can I ignore T38 and use only G729 for this call?.
Thanks for your help.
Damian
Perhaps you could add the following to the peer configuration
faxdetect=no
Larry
sure know what I'm thinking about) ?
In short, Yes!
Do it T.38 implementation works ok with Asterisk ?
Seems to though I really have only performed basic testing receiving a
fax through it from the PSTN (FXO port).
Regards,
Larry
that it would
accept to upgrade to T.38 ?
The SPA3102 only supports T.38 on the FXS port, the FXO port uses G711
for a fax session.
The Grandstream HT503 supports T.38 on both the FXO and FXS ports.
What problem do you have receiving a fax over G711?
Larry
;qualifyfreqnotok = 1 ; how frequently to ping the peer when it's
; either LAGGED or UNAVAILABLE, in
milliseconds
Larry.
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On 01/08/2013, at 2:20 PM, Zoltán Fekete bl...@gyoz.info wrote:
2013/8/1 Joshua Colp jc...@digium.com
Larry Moore wrote:
On 31/07/2013 8:08 PM, Joshua Colp wrote:
Zoltán Fekete wrote:
Thank You Larry!
I have discussed with my provider. They are not able to insert the
T38MaxBitRate
On 31/07/2013 8:08 PM, Joshua Colp wrote:
Zoltán Fekete wrote:
Thank You Larry!
I have discussed with my provider. They are not able to insert the
T38MaxBitRate value into the sip answer. :(
https://gist.github.com/anonymous/6120148 (line 559)
That means we are not able to passtrough T38
/anonymous/5701150.
What happens if you insert in your dialplan something like
Set(FAXOPT(minrate)=4800)
Cheers,
Larry.
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,
if the T38MaxBitRate attribute is omitted they suggest using the default
value, they indicate a default value of 14400.
Larry.
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On 22/07/2013 10:19 PM, Larry Moore wrote:
On 22/07/2013 5:40 AM, Zoltán Fekete wrote:
Hi!
I have exactly the same problem on asterisk 1.8.22.0 and also on
separate 11.2.1 when sending fax to PSTN.
Tryed with spa-3102, spa-2102, Patton Smartnode 4634, and Zoiper
softphone.
SpanDsp also works
On 22/06/2013 2:17 PM, Steve Edwards wrote:
On Sat, 22 Jun 2013, Larry Moore wrote:
echo $MSGFILE
printf %18s Sender: $MSGFILE; printf %-20s\n
${REMOTESTATIONID} $MSGFILE
printf %18s Pages: $MSGFILE; printf %-20s\n ${FAXPAGES}
$MSGFILE
printf %18s Signal Rate: $MSGFILE; printf %-20s\n
On 20/06/2013 2:03 AM, Daniel - Asterisk wrote:
Hello everyone,
I'm trying to send a received fax with mutt, when I try it from the
Linux shel it works, but when trying with Asterisk's System command it
doesn't.
Successful Linux command:
echo | mutt -s New fax earohua...@gmail.com
On 4/06/2013 4:53 AM, Andrey Polovov wrote:
On 06/03/2013 05:03 PM, Larry Moore wrote:
Have you checked the installed version of firmware against the latest
available from Cisco?
Oh! I didn't guess to check. The firmware was not fresh, but upgrading
doesn't help.
Looking at your SIP
On 3/06/2013 8:04 PM, Andrey Polovov wrote:
Thank you for reply, Larry!
On 06/03/2013 05:14 AM, Larry Moore wrote:
1) On SPA112 set FAX T38 Redundancy = 3
I have tried to change this value with no effect.
2) Add t38pt_usertpsource=yes in [mtt] section
This option take no positive effect
the sip.conf entry for the SPA112.
Where did t38pt_rtp t38_tcp come from?
You may also want to experiment with the SPA112 setting FAX T38 ECM Enable
Cheers,
Larry.
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extension according to your
environment.
Larry.
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My 2 cents worth.
Turn off faxdetect in the peer configuration for Asterisk.
Failing that, try the Fax Gateway feature in Asterisk 11 to Hylafax
listening on an IAX2 channel.
Larry.
On 24/01/2013 6:32 AM, Carsten Maass wrote:
Hello all,
we do have a problem here with Asterisk 11 talking T
for video calls
(default 384 kb/s)
; Videosupport and maxcallbitrate is
settable
; for peers and users as well
Larry.
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=no
May be worth checking the following;
directrtpsetup=no
Larry.
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On 28/12/2012 4:59 AM, Larry Moore wrote:
On 28/12/2012 1:55 AM, Eric Wieling wrote:
.
snip
.
directmedia=no
t38pt_udptl=no
snip
Hmm, the t38pt_udptl will need to be set to yes, this was set to no for
non T.38 capable devices
I had set faxdetect=no in the peer's configuration for the T
T.38 relaying working prior to the patch.
I have in my sip.conf;
[general]
t38pt_udptl=yes,redundancy,maxdatagram=1400
You may also want to enable;
t38pt_usertpsource=yes
Larry.
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Just a suggestion, check the current T.38 redundancy value in the SPA's,
if it isn't set to 3 then set it to 3.
If Asterisk 11 is still using udptl.conf then check the following;
udptlfecentries = 3
udptlfecspan = 3
You may also want to set;
use_even_ports = yes
Good luck.
Larry
Hi,
I'd like to replace my current VOIP provider with an Asterisk based
solution. I have some ideas I want to run by the list to see if they
are possible, and get answers to a couple questions.
Take a look at gafachi (https://www.gafachi.com/), good voice quality and
stable.
Larry
(david-test)
Seems to work with Asterisk 1.8.18.0.
I'm using extensions.ael and have tested the following;
400 = {
SIPAddHeader(Alert-Info: n=Classic-4;w=3;c=4);
Queue(400,inrt,,,30);
Hangup();
};
Larry
?
It doesn't seem to have any effect on the voice quality but the messages
on the console are quite annoying.
I suspect you will find the frequency of these messages is the value you
have set for rtpkeepalive.
I would suggest you include the following in your peer's configuration;
rtpkeepalive=0
Larry
I have the following in my intercom macro in extensions.ael;
SIPAddHeader(Alert-Info: Ring Answer);
SIPAddHeader(Call-Info:\;Answer-After=0);
SIPAddHeader(P-Auto-Answer: normal);
If memory serves me, respectively they are for the following vendors;
Grandstream
Linksys/Cisco SPA
Yealink
Larry
On 23/05/2012 10:46 AM, Ruddy Gbaguidi wrote:
I cannot find it
*From:*asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Danny
Nicholas
*Sent:* 2012-05-21 10:25
*To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
*Subject:*
the default of 64, you
would set the following
Class1PersistentECM:yes
Class1ECMFrameSize: 64
Perhaps the corruption is occurring at the senders end before the data
is pushed through the modem.
Cheers,
Larry
was received
however the responses to EOP timed out, I don't know if the is to do
with my Asterisk T.38 gateway or my VoIP providers T.38 gateway. The
result was the fax was retried for the defined number of attempts.
Cheers,
Larry.
On 16/05/2012 6:28 PM, gincantalupo wrote:
Hi all,
I'm
May 16 21:32:04.28: [ 2335]: REMOTE best 0 ms/scanline
May 16 21:32:04.28: [ 2335]: USE 9600 bit/s
Perhaps the issue is with Hylafax.
Setting the Transmit Receive strings to !24,48,72,96 seems to yield
the most reliability in transmission
Cheers,
Larry.
On 16/05/2012 7:23 PM, Larry Moore
On 17/05/2012 1:24 AM, Steve Underwood wrote:
Hi,
On 05/16/2012 09:59 PM, Larry Moore wrote:
Read the subject line more closely.
Tested receiving too,
I set the Send Receive speed of the receiving analogue modem to
that below, the log file on the sending modem (iaxmodem) reported
block to set the
attenuation, more modern devices would be configured from the front
panel, typically in a maintenance mode. Your good old dial-up modems
with fax capabilities would have an S-Register or two to set the
attenuation.
Larry
On 18/04/2012 6:39 AM, Kevin P. Fleming wrote:
On 04/17/2012 06:17 AM, Larry Moore wrote:
The send log you have posted does not show any outgoing T.38 packets
from your system.
I set up a test build of 1.8.11.0 using the patch recently released, I
have difficulties sending T.38 with this patch
when it is a different device than the SIP server it negotiates
with.
Cheers,
Larry.
On 17/04/2012 6:47 PM, Niccolò Belli wrote:
Il 17/04/2012 01:10, Niccolò Belli ha scritto:
Tomorrow I will try without directmedia=yes.
Unfortunately it didn't help.
Niccolò
is that the authentication
database at the VSP may have been offline momentarily hence why the
response of a wrong password, I wasn't convinced of this as the packet
capture of the SPA-942 did not reveal any authentication errors.
Cheers,
Larry.
On 16/04/2012 10:26 PM, sean darcy wrote:
We found
configuration?
Cheers,
Larry.
On 14/04/2012 8:33 PM, Niccolò Belli wrote:
Il 04/04/2012 07:45, Anton Kvashenkin ha scritto:
Check it out, thank you.
You're welcome.
New packages against dahdi-linux-2.6.0, dahdi-tools-2.6.0, libpri
1.4.12+svn20120409 and spandsp-0.0.6~pre20:
http
the change was?
Larry.
On 17/04/2012 4:58 AM, Niccolò Belli wrote:
Hi,
Il 16/04/2012 22:50, Larry Moore ha scritto:
Do you have directmedia=no in your SIP configuration?
Yes I have.
Niccolò
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On 13/03/2012 8:10 PM, Ishfaq Malik wrote:
On Tue, 2012-03-13 at 00:10 +0800, Larry Moore wrote:
On 12/03/2012 10:53 PM, Ishfaq Malik wrote:
Thanks for the input so far. I'm going to keep plugging away and if
anyone has any insights, they will be gladly appreciated. Ish
In SIP Account
it.
What am missing?
In your peer config set directmedia=no and faxdetect=cng, Asterisk needs
to be in the path to hear the CNG tones.
Larry.
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New
HT502 FXS Port 1
; Analogue FAX Modem attached
type=friend
defaultuser=903
secret=you_guessed_it
call-limit=2
disallow=g722
transport=udp
qualify=yes
canreinvite=no
directmedia=no
host=dynamic
context=FAX-T38
faxdetect=no
Larry
On 12/03/2012 10:53 PM, Ishfaq Malik wrote:
Thanks for the input so far. I'm going to keep plugging away and if
anyone has any insights, they will be gladly appreciated. Ish
In SIP Account Configuration on Draytek;
Set Voice Active Detect to Off
In Phone Settings on the Draytek;
Enable
=allow, this permits the administrator of the
UA to decide if it should use SRTP or otherwise traditional RTP is used.
So is it possible with asterisk.
Yes!
Was that one question!?
Larry.
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:-(
The Fallback option to T.30 is 'f'.
ReceiveFAX(filename,f)
See
https://wiki.asterisk.org/wiki/display/AST/Application_ReceiveFAX+%28res_fax%29
Larry.
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On 22/09/2011 4:12 AM, Ian Pilcher wrote:
I am looking for a simple way to send occasional faxes via the FXO
port on my SPA3102 -- without having to connect a fax modem to an
ATA. In an ideal world, this would be some sort of softfax that
runs on my Linux desktop and talks (via Asterisk) to the
On 5/09/2011 4:27 PM, Jonas Kellens wrote:
Hello,
I'm trying to page the Cisco SPA 941 by adding the SIP-header
Call-Info: answer-after=0
dialplan :
exten = _*XX*,n,SIPAddHeader(Call-Info: answer-after=0)
Try
exten = _*XX*,n,SIPAddHeader(Call-Info:\;Answer-After=0)
Larry
will need to make to your SIP peer is to set t38pt_udptl=yes
and in your dial plan before the Dial() enable the gateway with
Set(FAXOPT(t38gateway)=yes).
Larry.
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Larry.
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On 1/09/2011 7:04 PM, Tim King wrote:
I have found numerous claims that 1.8 can do T.38 gateway with a
patch, however I am yet to find the patch or any instructions on
implementing it. Anyone have a link?
https://issues.asterisk.org/view.php?id=13405
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the phone can be provisioned with a defined
time zone offset or accept the offset in DHCP is a matter of further
research.
Larry.
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due to the benefits tlsauth offers
against DoS.
I have used a Yealink T22 accross an IPSEC VPN using TLS Auth however I
have since configured it to connect directly via the Internet.
I have been keeping the devices firmware updated as they are released.
My two-bobs worth!
Larry
/255.240.0.0
permit=192.168.0.0/255.255.0.0
Larry.
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that one
should have a PSTN line connected directly to a fax device at CPE for
receiving said communications and one could use T.38 for Outbound faxing
providing the transmissions are of high enough quality
Larry..
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work. The key to my success was to ensure the SPA8800 did not do a
re-invite to the ISP for the RTP stream.
Larry.
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On 28/06/2011 6:59 PM, Matteo Campana wrote:
Hi Larry,
I have the SIP debug taken from asterisk.
In this debug: 1.2.3.4 --- IP SIP PROXY
5.6.7.8 --- IP UAC (Linksys SPA 962)
9.10.11.12 --- IP ASTERISK to connect to the
provider
' of the dialplan or the main part of the macro after ReceiveFAX(),
was when a T.38 fax was being received, when it was a G.711 fax no
matter what I did to the call it would always execute the System() call
whether it was in the macro or the 'h'.
Cheers,
Larry
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