Displaying 20 results from an estimated 5000 matches similar to: "Pattom M-ATA, T.38 and Asterisk 1.4. Canreinvite=yes ? [SOLVED]"
2009 Feb 05
0
Pattom M-ATA, T.38 and Asterisk 1.4. Canreinvite=yes ?
Hi,
Here http://www.voip-info.org/tiki-index.php?page=Asterisk%20T.38 is a table
listing ATA/Gateways combinations.
Could anyone successfully set a Patton M-ATA to work with another one, using
Asterisk 1.4 ?
Is reinvite (canreinvite=yes) necessary or not ?
Regards
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2009 Mar 17
0
ATA react to phone but unresponsive to fax modem [SOLVED]
2009/3/17 Olivier <oza-4h07 at myamail.com>
>
>
> 2009/3/16 Olivier <oza-4h07 at myamail.com>
>
> Hi,
>>
>> I'm rather new to this domain so I may be doing stupid things without
>> being concious of that.
>>
>> I've got a Patton MATA I'm trying to setup as T.38 fax adapter.
>> Whenever I connect a fax machine (Dell
2009 Feb 05
0
Patton M-ATA and T.38
Hi,
Has someone met success setting a Patton M-ATA to work in T.38 ?
In my experiences here, it seems this ATA don't switch to T.38 whenever a
fax signal is heard on its FXS port.
Regards
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2013 Jul 21
2
Fwd: Re: Asterisk T.38 Pass-Through doesn't work
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 without any problem on my box.
As I remember it was a bug in 1.8.1.x that the a=T38MaxBitRate paramater
was sent as "maxBitRate". Without capital "M".
Are you closer to
2009 Feb 26
0
Patton 5.3. How to get incoming calls ? [SOLVED]
Hi,
Changing the line bellow helped to get incoming calls but I add to remove
secret= option in sip.conf (otherwise, Patton wouldn't respond to 407 Auth
required challenges).
If someone could enable secret and still get incoming calls (in any
SmartWare 5.X), please, do not hesitate to share here ...
interface sip IF-ASTERISK
bind context sip-gateway ASTERISK
route call dest-table
2009 Mar 09
0
How to install spandsp from source in lenny ? [SOLVED]
2009/3/9 James Sneeringer <jsneerin at gmail.com>
> On Mon, Mar 9, 2009 at 11:13 AM, Olivier <oza-4h07 at myamail.com> wrote:
> > Anyway, whenever I'm typing make menuselect, app-fax is greyed out as in
> my
> > opinion, spandsp libriaries have not been found.
> >
> > Maybe, I should have typed something like (as suggested
> >
2009 Mar 16
1
ATA react to phone but unresponsive to fax modem
Hi,
I'm rather new to this domain so I may be doing stupid things without being
concious of that.
I've got a Patton MATA I'm trying to setup as T.38 fax adapter.
Whenever I connect a fax machine (Dell MFP1815dn) or a phone to it, I can
successfully send a fax or talk to the other end.
Whenever I connect a fax modem (Dell Inspiron 6400 laptop), I keep getting
"No signal. Line is
2009 Jan 16
0
No subject
"In computer software standards and documentation, the term deprecation =
is=20
applied to software features that are superseded and should be avoided.=20
Although deprecated features remain in the current version, their use =
may=20
raise warning messages recommending alternate practices, and deprecation =
may indicate that the feature will be removed in the future. Features =
are=20
2009 Feb 09
1
What t38pt_udptl is ? Explain T.38 in 1.4
Hi,
I would like to improve my understanding of T.38.
1. What T38FAX_VERSION_0 or T38FAX_VERSION_1 in chan_sip.c means ?
voip-info.org implies one has to change values in chan_sip.c to make it
work.
Shall I set T38FAX_VERSION_1 or leave T38FAX_VERSION_0 in
global_t38_capability ?
Source code says "This is default: NO MMR and JBIG trancoding, NO fill bit
removal, transferredTCF TCF, UDP FEC,
2010 Oct 08
3
looking for a better ATA
I currently us Linksys/Ciscio, Grandstream and AudioCodes ata's. none of
the three perform well in all enviroments. Between stablity issues, T38 and
DTMF talkoff all three suffer some combination of issues.
I am looking at Patton and Innomedia. Has any one tried either brand and
what is your experience with them. Which would be the base for stability,
audio quality, provisioning, DTMF
2009 Jan 27
0
Can't start Asterisk after installing Digium G729 licence [SOLVED]
2009/1/27 Olivier <oza-4h07 at myamail.com>
>
> 2009/1/27 Olivier <oza-4h07 at myamail.com>
>
> Hi,
>>
>> I carefully followed instructions in README file lasting with :
>> /root/register
>> ... blabla
>> asterisk -r
>> CLI> restart now
>>
>> Then asterisk -r fails with :
>> # asterisk -r
>> Asterisk
2009 Feb 26
5
ATA recommendation (wih FTP provisioning)
Hi,
I am looking for a good ATA recommendation, ideally something:
1) with one FXS and one LAN port (so it's as inexpensive as possible)
2) That can be provisioning using FTP (configuration and firmware upon
reboot, ideally remote reboot from a sip notify)
3) Supports T.38
Nice to have would be:
a) PoE powered and AC powered (my choice)
b) Small size-wise
I have been
2005 Sep 19
3
T.38 & Canreinvite (yes, again)
I know this has been asked before, but I've checked the archives and I
haven't found anybody that has given a definitive yes or no, just "yeah,
it should work.....". If I have a T.38 gateway like a Cisco 5300 and a
T.38 ATA (whatever model) and I have canreinvite=yes, should T.38 work?
I have it setup and it doesn't work, so I want to know if I am doing
something wrong,
2008 Dec 02
2
1.6, t.38 and zoiper - t38_udptl or t38pt_udptl ?
Hi,
1. Has anyone got any success when send a TIFF file form one zoiper
softphone to another ?
I tried using Zoiper 2.18 free edition in windows but I'm seeing 415
Unsupported media replies.
2. Here (http://www.voipinfo.org/wiki/view/Asterisk+T.38), you can read :
"Also, try using:
t38_udptl=yes
t38pt_rtp=no
t38pt_tcp=no
... in the general section of the sip.conf and under the VoIP
2009 Mar 09
0
Can't build today's AGX Asterisk Addon with spandsp0.0.6pre3 or 4 [SOLVED]
2009/2/26 Olivier <oza-4h07 at myamail.com>
> I must add I tried spandsp0.0.6xxx as a warning message advised me to do so
> (using 0.0.4 would be ok for me but current trunk doesn't allow this
> anymore, it seems).
>
>
> 2009/2/26 Olivier <oza-4h07 at myamail.com>
>
> Hi,
>>
>> With 0.0.6pre3:
>> # ./build.sh
>> CMake Warning (dev)
2007 Jul 12
0
No subject
Leg/Transaction Does Not Exist" and obviously not taken into account as
endpoint GUI remains unchanged.
Looking deeper into this here are :
NOTIFY message accepted by S450IP
NOTIFY sip:7531 at 192.168.100.197:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.254:5060;branch=z9hG4bK06adc48b;rport
From: "asterisk" <sip:asterisk at asterisk>;tag=as4ea953db
To: <sip:sip:7531 at
2015 Feb 02
0
Asterisk 13, PJSIP and T38 problem
Hello,
I need help to solve a problem that I am having using Asterisk 13, PJSIP and T38.
My setup is as follows:
SIP Provider --> Asterisk 13 --> Patton --> Physical Fax
I need to get the fax directly in T38 to Patton.
The provider sends me the fax in T38.
If I receive the T38 fax on Asterisk (using an hylafax device), I can properly receive the fax.
If I send a T38 fax with Asterisk
2010 Apr 15
0
Regarding remote registration of SIP user on zoiper
Hello list,
I am new to this list and have been using Asterisk as part of my research project for about 2 weeks now.
I would like to get your thoughts on a scenario that I am attempting at the moment. I haven't had luck until now.
In this scenario, I am trying to register a SIP user configured on the zoiper client installed on a laptop, which is on the same Local Area Network, with the
2009 Mar 10
5
Sending faxes with T.38 problem. Asterisk - 1.6.0.6
Hello,
I'm having difficulties to make Asterisk (1.6.0.6) interoperate correctly
with a CISCO mediaGW in order to send faxes to the PSTN using T.38.
When Asterisk sends the initial INVITE containing the T.38 media offer in
the SDP, the CISCO answers with a 488 Not Acceptable Media.
Apparently, it looks like a configuration problem in the CISCO, but I have
tested the CISCO with the Zoiper
2009 Feb 25
0
Patton 5.3. How to get incoming calls ?
Hi,
I'm trying to configure a 4638 to pass inbound and outbound to and from ISDN
and SIP interfaces.
I'm using web interface at the moment.
Setup is:
ISDN -- <BRI> -- Patton 4638 -- <SIP> Asterisk -- <SIP> -- <IP Phone>
I can call from IP phone but can't receive any incoming call : I can't see
any SIP message coming in when a call comes in.
Previously,