Displaying 20 results from an estimated 40000 matches similar to: "T.38 & Canreinvite (yes, again)"
2009 Feb 05
0
Pattom M-ATA, T.38 and Asterisk 1.4. Canreinvite=yes ? [SOLVED]
2009/2/5 Olivier <oza-4h07 at myamail.com>
> 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
>
>
Replying to myself, I
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
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
2007 Feb 14
6
Fax with T.38
Hi all,
I install the last version of Asterisk and I tried to send faxes, but
nothing works.
Here is my configuration:
Analog Fax <----> IP <----> Asterisk <----> IP <----> Patton M-ATA
<----> Analog Fax 2
I tried Analog Fax 2 -> Analog Fax but nothing works!!
In the Patton configuration I put G711 and no silence suppression.
In asterisk I have
2003 Sep 11
2
SIP client<->NAT<->Asterisk<->NAT<->SIP client. only works with canreinvite=no.
Hi!
I have this configuration:
SIP client A <-> NAT box A (real external IP) <-> Asterisk server (real
IP) <-> (real external IP) NAT box B <-> SIP client B
The echo test form any of the clients to the asterisk server is working
just fine, even without canreinvite=no.
When I try to call from SIP client A to B, wihtout the canreinvite=no in
the sip.conf, the call
2006 Jan 23
3
canreinvite always =no * no matter what we try :-(
been testing with a rather simple setup.
The mission is to actually get a reinvite to work on the lan.
I am trying with two sipura phones G.711 codec forced on both
both on the lan no nat no fancy options suchs as tT or H
No matter what we do asterisk hangs on to the media path, how
in the world do I get a reinvite to work where the media path
is actually handled by the two phones on the lan?
2004 Jan 05
2
Codec Negotiation Does not seem to work as expected ?? Help Please !!
Hello,
I have been trying to get my coders to work without a conversion. I have
read all the available asterisk documentation and support groups without
any luck. Here is my issue. (Please feel free to ask questions if you do
not understand what I am talking about.)
I am using Cisco ATA-186 set to g729 codec. (But it will switch to g711 if
sip-server request g711)
I have 2 SIP-services to
2005 Feb 17
4
functional difference: canreinvite=yes, no, or update
Can anyone give an example of the difference between the following:
canreinvite=no
canreinvite=yes
canreinvite=update
Here is the problem: I have an 800 number sent to me via SIP from a national
carrier. Asterisk gets the number and rings my desk phone. Asterisk has 2
NICs, one with public IP and private IP. My phone is on private IP, the
inbound call is on public.
My phone rings and I answer
2006 Oct 23
2
T.38 faxing with spandsp and Grandstream HT.486
Hello !
I 'm trying T.38 faxig with spandsp using rxfax/txfax as fax terminal.
As another endpoint I 'm using Grandstream HT 486 ATA and a fax machine.
Has anybody success with the HT486 as T.38 terminal ?
ATA as originator: I managed only onetimes a successfull T.38 fax
session. The other times the HT486 did not initiate a re-invite with
T.38 parameters. Or shall the Terminator
2005 Sep 14
6
T.38 ATA
Hello all !
Can anyone recommend me ATA device that REALLY has T.38 built in.
So far I have heard of Telco Systems Access201, which seems to be
impossible to bye in Europe (all resselers are droped Telco systems ATAs for
some reason (tried in Germany and in UK so far)), and I have heard that
SIPURA SPA-2100 should have T.38 built in into newer firmware, but I wasn't
able to confirm that
2013 Jun 02
1
Asterisk T.38 Pass-Through doesn't work
What I have is:
* Asterisk 1.8.10.1~dfsg-1ubuntu1,
* SPA112 ATA with analog fax in 1-st FXS port connected,
* SIP trunk with provider supporting T.38.
My network looks like this:
* spa112 (192.168.33.200/24) and Asterisk (192.168.5.253/24) in
neighbouring LANs,
* Asterisk connects to the provider (80.75.130.136) via router
(82.200.7.184). Router has full DNAT to Asterisk server.
What happens?
2009 Jun 08
3
T.38 pass-through 488 handling problem
Hi!
I have the following problem with Asterisk 1.4.23:
ATA w/ T.38 Asterisk ATA w/o T.38
--------INVITE-------->
--------INVITE-------->
<-------200OK----------
<-------200OK----------
--------ACK----------->
--------ACK----------->
--------INVITE
2005 Aug 23
3
Music On Hold + canreinvite=yes
For canreinvite=yes to work, I think I need to remove the t argument in
the Dial(SIP/ext|60|t) application. Otherwise, Asterisk will allways
stay in the middle. I don't want that, so I removed the 't' argument.
That works. Now, when two UA are calling, Asterisk gets out of the RTP
stream. However, when removing the 't' argument, the Music On Hold
doesn't work anymore
2011 Feb 10
2
Unable to make outgoing calls with Internode
Surely there must be someone here who can help me with this problem.
I have spent weeks trying to get this damned service to work with no
luck. I have incoming calls working, but no outgoing. If get outgoing
working then incoming don't work.
I have sent this problem to this list a couple of times with little or
no response, and I _really_ need some help to sort it out.
I have an asterisk
2008 Dec 03
3
canreinvite=yes problem
Hello,
I need to test canreinvite=yes with 2softphones and 1 asterisk.
I want to have that :
http://www.panoramisk.com/wp-content/uploads/2007/05/asterisk-call-flow-outb
ridge.png
But I have that http://www.zimagez.com/zimage/canreinvite.php
Canreinvite=yes work for all phones or just asterisk?...
Can you help me?
Thank you
-------------- next part --------------
An HTML
2005 Mar 08
1
SIP - Call Park/Pickup and Canreinvite=yes at the same time??
Hi all,
I am trying to use canreinvite in sip.conf and park/pick up calls at the
same time.
Problem:
When I have it set up so RTP goes through asterisk (sip.conf:
canreinvite=yes), # to xfer works fine. But, when I set it up so the RTP
goes direct between endpoints (sip.conf: canreinvite=no), the # to xfer
doesn't work. I believe this is because asterisk isn't in the RTP path and
2003 Sep 13
1
Caller-ID name delivered in double-quotes
I did some searching in the archive, but found only one message with
this same question and no answer. Hopefully it's a simple config problem.
When the Caller-ID is delivered, it is surrounded by double-quotes,
like this:
"ATA-57 1"
On long caller-id strings, the last character is cut off to make room
for the leading double-quote:
"BudgeTone 1234
instead of
BudgeTone
2003 Nov 25
3
Handytone 286 - calling out
Hi,
Just received recently released Grandstream handytone 286 ATA for
testing.
I can call ATA from any other extensions and conversations seems to be
of quite good quality. However placing calls from ATA is not possible at
all to any extensions.
After dialing there no indications of any kind from ATA at all. It just
"hangs in there".
ATA is behind NAT, registers to an * with public IP
2014 Nov 17
2
OT - Is T.38 possible on SPA8800 FXO port ?
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 ?
Regards
2007 Jan 17
2
AbsoluteTimeout with canreinvite=yes
Is AbsoluteTimeout designed to work with canreinvite=yes?
If not, are the any other options for disconnecting a call after a
predefined duration when using canreinvite=yes?
Thanks!
David
2005 Jun 14
1
canreinvite=yes not working with sipura device.
I'm trying to get canreinvite=yes to work. I would like
asterisk to release the line and let the 2 ports on the sipura
device to talk to each other directly. Is there a setting
I need to activate on the sipura device, or is there something
else I need to do? It's possible that it is a nat problem as the
sip device is behind a firewall, but it works fine otherwise.
Any suggestions?