Cyprus VoIP wrote:
Yes. I saw the message and the required addition in the sip.conf. The
problem is that if I set it to 72, other terminating gateways that
support 400 or more would also be limited to 72.
This is incorrect. First, you would not set it to 72, since the endpoint
is already
Cyprus VoIP wrote:
This is the reINVITE SDP received from the SIP Proxy:
---
Content-Type: application/sdp
Content-Length: 353
v=0
o=root 30427 30428 IN IP4 194.98.xxx.xxx
s=session
c=IN IP4 194.98.xxx.xxx
t=0 0
m=image 17548 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
-users] Fax throughput - Asterisk 1.6.1.9
From: Cyprus VoIP voi...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Friday, 04 December, 2009 18:21:59
It's probably because you are using 1.6.1.9; that release (and older)
had a 'feature
Cyprus VoIP wrote:
This is the reINVITE SDP received from the SIP Proxy:
---
Content-Type: application/sdp
Content-Length: 353
v=0
o=root 30427 30428 IN IP4 194.98.xxx.xxx
s=session
c=IN IP4 194.98.xxx.xxx
t=0 0
m=image 17548 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
Cyprus VoIP wrote:
This is the reINVITE SDP received from the SIP Proxy:
---
Content-Type: application/sdp
Content-Length: 353
v=0
o=root 30427 30428 IN IP4 194.98.xxx.xxx
s=session
c=IN IP4 194.98.xxx.xxx
t=0 0
m=image 17548 udptl t38
a=T38FaxVersion:0
Cyprus VoIP wrote:
Thank you for your answer. The 'internal extension' is indeed a T.38
capable device that works perfectly when connected directly to the
Proxy/ITSP.
As you said, the key to debugging/resolving this issue is the logger. I
wasn't aware of this file. this is what I have
Set 'canreinvite=no' on all applicable peers?
I tried with yes and no. No difference. I'm almost certain it's related
to the Keeping RTP active during T.38 session issue.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Cyprus VoIP wrote:
So, I enabled the full logger, and the strange thing I see is this message:
Got T.38 Re-invite without audio. Keeping RTP active during T.38 session
It seems that this might be the reason Asterisk initiates a reINVITE
with voice codecs, after connecting the 2 parties.
Cyprus VoIP wrote:
So, I enabled the full logger, and the strange thing I see is this message:
Got T.38 Re-invite without audio. Keeping RTP active during T.38 session
It seems that this might be the reason Asterisk initiates a reINVITE
with voice codecs, after connecting the 2 parties.
Cyprus VoIP wrote:
If it's not related, why does Asterisk send again INVITE messages to
both parties? How can this be prevented? I don't see more debug data
prior to the new INVITE.
It's probably because you are using 1.6.1.9; that release (and older)
had a 'feature' that allowed automatic
It's probably because you are using 1.6.1.9; that release (and older)
had a 'feature' that allowed automatic switching back to audio from T.38
if one of the endpoints sent an audio packet. It turns out that wasn't a
good idea, and it's been removed... but in later versions. You'll have
to
Hello,
We are trying to send faxes by T.38 protocol to a remote SIP proxy from
a local extension. The local extension sends the INVITE, Asterisk sends
the call to the Proxy the call is connected with a regular audio codec.
After a few seconds the remote proxy sends an INVITE with UDPTL and the
Cyprus VoIP wrote:
We set t38pt_udptl=yes in sip.conf and allowed all the codecs to the
local extension and remote Proxy, but it still forces the call to go
back to a voice call.
Define 'internal extension'. Is this a T.38-capable device? If not,
Asterisk doesn't support TDM-to-T.38 FAX
Cyprus VoIP wrote:
We set t38pt_udptl=yes in sip.conf and allowed all the codecs to the
local extension and remote Proxy, but it still forces the call to go
back to a voice call.
That's correct behavior if T.38 cannot autonegotiate.
What happens in the reverse direction, trying to send faxes
We set t38pt_udptl=yes in sip.conf and allowed all the codecs to the
local extension and remote Proxy, but it still forces the call to go
back to a voice call.
Define 'internal extension'. Is this a T.38-capable device? If not,
Asterisk doesn't support TDM-to-T.38 FAX relay (yet). If it
Cyprus VoIP wrote:
Thank you for your answer. The 'internal extension' is indeed a T.38
capable device that works perfectly when connected directly to the
Proxy/ITSP.
As you said, the key to debugging/resolving this issue is the logger. I
wasn't aware of this file. this is what I have
Set 'canreinvite=no' on all applicable peers?
Cyprus VoIP wrote:
Hello,
We are trying to send faxes by T.38 protocol to a remote SIP proxy from
a local extension. The local extension sends the INVITE, Asterisk sends
the call to the Proxy the call is connected with a regular audio codec.
On Wed, 27 Jun 2007 09:08:21 -0500, Matthew Fredrickson wrote:
You fixed your clocking then. That was what I was thinking of. Make
sure that your Dialogic card is also pulling timing from the Digium
card as well. What version of zaptel drivers are you running?
on a related issue, using
- Non-Commercial
Discussion
Subject: Re: [asterisk-users] Fax Throughput
Can you post your zaptel.conf so we can verify your timing settings?
---
Matthew Fredrickson
Software Engineer
Digium, Inc.
On Jun 25, 2007, at 11:10 PM, Don Kelly wrote:
I've tried timing faxes two ways:
From
Don Kelly wrote:
I've tried timing faxes two ways:
From a fax machine on a station port of an AltiGen PC/PBX served by an MCI
PRI calling back into the same PRI and reaching a RightFax server on a
station port behind the AltiGen.
From the same fax machine on the same station port of the AltiGen
Can you post your zaptel.conf so we can verify your timing settings?
---
Matthew Fredrickson
Software Engineer
Digium, Inc.
On Jun 25, 2007, at 11:10 PM, Don Kelly wrote:
I've tried timing faxes two ways:
From a fax machine on a station port of an AltiGen PC/PBX served by
an MCI
PRI
for your Virtual Office
651 842-1000
888 Don Kell(y)
651 842-1001 fax
-Original Message-
From: Matthew Fredrickson [mailto:[EMAIL PROTECTED]
Sent: Tuesday, June 26, 2007 9:22 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Fax
I've tried timing faxes two ways:
From a fax machine on a station port of an AltiGen PC/PBX served by an MCI
PRI calling back into the same PRI and reaching a RightFax server on a
station port behind the AltiGen.
From the same fax machine on the same station port of the AltiGen PC/PBX
served by
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