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 thus im now getting
Feb 3 12:32:55] WARNING[9942][C-00000004]: chan_sip.c:10353
process_sdp:
Failed to initialize UDPTL, declining image stream
[Feb 3 12:32:55] WARNING[9942][C-00000004]: chan_sip.c:10497
process_sdp: Insufficient information in SDP (c=)...
and then the fax session starts recording data
In udptl.conf set use_even_ports=yes and then issue a reload.
You can confirm the settings have been applied by performing udptl
show config.
Change the the t38 line to read as;
t38pt_udptl=yes,redundancy,maxdatagram=400
Reload sip and test.
after that i started udptl debug as well and now i'm getting lots of
UDPTL (SIP/sipcall.ch-00000007): packet to 212.117.203.76:24492 (seq
152, len 11)
and in between
[Feb 3 13:18:30] WARNING[26313][C-00000007]: res_rtp_asterisk.c:3548
ast_rtp_read: RTP Read too short
and in the end
[Feb 3 13:18:37] WARNING[9942]: chan_sip.c:4409 __sip_autodestruct:
Autodestruct on dialog
'[email protected]~1o' with owner
SIP/sipcall.ch-00000007 in place (Method: BYE). Rescheduling
destruction for 10000 ms
[Feb 3 13:18:41] ERROR[26313][C-00000007]: res_fax.c:1535
generic_fax_exec: channel 'SIP/sipcall.ch-00000007' FAX session '7'
failure, reason: 'fax session timed-out' (TIMEOUT)
== Spawn extension (fax-rx, receive, 11) exited non-zero on
'SIP/sipcall.ch-00000007'
Thx, Jakob
may do i have to open more ports then udp 10000:20000 (RTP), udp
4000:4999 (UDPTL) and tcp 5060,5061(SIP/TLS)
The T.38 connection will be attempted when ReceiveFax is called.
The port number to use should be in the SDP information, yes, allow udp
ports 4000-4999 in and out. If your firewall can be so configured you
could set it to allow traffic in and out based upon the user ID Asterisk
is running as, assuming it is using a unique unprivileged id.
You may like to try the following to see if your SIP provider will
initiate a T.38 re-invite.
sip.conf
[general]
faxdetect=t38
[sipcall.ch]
directmedia=no
In extensions.conf change Wait(2) to Wait(5), if your VSP sends you a
T.38 re-invite this will trigger the switch to the Fax extension.
If this proves successful you can work on removing the Wait() from your
dialplan as Asterisk will remain in the audio path and should
successfully switch to the fax extension if extension 200 or 201 answer
a call that happens to be a fax.
Larry.
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
