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 Configuration on Draytek;
Set Voice Active Detect to Off
In Phone Settings on the Draytek;
Enable Symmetric RTP
Check Start& End RTP Ports match values set in /etc/asterisk/udptl.conf
for udptlstart& udptlend
In /etc/asterisk/udptl.conf set;
use_even_ports=yes
Thanks for the above, I was hoping to have replied earlier with a
success message buy alas, no joy to be had.
Could I be having some sort of DTMF issue? I noticed this in amongst the
console output once I set the console logging level to include dtmf
[2012-03-13 12:06:39] DTMF[24784]: channel.c:3976 __ast_read: DTMF end 'f'
received on SIP/588-0000000c, duration 0 ms
[2012-03-13 12:06:39] DTMF[24784]: channel.c:4002 __ast_read: DTMF begin
emulation of 'f' with duration 100 queued on SIP/588-0000000c
[2012-03-13 12:06:39] DTMF[24784]: channel.c:4138 __ast_read: DTMF end
emulation of 'f' queued on SIP/588-0000000c
does the above look correct for an inbound fax?
Thanks in advance (again!)
Ish
It's now time to do some debugging.
I would suggest you capture packets between asterisk and peer 588 using
tcpdump, make sure you enable a large enough snaplen (-s) to ensure you
capture all packets in the frame.
Submit your fax and upon completion of the session whether or not it is
received successfully, transfer the file where you can open the captured
file in Wireshark and select VoIP Calls located in the Telephony menu.
You can then select the relevant line or lines in the session and click
on the "Flow" button and review what is happening.
I have a Grandstream HT-503 at the other end of an IPSEC vpn which has
the FXO port connected to a PSTN line.
I have configured the HT-503 to call the fax extension in the dialplan
when it answers a call hence I have disabled faxdetect in the peer
configuration.
Looking at the Draytek manual I think this would be setup in VoIP >>
Phone Settings by enabling Call Forwarding and setting it to "Always"
and defining the SIP URL as fax@<astersk_server_ip>, assuming you have a
fax extension enabled in the context of the peer. I am assuming you
currently have this set to 200@<astersk_server_ip>.
Did you disable VAD on the Draytek.
I would also suggest you disable "Call Waiting" & "Call Transfer".
You may also want to look at "Volume Gain" in case that affects the
level of the signal being converted to T.38 on the Draytek. Testing by
progressively decreasing the level and if that doesn't help then
increasing it.
Here is the peer configuration I just tested with my HT-503.
T.38 is enabled in the [general] section of sip.conf
[0123456789]
type=peer
defaultuser=0123456789
secret=you_guessed_it
call-limit=2
host=dynamic
disallow=g722
g726nonstandard=yes ;(this is required for Sipura and
Grandstream ATAs, among others).
transport=udp,tcp
encryption=no
directmedia=no
faxdetect=no
context=Fax-Test
qualify=yes
Good luck.
Larry.
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