On 4/06/2013 4:53 AM, Andrey Polovov wrote:
On 06/03/2013 05:03 PM, Larry Moore wrote:
Have you checked the installed version of firmware against the latest
available from Cisco?
Oh! I didn't guess to check. The firmware was not fresh, but upgrading
doesn't help.
Looking at your SIP information when your ITSP initiated a T.38
session it did not indicate a maxmimum bitrate, it would appear your
spa112 attempted to negotiate a connection at 2400bps.
Whether there is a way to force my provider to indicate maximum bitrate?
Do you have a sip debug session when you sent a fax from your Asterisk
box to the PSTN, it would be interesting to see if it sends it as a
t.38 or reverts to G711 audio.
I have collect a set of debugs (with fresh SPA112 firmware) and actual
config files:
I would suggest you test the SPA112 directly against your SIP provider
however ensure you have the latest firmware, there appear to have been
some FAX related fixes in recent versions.
To do this change the following (based upon the screen shot you made
available);
Nat Mapping Enable: yes
Call Waiting Serv: no
MWI Serv: no
Proxy: 80.75.130.136
Register: no
Make Call Without Reg: yes
Ans Call Without Reg: yes
User ID: 74957777777
Password: <remotesecret>
FAX T38 Redundancy: 3
FAX Tone Detect Mode: callee
FAX T38 Return to Voice: no
When you get this working you can then look at making it work through
Asterisk - this is how I got my SPA8800 working.
I am assuming your network configuration is set up correctly on the spa112.
You may want to look at enabling the following options, on my SPA8800
they are located under the SIP tab in the section headed NAT Parameters:
Handle via rport: yes
Insert via rport: yes
Send Resp to Src Port: yes
In addition, once it is working and providing your SIP provider permits
ECM through a T.38 service I would encourage one to enable options such
as FAX T38 ECM Enable.
If you are still experiencing problems you may want to perform a packet
capture (set the snap size to 1500) of the communications between the
spa112 and the other end point and run it through Wireshark and examine
the VoIP calls in the captured packets.
Good luck.
Larry.
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