[asterisk-users] Problem with ReceiveFax

Kevin P. Fleming kpfleming at digium.com
Tue Mar 13 07:13:22 CDT 2012


On 03/13/2012 07:10 AM, 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?

Yes. In Asterisk, a DTMF 'f' is the FAX CNG tone emitted by calling FAX 
terminals.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kfleming at digium.com | SIP: kpfleming at digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org



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