[asterisk-users] Digium FFA + Gafachi T38 outgoing issues

[Nasir Iqbal]

for which user/number sip reinvite is for? ooh! you are running a direct
application without any dialplan or user, may be that is the cause!  I think
you should first write fax dialplan, reload asterisk and test again with
originate but this time with extension not application.

Nasir Iqbal

ICT Innovations
http://www.ictinnovations.com/



On Sat, Oct 8, 2011 at 12:20 AM, James Sharp <james at fivecats.org> wrote:

> On 10/07/2011 12:27 AM, Nasir Iqbal wrote:
>
>> Check firewall and NAT settings!
>>
>> Monitoring sip and media flow from asterisk cli can help, use "sip set
>> debug on", "rtp set debug on" and "udptl set debug on"
>>
>>
> No NAT involved and I shut off IPTables.  Still no luck.  Debug shows the
> SIP invite, RTP frames going in & out, the SIP reinvite, and then UDPTL
> frames coming in until timeout.
>
> See the entire transaction at http://pastebin.ca/2087758
>
>
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