asterisk users - Nov 2004 - Max retries exceeded to host ...

Hi,

I'm using firefly to connect from a NATed network to a NATed asterisk
server, the lag between them is about 260-300 ms.

The problem is that the calls regularly hangup after a message like that :

chan_iax2.c:1139 attempt_transmit: Max retries exceeded to host
x.x.x.x on IAX2[silvana@silvana]/3 (type = 6, subclass = 11, ts=70010,
seqno=15)

What could be the reason of the hangup ?
I've searched in google and in the wiki but nothing seems appropiate.

If needed I can post more information on the configurations

Excuse my english, its not my language.

Regards,
Fernando

I've noticed this quite a bit recently as well. Also having major
problems with consistency registering with FWD. Is this a problem on
the IAXTel/FWD side, or is it possibly related to my configuration?

PF

On Wed, 17 Nov 2004 13:09:10 -0200, Fernando Pieri <fpieri@gmail.com>
wrote:
> Hi,
> 
> I'm using firefly to connect from a NATed network to a NATed asterisk
> server, the lag between them is about 260-300 ms.
> 
> The problem is that the calls regularly hangup after a message like that :
> 
> chan_iax2.c:1139 attempt_transmit: Max retries exceeded to host
> x.x.x.x on IAX2[silvana@silvana]/3 (type = 6, subclass = 11, ts=70010,
> seqno=15)
> 
> What could be the reason of the hangup ?
> I've searched in google and in the wiki but nothing seems appropiate.
> 
> If needed I can post more information on the configurations
> 
> Excuse my english, its not my language.
> 
> Regards,
> Fernando
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>

Hi

I posted this on the ser list also, not sure what the best place for the 
question

--
  I have ser front ending asterisk, and all calls except sip-->sip go 
via asterisk, which then connects to a cisco box. Now I want to do 
inbound from pstn, is is better to go again via ser, and then route 
either to sip device or to alternate pstn number via asterisk, or push 
directly into asterisk, I think the former...any suggestions, or config 
examples, did find on wiki

Iqbal