[asterisk-users] Phone -> NAT/FIREWALL -> Internet -> NAT/Firewall-> Asterisk

[Adam Moffett]

top posting is superior anyway --- *ducking to avoid thrown objects*

If I recall correctly, when doing something like that with a polycom I 
had to set the registration interval absurdly low, like 20 seconds or 
something.  I think the Polycom didn't send keepalives and that was the 
workaround.


> top posting so as to not make thread even more confusing.
>
> Nick,
> I have nat=force_rport,comedia in sip.conf.  It is my understanding that
> nat=yes is deprecated?
>
> Thanks,
> JohnM
>
>
> On 01/02/2014 10:51 AM, Nick Olsen wrote:
>> Make sure you have nat=yes in your sip.conf either under globals or
>> individual sip peer settings.
>>
>> Nick Olsen
>> Network Operations
>> (855) FLSPEED  x106
>>
>>
>>
>> ------------------------------------------------------------------------
>> *From*: "John Millican" <john at millican.us>
>> *Sent*: Thursday, January 02, 2014 10:50 AM
>> *To*: "Asterisk Users Mailing List - Non-Commercial Discussion"
>> <asterisk-users at lists.digium.com>
>> *Subject*: [asterisk-users] Phone -> NAT/FIREWALL -> Internet ->
>> NAT/Firewall-> Asterisk
>>
>> Hello,
>> CentOS 6.x and Asterisk 11.x
>> I have an interesting, to me at least, situation. Using a Polycom
>> 501(also tried with X-Lite). I have set up Asterisk to accept
>> registration from the Polycom and it registers successfully but then
>> withing 30 seconds on the CLI I get the message that the Polycom is
>> unreachable. The phone still shows that it is registered and if I try
>> to place a call from the phone to my Cell, my cell rings once and then
>> stops. I get a packet retransmission error:
>> WARNING[1303]: chan_sip.c:4174 retrans_pkt: Retransmission timeout
>> reached on transmission 689874757 at 192.168.0.100 for seqno 2 (Critical
>> Response)
>> Followed by:
>> n_sip.c:4203 retrans_pkt: Hanging up call xxxxxxxxxx at 192.168.0.100 - no
>> reply to our critical packet
>> I am "assuming" that there is a problem with NAT. I have externip set
>> in sip.conf.
>> Any pointers to what I am missing?
>> Thanks,
>> JohnM
>>
>>
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>>
>