[Asterisk-Users] realtime sip show peers no nat

[G.Marshall]

Hello Matthew,

Thank you, yes, nat is on, unfortunately, the contact points to the
private IP address behind 212.74.112.53, but at least now I have somehting
else to work on.

I have cc'd the mailing list because I think it would be useful for others.

Many thanks for your help,

Spencer

> To correctly verify if NAT is on a peer or not:
>
> realtime load sippeers name 5561 (look for the NAT column, should be "yes"
> or
> "no")
>
> if you need to change:
>
> realtime update sippeers name 5561 nat yes  (or nat no)
>
> then do:
>
> sip prune realtime 5561
>
> then:
>
> "sip show peer 5561 load"
>
> It should correctly display your nat'd option now.
>
> -Matthew
>
> Quoting "G.Marshall" <g.marshall at dalmany.co.uk>:
>
>>
>> Hello
>>
>> sip show peers does not mark hosts as NAT even though sip.conf and
>> sip_peers table has nat=yes.
>>
>> spitfire*CLI> sip show peers
>> Name/username              Host            Dyn Nat ACL Mask
>> Port     Status
>> voipuser.org/gdsm          216.127.66.119       N      255.255.255.255
>> 5060     Unmonitored
>> 5560/5560                  192.168.4.5      D   N   A  255.255.255.255
>> 5060     Unmonitored
>> 5561/5561                  192.168.4.5      D   N   A  255.255.255.255
>> 5061     Unmonitored
>> 4561/4561                  212.74.112.53    D   N      255.255.255.255
>> 8413     Unmonitored
>> 4 sip peers [4 online , 0 offline]
>> spitfire*CLI>
>>
>> asterisk listens on 192.168.4.3 and 82.70.154.145.  The host
>> 212.74.112.53
>> is the external (NAT) address for a sip phone whose LAN address is
>> 10.44.16.163.
>>
>> sip debug shows the following
>> spitfire*CLI>
>> <-- SIP read from 212.74.112.53:8413:
>> REGISTER sip:82.70.154.145 SIP/2.0
>> Via: SIP/2.0/UDP 10.44.16.163:5060
>> From: <sip:4561 at 82.70.154.145;user=phone>;tag=2361964166
>> To: <sip:4561 at 82.70.154.145;user=phone>
>> Call-ID: 1812954233 at 10.44.16.163
>> CSeq: 1 REGISTER
>> Contact:
>> <sip:4561 at 10.44.16.163:5060;user=phone;transport=udp>;expires=120
>> User-Agent: Cisco ATA 186  v3.1.0 atasip (040211A)
>> Content-Length: 0
>>
>>
>> --- (9 headers 0 lines)---
>> Using latest request as basis request
>> Sending to 10.44.16.163 : 5060 (NAT)
>> Transmitting (NAT) to 212.74.112.53:8413:
>> SIP/2.0 100 Trying
>> Via: SIP/2.0/UDP 10.44.16.163:5060;received=212.74.112.53;rport=8413
>> From: <sip:4561 at 82.70.154.145;user=phone>;tag=2361964166
>> To: <sip:4561 at 82.70.154.145;user=phone>
>> Call-ID: 1812954233 at 10.44.16.163
>> CSeq: 1 REGISTER
>> User-Agent: Asterisk PBX
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
>> Expires: 120
>> Contact: <sip:4561 at 82.70.154.145>;expires=120
>> Content-Length: 0
>>
>>
>> ---
>> Transmitting (NAT) to 212.74.112.53:8413:
>> SIP/2.0 200 OK
>> Via: SIP/2.0/UDP 10.44.16.163:5060;received=212.74.112.53;rport=8413
>> From: <sip:4561 at 82.70.154.145;user=phone>;tag=2361964166
>> To: <sip:4561 at 82.70.154.145;user=phone>;tag=as0771f231
>> Call-ID: 1812954233 at 10.44.16.163
>> CSeq: 1 REGISTER
>> User-Agent: Asterisk PBX
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
>> Expires: 120
>> Contact:
>> <sip:4561 at 10.44.16.163:5060;user=phone;transport=udp>;expires=120
>> Date: Fri, 13 May 2005 01:59:09 GMT
>> Content-Length: 0
>>
>>
>> ---
>> Scheduling destruction of call '1812954233 at 10.44.16.163' in 15000 ms
>>
>>
>> Does anyone know how to rectify this?  By the looks of things,
>>
>> Many thanks,
>>
>> Spencer
>>
>> _______________________________________________
>> Asterisk-Users mailing list
>> Asterisk-Users at 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
>>
>
>
>
> ----------------------------------------------------------------
> This message was sent using IMP, the Internet Messaging Program.
>
>