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" <[EMAIL PROTECTED]>: > >> >> 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:[EMAIL PROTECTED];user=phone>;tag=2361964166 >> To: <sip:[EMAIL PROTECTED];user=phone> >> Call-ID: [EMAIL PROTECTED] >> CSeq: 1 REGISTER >> Contact: >> <sip:[EMAIL PROTECTED]: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:[EMAIL PROTECTED];user=phone>;tag=2361964166 >> To: <sip:[EMAIL PROTECTED];user=phone> >> Call-ID: [EMAIL PROTECTED] >> CSeq: 1 REGISTER >> User-Agent: Asterisk PBX >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER >> Expires: 120 >> Contact: <sip:[EMAIL PROTECTED]>;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:[EMAIL PROTECTED];user=phone>;tag=2361964166 >> To: <sip:[EMAIL PROTECTED];user=phone>;tag=as0771f231 >> Call-ID: [EMAIL PROTECTED] >> CSeq: 1 REGISTER >> User-Agent: Asterisk PBX >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER >> Expires: 120 >> Contact: >> <sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp>;expires=120 >> Date: Fri, 13 May 2005 01:59:09 GMT >> Content-Length: 0 >> >> >> --- >> Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms >> >> >> Does anyone know how to rectify this? By the looks of things, >> >> Many thanks, >> >> Spencer >> >> _______________________________________________ >> 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 >> > > > > ---------------------------------------------------------------- > This message was sent using IMP, the Internet Messaging Program. > > _______________________________________________ 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