Verision FreePBX: 188.8.131.52
Running on a VMware server
I have two remote SIP client Aastra phones, one 6730i and the other 6731i. If I have one or the other up, by itself, I can get the phone to register to my SIP server and make/receive calls. When I add the second SIP phone, it fails to register and gives me the status error 408. Basically, I’ll have one working SIP client and one non-working SIP client (simultaneous). At one point both were working simultaneously and as of recently, one just stopped and starting giving error 408.
I’ve tried using these commands in the Asterisk CLI module:
core set verbose 5
sip set debug ip x.x.x.x
And looking at the Asterisk Logfiless module, It only lists the one extension that is working. Never saw mention of the non-working extension.
Any help and/or advice would be appreciated.
P.S. If I need to post anything let me know. I’ve done everything through the GUI besides the initial installation.
Are you trying to register both phones to the same extension?
Are the phones and the server in the same network?
And are you effectively trying to register both phones from the same apparent IP address on the same port? (think firewalls/PNAT here)
@SkykingOH:Both phones have their own extension. The phones are not on the same network or subnet. Both sit behind Mikrotik Routers.
@dicko:Yes, both phones are going out the same public IP address to my SIP registrar back at the main office. One goes out the default port of 5060 and the other goes out 5061. In my pbx extensions, I have the ports changed there as well.
Brief extension configurations:
Outbound CID=DID# Outbound CID=DID
I also have trustrpid=yes and sendrpid=send-remote-party-id-header.
Brief configuration on phones:
NAT IP: public IP
NAT SIP Port:5061
NAT RTP port:51721 (don’t really know what this is, left it at default +1)
Authentication name=extension #
The other phone is the same except its NAT SIP port is 5061 and its NAT RTP port is the default 51720. Don’t know if I need to change the NAT RTP to be something within 10000-20000 or the proxy ports to be different either.
I should clarify; both phones sit on the same network, but do not sit on the same network as the PBX.
You don’t need the phones on two ports. Leave the port settings lone.
And make sure your NATting is correct.
I don’t know if I’m following then. If I have the phones communicating on the same ports, then 5060 will be sent out the External IP looking the same. Phone 1 will look like x.x.x.x:5060 and phone 2 will look like x.x.x.x:5060. To the PBX, it will accept it, but sending it back to the router, on the SIP clients side, the router will just hand it off to only 1 of the phones right?
So what do I need to change?
You are overthinking this. You can have two computers surfing same web site behind NAT right? SIP works same way, in fact all IP does!
I forgot to update my forum post, but I was able to fix my issue. I had a firewall rule that was looking for traffic with src-port=!5060 dst-port=5060 and dropping it. The second phone was trying to come in on port 1030 and requesting a redirect to 5060. I just created another rule to allow traffic from the Public IP address to passthrough my drop rule.
Thanks for the help.