sip show peers

Erik, are you asking to see my peers...

I am not talking about SEEING... I am talking about they are actually not registering

if i ask for it, its a valid question. if they arent registering then sip show peers will show that. I want to see the result of your sip show peers.

whats ur email erik..

eeman (at) bluegrass (dot) net

check your mail

I was hoping to see some as UNAVAILABLE instead of UNKNOWN.. can you see if by changing those extensions to qualify=no it affects your ability to send/receive calls on those ones that cant seem to register.

Erik..

I am goin to email you Asterisk Debugs....

I backed up my config and restored it to a server in a completely different DataCenter, Different Public IP, NO Firewall...

I experience the same behavior only 16 peers register and NO more can after that limit.

The second install is even using Asterisk 1.6.0 not 1.6.2 as the first install...

I am lost.... I know asterisk very well, I havent a clue why this is happening

just to be sure, have you turned off iptables, atleast long enough to conduct testing? Maybe there's a problem with a rule there?

Its Off on both servers

[root@thirdlane ~]# iptables -L
Chain INPUT (policy ACCEPT)
target prot opt source destination

Chain FORWARD (policy ACCEPT)
target prot opt source destination

Chain OUTPUT (policy ACCEPT)
target prot opt source destination
[root@thirdlane ~]#

are these peers all coming from a single network or are they coming from multiple locations?

All the peers represent clients all over in different networks...

All i am doing is updating there tftp configs to point to this new server with new userid and passwords...

I have verified all the userid and passwords.

As a matter of fact if I power off/shutdown those that are registered... the others start to register themselves...

I dont know where is limit is...

asterisk 1.6.2 i built using http://www.voip-info.org/wiki/view/CentOS+5.2+and+Asterisk+1.6.x+install...

asterisk 1.6.0 I built using MTE ISO from this site

i backed up the configs from 1.6.2 and restored to 1.6.0

Still have the same behavior

Two different datacenters, Public IP, NO nat, No firewall

ok lets try this.... on the 1.6.2 machine back up your /etc/asterisk to like /etc/asterisk.bak ..

then in /etc/asterisk remove everything and go into your source code and do a make samples.

then copy sip.conf from /etc/asterisk.bak to /etc/asterisk

at this point you have a vanilla asterisk server with a bunch of peers in sip.conf. If you still have the problem then you'll know to look elsewhere. If the problem vanishes then its a matter of making a few changes at a time to see where the problem originates.

Erik...I feel so stupid...

I discovered the problem...

In my office networks we have 18 phones/peers all others are clients in outside networks.

I decided to apply a static ip to one of our phones and route it thru another Gateway and it immediately registered.

In my office I have 3 gateways all running pfSense as the firewall...

It is my conclusion that pfSense is limiting my SIP registration in some form originating from my Network...

I will imvestigate more with pfSense and will post my resolution once i figure out what the problem with it

Thanks for you help and speedy responses in these forums

it just so happens that pfsense has a package called siproxd. Install that and use it as your outbound proxy for your phones.

Sip proxy in pfSense did the trick...

Thanks Erik

Thanks All

good to hear.. btw if your interested we have pfsense based sip proxies we sell that are pre-configured to do separate voice vlans for qos. The cost really isn't much more than the cost of the hardware and assembly. We sell them for around $325 and the cost of the hardware, labeling, etching etc is around $250 which doesnt leave much left for flashing and assembly. If you are looking for a good solution for your customers this could do the trick for you.