SOLVED I hate networking! Help needed with RentPBX...

[mark_o]

Hello fellow nerds,

I have a beautiful newly configured PIAF Purple on RentPBX, and a network here in my office with about 8 different phones, all Cisco 7940's.

THE PROBLEM

Phone calls are connecting but audio is not going through. The worst problems are calls between phones in the office.

MORE INFO

We have AT&T DSL, with a Static IP address. Currently I am connecting to that with my trusty Linksys WRT-54GL router with Tomato Toastman firmware installed. Also currently, the DSL modem is set to Bridged mode and the router is directly connecting via PPPoE to the DSL network and receiving a Static IP address.

Now here's the Static IP information, sanitized for the web. In the following information, "z" refers to a top secret number.

Current Static IP: 108.224.z.102

According to AT&T, here is the rest of the Static IP information.

IP Address: 108.224.z.97
Subnet: 108.224.z.96
Subnet Mask: 255.255.255.248
Default Gateway: 108.224.z.102

I ordered 8 static IP addresses, but as many as 3 might already be taken by the networking setup (1 for gateway, 1 for the first address, 1 for the last address)

WHAT I WANT TO DO

I want calls to go through! Preferably I would assign a separate static IP to at least 5 different phones, so they would be most reliable and always connect to the RentPBX server correctly (hardcoded).

Also acceptable would be having all the phones, and the whole office network, on the same static IP, and just fixing this issue. SIP DEBUG information is posted below.

Also acceptable would be having all the phones on one static IP and the rest of the office on another static IP.

The Tomato router has VLAN capability and we have all the phones going through a Cisco Catalyst 3500XL series, which I know has VLAN support as well. So that could be a possibility too.

Also acceptable would be utilizing the VPN tunneling feature built into the Tomato router software. It has an OpenVPN client that could connect to an OpenVPN server on the RentPBX server.

SIP DEBUG INFO

Attached to this post in Zipped TXT format. This is the debug info for a call from Extension 104 to Extension 102, with the top secret number for the internal network changed to "z" and the number for the RentPBX server changed to "y".


Thanks in advance for all your help and thank you Ward for your wonderful software!

 

[wardmundy]

In FreePBX, try setting NAT=yes on the extensions.

 

[mark_o]

Tried it.

I agree that it's probably a NAT issue.

 

[dad311]

In FreePBX, try setting NAT=yes on the extensions.

Also make sure you "Asterisk Sip Settings" are correct. If these settings are not correct, you will have NAT issues.

Look in FreePBX under Tools>Asterisk Sip Settings.

 

[rossiv]

I want calls to go through! Preferably I would assign a separate static IP to at least 5 different phones, so they would be most reliable and always connect to the RentPBX server correctly (hardcoded).

Also acceptable would be having all the phones, and the whole office network, on the same static IP, and just fixing this issue. SIP DEBUG information is posted below.

Also acceptable would be having all the phones on one static IP and the rest of the office on another static IP.

The Tomato router has VLAN capability and we have all the phones going through a Cisco Catalyst 3500XL series, which I know has VLAN support as well. So that could be a possibility too.

Also acceptable would be utilizing the VPN tunneling feature built into the Tomato router software. It has an OpenVPN client that could connect to an OpenVPN server on the RentPBX server.

To my knowledge, Cisco 79XX phones in SIP have *never* played nice when registering to a server through NAT. The only way I could get it to work is if both the phone and server are on the same network, such as at the same physical location or bridged with a VPN - but never behind an NAT. I would *highly* recommend using the OpenVPN tools you said that you have. That is the only sure-fire way that I know of to get your phones working 100%.

 

[jmullinix]

Mark,

I hope this is a typo.

Current Static IP: 108.224.z.102

According to AT&T, here is the rest of the Static IP information.

IP Address: 108.224.z.97
Subnet: 108.224.z.96
Subnet Mask: 255.255.255.248
Default Gateway: 108.224.z.102

In this scenario, your Static IP address is set to your default gateway.

I suspect that your network from ATT is 108.224.z.96/29 or 108.224.z.96 255.255.255.248. Your network address (unusable) is 108.224.z.96. Your first usable address is 108.224.z.97 and your last usable address will be 108.224.z.101, giving you 5 usable addresses. Your default gateway (address on first ATT router) will be 108.224.z.102 and the network's broadcast address is 108.224.z.103 (not usable).

I am not familiar with Tomato's firmware, but the wan side should be set for 108.224.z.97 255.255.255.248 with default gateway of 108.224.z.102. Add the rest of your usable addresses as virtual addresses, if that is how Tomato handles them.

Since your phone server is in the cloud and your phones are behind NAT, you might try turning on SIP ALG on the router, if that is an option. I have read many other posts on these and other forums that Cisco 7900 series phones are not NAT friendly.

 

[rentpbx]

Based on your debug info, I see one thing that may be the problem with your configuration.

I see your SIP SDP has

v=0
o=Cisco-SIPUA 12121 0 IN IP4 192.168.1.252
s=SIP Call
t=0 0
m=audio 19844 RTP/AVP 0 8 18 101
c=IN IP4 192.168.1.252
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

Our server does not know where 192.168.1.252. I am guesting if you turn on rtp debugging in your asterrisk CLI (type rtp debug <enter> and watch as you are making call),

You should observe the ip on the following messages

Got RTP packet from
Sent RTP packet to

Please make sure that the IP are sent and received from the right IPs. In your case, you may see a lot of packet being sent to 192.168.1.252. While, your phone is behind 108.224.z.102.

This is a common issue with connection behind NAT.

As to the solutions, you may have a good suggestion by now already. I would like to summarize possible solution in more general case.

1. If your phone support STUN, ICE , etc. you may try them. You can use google to find a free public STUN server.

2. Your phone may support outbound proxy. In this case, you can use a sip proxy as the middle man that will translate your SIP/SDP information to your PBX. DD-WRT has a voip version that come with this feature. It comes with a milkfish sip proxy.

3. Dad311 has a good solution using VPN + DD-WRT. You may want to get some consulting help with him. His method has additional advantage in term of security and if you eventually would like to do automatic phone configuration using endpoint manager.