30 second phone call ??
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Hi
I have set up acouple of systems with only a few hicups along the way. Toshiba phone system, with SIP Trunks behind pfSense.
I have enjoyed this program. :)On the last system, I have a problem. Outgoing calls work fine, can talk as long as you like. Incomming calls drop out after exactly 30 seconds, every time.
Calls operate perfectly when I swap back to the old netcom router.
I factory defaulted pfSense box and reconfigured in case I had "broken" something.2.0.1-RELEASE (i386)
built on Mon Dec 12 19:00:03 EST 2011
FreeBSD 8.1-RELEASE-p6Any clues would be greatly appreciated.
Regards
Mark -
I have seen this. I use 3CX but my phone calls cut out in and out every 32 secs. Be interesting to see what the out come is…
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Look at;
Advanced -> Firewall/NAT -> Firewall Optimization Options
Try "Conservative" instead of normal.
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Thanks for your reply
I tried that with no positive result.
????
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So after some more digging (wireshark) ….........
The SIP Trunk provider advises that he is receiving the customers internal IP address so has no place to send his Acknowledge to.
Wire shark shows:
Invite SIP
100 Trying
183 Session ProgressIn 183 Session Progress message header it says "Contact: sip:(customers phone number)@10.1.1.201:5060"
10.1.1.201 being the IP card in the phone system.
It needs to say the WAN Static IP, not the internal IP.Any ideas of a rule to make this happen ??
I have tried a few !!Regards
Mark -
Does running siproxd help? (See siproxd in packages list.)
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Unable to see any settings that are relevant ??
I don't have handsets with username password. They are SIP trunks that register to a Public IP ? -
@Oi:
Unable to see any settings that are relevant ??
Settings in what?
For more information on what siproxd does see its home page: http://siproxd.sourceforge.net/ The overview there says
Siproxd is a proxy/masquerading daemon for the SIP protocol. It handles registrations of SIP clients on a private IP network and performs rewriting of the SIP message bodies to make SIP connections work via an masquerading firewall (NAT). It allows SIP software clients (like kphone, linphone) or SIP hardware clients (Voice over IP phones which are SIP-compatible, such as those from Cisco, Grandstream or Snom) to work behind an IP masquerading firewall or NAT router.
SIP (Session Initiation Protocol, RFC3261) is the protocol of choice for most VoIP (Voice over IP) phones to initiate communication. By itself, SIP does not work via masquerading firewalls as the transfered data contains IP addresses and port numbers. There do exist other solutions to traverse NAT existing (like STUN, or SIP aware NAT routers), but such a solutions has its disadvantages or may not be applied to a given situation. Siproxd does not aim to be a replacement for these solutions, however in some situations siproxd may bring advantages.
which looks as if it might address your issue of private IP addresses in the message body.
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So I got a /30 subnet.
Used a device with an additional NIC for pfsense.
Called the extra interface DMZIPCard and connected the phone sytem to it.
Gave the IP card in the phone system a public IP address.All is good ;D
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With Siproxd you set it to look at a particular port. Ive only been able to get 5060 to work here. But then to the provider it looks like your natted device has a public IP.
But looks like you got it working. :)
