Facing this issue for sometimes now and hoping someone would help resolve it soon as I haven’t found an answer on the net yet.
Setup and Background:
I have FPBX 2.10/Asterisk 1.8.11 with two phones,
Poly IP 501 - 3 lines (Fully functional, registered, and monitored by asterisk)
Cisco 7940 - 2 lines (Fully functional, registered but NOT monitored by asterisk).
My phones are located in a remote location behind NAT/FW.
My server is located in another different location behind NAT/FW.
All proper NAT translations, port forwarding, and firewall holes are in place.
As long as I set qualify=no for Cisco 7940 lines, I am able to make calls normally without any issues and the status in ‘sip show peers’ goes to Unmonitored which I understand is the expected behavior of asterisk.
However, I would like to use qualify=yes and be able to fully monitor all lines and phones from my Asterisk Server.
Thanks in advance.
Qualify sends a SIP notify packet to the user/peer and times the reply back.
You say you have your network setup correctly but if qualify can’t correlate SIP replies then something is wrong in your NAT setup. I am sure you are having other problems such as one way audio, intermittent inbound etc.
Thanks for the prompt reply.
It works fine for Polycom and also when I setup X-Lite soft phone with the same extension used by Cisco.
My NATs/FW rules are as follows on both sites.
PBX Server Location
1-1 NAT on Cisco 2911 for Asterisk Server
- Opened from selected remote destinations for
UDP 1024-65535 (including ephemeral ports)
UDP 69 (TFTP)
1-many PAT on Motrola Router
- Allow All from AST Public IP Address
- Port fowarding enabled for all source/destination ports from AST Public IP Address (Not really needed as Polycom works fine without it)
I don’t have any issues with audio when I set qualify=no but I can’t send calls to Cisco if I set qualify=yes. However, if I set cisco’s extension on X-Lite with qualify=yes, everything works as expected.
I think this is specific to Cisco 79xx. In ‘sip show peer xxxx’ I do see that asterisk is sending SIP notify on x.x.x.x:5060 (phone’s NATed public IP) and even created a manual entry for that on both sides but not no avail.
I have tried setting all NAT related settings on Asterisk as well as on Cisco phone as per this articles, http://www.asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.html.