Hosted VOIP and pfSense
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I'm seeing the same thing with our hosted PIAF setup. We have 4 SPA-942 phones and 1 Aastra 57i CT and they randomly unregister over the course of the day. Siproxd didn't do anything, manual NAT works until the phones try and re-register and then they fail.
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Did you put
nat=yes
externip=xxx.xxx.xxx.xxx
externhost = mypbx.mydomain.com
localnet=192.168.1.0/255.255.255.0
externrefresh=10in SIP_NAT.conf
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localnet=192.168.1.0/255.255.255.0 make it to match your network
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@mst:
localnet=192.168.1.0/255.255.255.0 make it to match your network
Our PIAf hosted box has a public IP, do I still need this? Our setup worked perfectly fine with an SG565 in place and Sip Proxy turned on.
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if you have public IP then no
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@mst:
if you have public IP then no
Thanks for the clarification. I didn't think it was needed. Our phones just unregistered again. I'm pulling this box until this issue is fixed somehow. I'm beyond frustrated and we NEED our IP Phones to work reliably.
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check this post: http://www.trixbox.org/forums/vendor-moderated-forums/aastra-endpoints/57i-not-registering-no-service can be usefull
MST
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My phone ARE getting separate ports when they boot up initially, they only lose the registration when they try and re-register. I put the SG565 back into service at that office and the phones have been rock solid for the past few hours.
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I use Askozia PBX in VmWare setup… Works like a charm.....
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I use Askozia PBX in VmWare setup… Works like a charm.....
I don't think this has to do with the PBX so much as the natting of SIP ports. We are going to demo OnSIP in the coming weeks and I saw one of the threads pertaining to SIP nat.
I'm just frustrated because this should just work and it's "sort of" working which is worse than not working at all.
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I tweeted about my problems and Chris sent me this link:
http://doc.pfsense.org/index.php/VoIP_Configuration
I think #2 should help me out, but I can't test until next week.
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I know, but I just forward the used ports through PFSense to the PBX, handling the SIP traffic.
Good audio and no problems at all.
I use Askozia PBX in VmWare setup… Works like a charm.....
I don't think this has to do with the PBX so much as the natting of SIP ports. We are going to demo OnSIP in the coming weeks and I saw one of the threads pertaining to SIP nat.
I'm just frustrated because this should just work and it's "sort of" working which is worse than not working at all.
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I changed the System->Advanced-> Firewall Optimization options to conservative and the phones have stayed registered for an hour which is longer than normal.
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Just change the keep connection alive settings in the SIP phones…..
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I could do that, but with 20 phones in 3 states this was much easier to do.