PBX Servers?

[StanAccy]

Stan, you might have better luck posting that question on a FreePBX forum. But generally speaking, your pbx does not normally monitor the voipo mailbox. Rather, your extension phones monitor their own individual mailboxes on the pbx. You could setup one of the Cisco phones with a direct connection to the voipo line on one of the additional Cisco accounts, then it might monitor the voipo trunk voicemail box for you. Good luck.

Thanks for the response. I did finally get this working.
Cisco 7960/7961/7940 have a working NAT implementation in their SIP firmware so they can talk directly to Voipo over my NAT. However, the 7970 has a totally screwed up NAT implementation so I had to put Asterisk in between to handle the NAT correctly. I also configured the Cisco 7970 extension (Im running FreePBX/Asterisk in Device mode, not User mode) to check a virtual mailbox setup on Asterisk, and then I told Asterisk to allow the unsolicited SIP MWI packets that Voipo sends out to all its registered SIP connections (the Grandstream device and the Asterisk server). Voipo's server does not support MWI Subscription (there's an RFC for this) so you have to tell Asterisk to allow the unsolicited MWI SIP NOTIFY packets. You also have to tell Asterisk to relax its SIP packet checking constraints since the SIP NOTIFY packet coming from Voipo is not valid and Asterisk simply drops it by default.

I used Asterisk 11.x (latest), running FreePBX (latest at time of writing) and a Cisco 7970 with 8.5.4 firmware (and I tried a *lot* of different firmwares). My firewall in between Asterisk and the internet is pfSense 2.x. Nothing special configured on the firewall (no SIP ALG, no siproxyd etc), just open UDP 5060 to forward to Asterisk server, and UDP RTP ports (I opened a range of 20) to also forward to Asterisk.

Set your trunk register string in Asterisk, the extension mailbox needs to be altered, and then tell Asterisk to allow unsolicited NOTIFY SIP packets and relax its SIP packet checking, and finally tell Asterisk how to link the unsolicited MWI NOTIFY packets to the virtual mailbox that the extension is listening on.

One other aspect - I configured my 7970 with 2 lines (I have two separate Voipo accounts). In order to get the unsolicited MWI notifications to work for two separate lines, you have to register the line 2 trunk within Asterisk to talk to a different host/port combination at Voipo. You can vary either the host and/or port. Fortunately Voipo accepts SIP registrations on UDP port 5061. This is vital since Asterisk uses the remote host/port combination to key off the unsolicited MWI NOTIFY packets (not the internal contents of the SIP packet - there's already a feature request logged to improve this in Asterisk). Once you do that, you can get the MWI light on the 7970 to light up when you have mail in either mailbox (given a suitable phone SEP<MAC Addr>.xml config). I also had to open up UDP port 5061 on my firewall to account for the second line.

PM me for config details if anyone wants to know more - I spent a heck of a lot of time on getting this working but now it works great and the 7970 is a great upgrade from the 7961 and 7960 we were using before.

[GreenLantern]

Aha, that makes a lot more sense now. I didn't fully understand what you were running into before. Great workaround, but wow, a lot of work.

So all of this was unique to the 7970? I haven't noticed any trouble with MWI for multiple accounts with other phones. But I rarely use Cisco. Cisco (and Polycom) seem overly complicated to me.

[StanAccy]

Aha, that makes a lot more sense now. I didn't fully understand what you were running into before. Great workaround, but wow, a lot of work.

So all of this was unique to the 7970? I haven't noticed any trouble with MWI for multiple accounts with other phones. But I rarely use Cisco. Cisco (and Polycom) seem overly complicated to me.

Yes, just the 7970. My 7960s and 7961s have worked fine for well over a year. Cisco made some intentional changes to the SIP firmware on the 7970 that broke all sorts of NAT related stuff. While the change was technically within the RFC spec, it broke most direct connections traversing NAT. They claim it was a change made for security, but I just don't see it how it makes anything more secure. But it does make things more difficult for you to use a 7970 outside of a Cisco Call Manager environment... go figure.

Now that I know how, setting up another one is simple. It didn't help that Voipos NOTIFY packets are missing a required SIP header, and it didnt help that Voipo doesnt support MWI subscription, and it didn't help that Cisco intentionally screwed up NAT functionality. I spent a lot of time thinking the phone specific config file was wrong (it didn't help that Cisco reports nothing for an invalid config), but as it turns out, once you get Asterisk running with NAT, and the trunk mapping correct (on different UDP ports for different lines), pretty much any of the available configs for the 7970 out on the internet work fine.