niddnet

Another update... I've packet-sniffed the call, and it all seems fine... there's a roughly equal number of RTP going to and from the Axon PBX to the world, and back again... but no audio. I do get a couple of funny messages though - 1 - Unknown RTP Type 0... and something about 'Powerschool' which seems to have some relation to UDP 5071. Anything else I can try? To recap - SIP - SIP works fine, on and off my network. 'Normal' SIP-Trunk calls work fine, as do Trunk-SIP. However, when using Trunk-SIP or SIP-Trunk, VMB and MoH do not seem to pass audio. Config: Trunk line 014xxxxx999 goes straight to ext 197 (MoH), 014xxxxx666 goes to ext 199 (VMB) 014xxxxx777 goes to ext 101 Dialing 197 or 199 from SIP devices direct work fine. Dialing from a trunk does not. Dialing the trunk-sip number (xxx777) rings on ext 101, and calls take place normally. Dialing from ext 101 to trunk, calls are fine too. HELP!!! (please?) R.

I should add as well, I have two incoming SIP lines - one of which rings on ext 199 directly (to go straight to VMB) and another that rings straight to 197 (for MoH) - and reading all the logs, there's no actual 'CALL ON HOLD' data - but then no calls are actually being placed onto hold... so I'm confident it's not the SIP provider intercepting the 'hold' signal. If anyone would like to look at the system, let me know, and I'll arrange some access to the system directly for you.

Hi all, pretty new to this, but reasonably familiar with networking / ports / forwarding / etc. Here's the problem, and here's what I've tried, found out, discovered etc. Running latest (demo) versions of Axon, Voicemail and Music-on-Hold (or whatever they're called!!) No hardware - just a NAT / uPnP router, a couple of SIP registrations with local numbers, and a few extensions (IP phones, and softphones). Calls INTO the PBX on my SIP line, and calls FROM the PBX over my SIP line are functioning mostly as they should - two-way audio, conversations, little lag - a perfectly usable system with the hunt groups, extension assignment etc, working fine. However, the Voicemail / Music-on-Hold systems do NOT send any audio out over the SIP line. (audio is received and recorded on voicemail) However, using IP phones (etc), everything works exactly as it should, including both VMB and MoH, and the really strange thing, this works either side of my router (indeed, either side of the world - I got a friend in the USA (I'm in UK) to set up an extension on my PBX and we talked, and played VMB and MoH by dialing their respective extensions). I know I've set this up once before as proof-of-concept, and everything worked fine. It seems that the RTP packets are seemingly disappearing before, or when they get to my SIP Provider or something. I haven't run ethereal or anything yet, but that's my next option. For clarification, I run a 24MBit ADSL line with (for testing) no firewalls on my PBX or the router, the PBX which is, just for the purposes of this exercise, in the DMZ. No, it's not secure, but it's all I could think of to try and get it to work. My SIP provider is SipGate (who I know can be flaky at times...) but this exact set-up has worked for me in the past. So where should I look? Axon? Voicemail? MoH? Or SipGate? I'd welcome some knowledge, so I can narrow it down somewhat before I pay for support or pay for the product!! (fwiw: TrixBox II worked fine doing everything, but I needed a Windows solution!!) Thanks to anyone who can help out!