VOiP Problems

We have a VOiP server (Trixbox CE, freepbx server v2.6.2.3) with approximately 75 extensions.
The server is in a datacenter, connected directly to an external IP.
We have 4 offices, each office uses a Sonicwall Firewall (NSA 2400, TZ100, TZ210 and a TZ200)
We are encountering multiple problems

1.) No voicemail.  
The voicemail on the phone works fine most of the time, maybe 60%.
But other times we click on the voicemail button on our phones (Cisco SPA303)
There is no sound what so ever. Other times we can hear the computer voice asking for password, after the password is entered it says "you have x new messages" "to listen to your messages press one"
After pressing 1, no message is played and the computer voice says "to listen to your message again press 2" etc.
But if I log into the trixbox gui ,  I am able to listen to all messages there.

2.) IVR no sound
We have a IVR setup to route calls to specific extensions.
Sometimes when calling our main number there is no sound either.
I think the times are the same as the voicemail issue.

Please help,

Thank you
Armin
ascherkeyAsked:
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PerarduaadastraCommented:
A fruitful source of VoIP problems is firewall configuration, and in particular double NAT-ing between endpoints.
Because VoIP uses UDP there is no error checking and packets that are lost or dropped disappear without trace.

Have you looked at the SIP traces from the phones or run a PCAP to see what's happening?
Wireshark is excellent for this kind of work.
ascherkeyAuthor Commented:
I am not familiar with wireshark, how would I capture packages from a phone to the voip server?
I have been looking at Firewall forums, I found 2 articles, but when I changed the sonicwall setting accordingly, even more problems arose.
PerarduaadastraCommented:
Most phones have a utility for logging SIP traces built-in. My experience is mainly with Snom phones, and I have to say that I've never tinkered with a Cisco phone. Find the utility and try a short SIP trace first - just a few seconds, as quite a lot of data is generated.
You're looking for sending and receiving IP addresses that you're expecting, and the SIP status codes, eg: SIP/2.0 100 Trying.

A lot of what you see are unique identifiers, which aren't relevant to what you're trying to discover. However, the error messages should give at least a pointer as to the nature of the problem.

Regarding Wireshark, there is documentation here:

http://www.wireshark.org/docs/wsug_html_chunked/ChapterIntroduction.html

Section 3 deals with the interface and how to use it, but there is a lot of interesting stuff before you get there, though you are unlikely to need to know how to build it under Unix and Windows - I skip all that, and just download the latest stable release for the Windows platform. You may be a Linux, Mac, or Unix man, in which case you would use those versions.

Another question: What firewall installations does the datacentre have? Or are you expected to provide your own? Either way, the fact that all your branch offices are experiencing these problems but direct access to the Trixbox works fine, suggests that the problem is likely to be at the server end of the connection rather than the phones.

Has this problem been present ever since the installation went live, or has it happened recently? If the latter, what changed, and who changed it? Has your ISP made some changes and not told you?

Have you spoken to Sonicwall? As you have four different appliances of theirs (even though three of them are getting on a bit), they ought to be able to shed some light on how best to configure them for VoIP. At least they should be able to confirm that your present configurations are OK, or that they are contributing to the problem.
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ascherkeyAuthor Commented:
I tried all the above, nothing seems to fix the issue.
My last option is to switch the server. which i will do tomorrow

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PerarduaadastraCommented:
Are you able to post any SIP traces?

The fact that sometimes the call is established but nothing can be heard suggests that the RTP stream has not been configured correctly. RTP is configured from information placed in the SIP packet body by SDP (Session Description Protocol), which defines such things as port numbers, codecs, protocols etc. If, for example, there is a request for a particular codec and one of the devices doesn't support it, then if an alternative can't be negotiated the media part of the call (that is, the speech) will fail and you won't hear anything.

Have you checked that everything is using the same codecs and protocols? Are some of the phones using SIP while others are trying to use secure SIP?

Without some more detailed information I can't really be of any help.
ascherkeyAuthor Commented:
Was a hardware issue, installed new server, problem is resolved
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