Exchange Unified Messaging - Unknown RTP Version 0 = HELP

Asterisk Asterisk Support
1

Hi All,

I have been trying to get asterisk 1.6.0.15 to work with Exchange Unified Messaging, seemingly a simple task since SIP TCP support has been added to Asterisk however for some reason I am not having much luck.

I actually have two issues outstanding but as they are seperate I have split them into seperate posts to avoind confusion.

This issue relates to a failure to negotiate a valid RTP stream between the two endpoints (Asterisk and Exchange). For the purposes of this test I have removed the other issue (the 302 redirect from 5060 to 5065 by Exchange) by setting the peer up with the port value of 5065.

My environment:

Exchange UM: 192.168.37.22
Asterisk: 192.168.37.100

sip.conf

[code][general]
tcpenable=yes
tcpbindaddr=0.0.0.0
bindaddr=0.0.0.0
bindport=5060
context=general
srvlookup=yes

[ext2001]
type=friend
host=dynamic
context=phones

[ext2002]
type=friend
host=dynamic
context=phones

[ExchangeUM]
type=peer
host=192.168.37.22
transport=tcp
port=5065[/code]

extensions.conf

[code]
[global]

[general]

[default]

[phones]
include => internal

[internal]

exten => 7000,1,Dial(SIP/ExchangeUM/7000)
exten => 7000,n,Busy()[/code]

The SIP conversation seems to be normal up until the point that Exchange provides SDP information for the RTP stream and subsequently there are several RTP messages with:

I have included the packet capture from Wireshark for reference:

Wireshark Packet Capture

Many thanks
Tom

2

no one else had these sorts of issues? :frowning:

3

My guess is that no regular on this forum would have tried this and anyone else with problems has either reported it through other channels, fixed it quietly, or given up.

At least from my point of view, for a problem that doesn’t affect me, I am not going to take the trouble of downloading, unzipping, putting onto a Linux system, and possibly getting back to you because it isn’t tcpdump compatible. If you post the plain text trace of the SDP, people might look at it.

4

Hi David,

Understand completely. Ultimately having no need for the non-SIP features I simply switched to sipXecs, problem solved.

Thanks anyway,
Tom