Asterisk/SipX bugs and modifications for UM

There are a few problems people have been running into with their UM setups.

Intermittent Timouts

The first is a problem where a timeout occurs intermittently when trying to call the Exchange UM server (approx 1 in 4 calls fails). This is caused by a bug in sipX 3.6 sending a malformed SIP header. The good news is that this has been fixed in sipX 3.8, however this version is still in beta (RC2). I have been waiting for a few months for the final release which is apparently 'just around the corner' to update the guide, but it seems to be causing people enough grief to justify posting about this issue now. I have been using 3.8 RC2 myself for some time, and have not run into any problems. The repo can be downloaded from As soon as 3.8 is released, I will update the instructions accordingly.

Play on Phone

The other issue people have been encountering is 'Play on Phone' not working from Outlook or OWA. A SIP trace reveals that Asterisk is sending a 407 Proxy Auth Required to the Exchange server, which it is unable to respond to. In order to get this working, we need to change the SIP connection type settings in each extension definition from friend or user to peer.

If you are using Trixbox (2.2 and above), then using FreePBX, go through each extension in the extension configuration menu, and change the 'type' option to peer as shown below.


If you are not using Trixbox, then you will need to manually modify your extension definitions in sip.conf and ensure the type is specified as 'peer'.