How do we determine or set the transport protocol used? We are currently on version 6.1.1.12, Asterisk version 1.8.30.
As background, we are attempting to connect some Cisco 8861 phones and from what we've been able to find, apparently they require that you enable TCP transport. The question we had from support personnel was "With your Asterisk Server, what is under Extension > extension number > Transport?" Obviously, that doesn't precisely apply on Thirdlane, so we're trying to find the equivalent setting.
Any other tips on getting these phones up and running would be welcome. We keep hitting walls on this.
Ok, thanks for the assistance
Ok, thanks for the assistance. I had come to the conclusion that I would have to do something like that. Fortunately, at about that time, I found a config file setting on the phone: 2 that enables UDP, thanks to this [site](http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+config…), in case someone else is looking for help on this.
In a nutshell, we had to:
1. Set up a TFTP server.
2. Make sure DNS option 150 pointed to it.
3. Create the XML config file referenced in the above link.
4. Download the Cisco Call Manager Express Locale pack and extract the correct regional pack, putting the contents on the TFTP server. This was mainly for the one JAR file the phone needed.
5. Tweak that setting above to point to UDP.
And after only about a week of messing with it, we now have the phone registering!
You might experiment with this:
To allow SIP TCP clients to connect with asterisk update sip.conf with the following
[general]
tcpenable=yes
tcpbindaddr=0.0.0.0
Within your SIP clients definition you have to add transport=tcp for each individual connection.
[client001]
callerid="Client 001" <001>
username=client001
secret=password
type=friend
host=dynamic
context=internal
canreinvite=yes
mailbox=001@default
transport=tcp
disallow=all
allow=alaw
nat=route
dtmfmode=inband