We recently encountered an interoperability issue between Asterisk and various SIP agents. It is related to the proper behavior of SIP ReINVITE. Asterisk uses ReINVITE to connect the media and DTMF paths of a caller and called party directly. The simple workaround we have found is to disable ReINVITE in Asterisk by setting "canreinvite=no".
Sample Asterisk configuration for SIP call routing
(Note: this information is provided "as-is")In sip.conf:
Please make note of the "canreinvite=no" setting below. A few SIP agents (SIP Gateways, etc. ) require this setting to interop with Asterisk.
[general]
bindport=5060 // UDP Port to bind to - SIP standard port is 5060
bindaddr=0.0.0.0 // IP address to bind to - 0.0.0.0 binds to all
srvlookup=yes // Enable DNS SRV lookups on outbound calls
disallow=all // First disallow all codecs
allow=ulaw // Allow codecs in order of preference
[VailTIM]
type=friend
context=default
callerid=Vail TIM <599>
host=192.168.142.8 // Change this to the IP address of the Vail SIP TIM machine
dtmfmode=rfc2833
canreinvite=no // Needed to interoperate with SIP agents that do not deal with ReINVITE well
mailbox=599@default
Note: If you run Asterisk on the same machine as the Vail SIP TIM, you should set the Vail SIP TIM's listen port to 5060 and change the Asteisk bind port to something other than 5060.
If you experience problems with DTMF collection (e.g., not all the digits are collected), or speech recognition is not working properly with an outbound call, we recommend that you try the following:
In the Vail SIP TIM configuration file, csta.conf.xml (in the Vail SIP TIM bin directory):
- Search for "rtpReceiver"
- Set "rtpReceiver" to false
- Restart Vail services