SIP Trunk - SIP<->SIP phone call rings but disconnects

Hoi *Guru’s,
Our prof has us doing an Asterisk lab for our Carrier Services class. Cudos on the O’Reilly book, so far it’s been great for teaching us how to get this badboy up and running :wink:

One problem that we all seem to be running into is setting up a SIP trunk between 2 *boxes. We’ve been following the example in the book “Connecting Two Asterisk Boxes Together via SIP” and have had some partial success. If we make a call from an analog phone connected to the digium card to another SIP or analog phone connected to the other box, phone rings and call goes through just fine. However, if we make a call from a SIP phone to another SIP phone, cisco or X-Lite, connected to the other *box, the phones ring both ways but as soon as the callee picks up - call disconnects.

Seems to be the same problem throughout the class and I haven’t been able to find a similar issue on the web. Just wondering if anybody has run into this before, or can offer some advice…

Thanks for your attention,


Can you give us some output from the cli>.

At a guess; do you have canreinvite=yes and are there compatible codecs
in the sip.conf?

Your the MAN!!!

It was the canreinvite=yes!!!

Just added it to our SIP.conf for each other and things work great.

Thanks bwilks!