At the end of last week we managed to create an IAX to IAX trunk between 2 asterisk servers on the local wired LAN. IAX trunking is a way to connect multiple Asterisk servers and allow clients of different servers to place calls to each other. Continue reading
Tag Archives: sip
MeetMe testing
Last Wednesday we did some testing of our Asterisk MeetMe setup, at dekspc medialab in London. Setting up the Asterisk server on the local wired LAN and assembling an assortment of 6 SIP clients, using Ekiga on both Windows and Linux platforms, and a Mac running the SIP client Telephone. Each client registered with Asterisk as users era1 -> era9 and dialled ‘1234’ for the MeetMe conference room. Continue reading
Set up Asterisk conference calling with MeetMe.
I made my first (2-way) conference call on Asterisk/Meetme just now. To enable MeetMe with Asterisk, you first need to edit your meetme.conf in /etc/asterisk/ , mine looks like this:
Sipdroid works on G1 Android 1.5
Good news sipdroid works with our Asterisk server. Following up on a lead I received from Ben Charlton at the JISCRI developers workshop last week, I tried running sipdroid on a borrowed G1 Android phone (big thanks to Paul Hogan). The default SipDroid pretty much works, but you need to set the nat setting in the Asterisk sip.conf file to yes (see sipdroid website – issue 15). No big deal but SipDroid does not authetnticate without it.
Once I got over the above registration issue, it was just a case of setting the Asterisk audio codecs to include alaw. Seems pretty OK – supports a steady audio stream. We could use this with a separate webpage displayed on the phone web browser for showing images and the video stream. Currently the SipDroid does not display a remote video and I was unable to get it to stream video from the phone within SipDroid, but audio is certainly doable.