On Thu, 2013-01-03 at 12:47 -0500, Nick Khamis wrote: > Hello Everyone, > > Before getting into SIP and RTP traces, I wanted to clarify some of > the sip.conf settings that may to some seem redundant or have a > misconception with. I do apologize if this has been discussed time and > time again as I would imagine. If anything, this email would make > google search results that much stronger :). > > With the UA local to my network I had tested two way audio, and now > with the phone outside of network, we have no way audio. Before > discussing NAT (which is enabled on the peer), and port forwarding > (which is setup on the remote location), I would like to make sure I > fully understand all the sip.conf settings. We are using Asterisk > realtime via sip_buddies, and the fields in question are: > > (Enclosed in brackets are an example value for the setting) > > * host (dynamic): No problem here. Just wanted to mention that it's > set as such.... > * nat (yes): No problem here either.... > * defaultuser (1...@example.com): Does the "@example.com" have to > point to the UA (i.e., (1003@ua-public-ip), or is it just a name type > field? > * fullcontact: What to put here for a UA that is running at a remote > location with dynamic external IP? > * ipaddr (ua-public-ip): I did try setting it to the public ip of the > UA, but is that really practical? > What if I don't know where the initial registration request is coming > from? I am guessing "host=dynamic" takes care of that. > * defaultip?? > * dynamic: Should this be set to yes, or is "host=dynamic" sufficient? > > The phone registers fine, and terminates a call through our providers. > Just no audio both ways, which would suggest something more that an > RTP issue which should at least have one way outgoing audio. > > Things that have been attempted: > * Port forwarding to the phone > * Changing defaultuser to 1003@ua-public-ip. This made our OpenSIPS > sip proxy through a fit. > > Things I will attempt today: > Calling the UA extension from an extension here > SIP trace > > Your help is greatly appreciated!!! > > Nick. >
Hi Is your directmedia/canreinvite (depending on asterisk version) set to no? Regards Ish -- Ishfaq Malik <i...@pack-net.co.uk> Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users