Eric "ManxPower" Wieling wrote: > Steve Totaro wrote: >> Eric "ManxPower" Wieling wrote: >>> Steve Totaro wrote: >>>> Steve Totaro wrote: >>>>> I was using IAX2 to send traffic from a PSTN/SIP box to a PBX and it >>>>> worked fine except for audio issues that I believe are directly related >>>>> to IAX2 in version 1.2.x. I have four PRIs and want a separate context >>>>> for each going into the PBX. This worked very well with IAX. >>>>> >>>>> I want to use SIP to see if the audio issues are eliminated but Asterisk >>>>> does not seem to like multiple SIP account from one box to another (four >>>>> to be exact) >>>>> >>>>> I found this http://www.voip-forum.com/news.php?p=187 which makes me >>>>> think this is a known problem. Unfortunately, the link goes to an error >>>>> page. >>>>> >>>>> I have tried ever combination of credentials and setting in SIP conf but >>>>> the calls still fail. I tried friend, user, insecure=very, username, >>>>> from user, and anything else I could think of. >>>>> >>>>> Is there something I am missing or a workaround for this issue? >>>>> >>>>> PSTN Box (4PRIs)--Each PRI has it's Own Context in sip.conf, SIP--> PBX >>>>> (calls fail) >>>>> PSTN Box (4PRIs)--Each PRI has it's Own Context, IAX.conf--> PBX (calls >>>>> work) >>>>> >>>>> Thanks, >>>>> Steve Totaro >>>> I think I may have figured out my own issue. Since I am creating >>>> multiple SIP peers on two boxes that point to each other, I need to >>>> define separate ports for each one. Anyone know if that is the case? >>>> Makes sense to me but I cannot try it on the live server and my dev >>>> boxes are all doing other things. >>> no. It might be the case if you had multiple SIP clients behind the >>> same NAT router connection to a non-local Asterisk box. >>> >>> The userid and password that is sent with the call should make it hit >>> the correct sip.conf entry. Perhaps you are doing something silly in >>> your sip.conf configs. >>> >> Perhaps I am, let's hope so. This was my latest attempt to get it to >> work. The other server looks identical except the host IP. >> >> [general] >> ;bindport=5060 >> bindaddr=0.0.0.0 >> >> [default] >> >> [span1] >> type=friend >> host=192.168.6.2 >> username=span1 >> secret=xxxx >> context=to-span1 >> auth=rsa >> inkeys=span1-2-fast1 >> outkey=fast1-2-span1 >> qualify=yes >> disallow=all >> allow=ulaw >> allow=slin >> allow=alaw >> insecure=very > > I don't use RSA auth so I can't comment on that. My understanding of > insecure=very is vague, but you do NOT need it for Asterisk<->Asterisk > SIP connections and I suspect that is what is causing your problem. I > recommend against using qualify. >
Thanks, I will give those recommendations a try. If not, I am going to re-do their entire setup in a dev environment and then just move it over after testing. Thanks, Steve totaro _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
