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. _______________________________________________ --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
