Re: [asterisk-users] insecure=invite - not working for different dial plan
Hi Joseph, Sorry, I still haven't been able to find the reference for you once again. One way to confirm this is by changing the order of codecs around and doing "sip show peer" on the peer. This reorders the codec preference around. Still looking for the actual reference though. Maybe it was in TFoT... :S --uzzi On Tue, Feb 16, 2010 at 9:32 PM, Joseph wrote: > Thanks for the input. > I know that order in extension.conf makes a difference but I did not know > that it applies to sip.conf as well. > I would like to find this article you have mentioned on WIKI what should I > look for :-/? > > -- > Joseph > > On 02/16/10 18:50, uzzi wrote: > >Order of configurations does make a difference so you may want to try with > >the same order as the one that works. Saw it on the voip-info.org wiki > >somewhere, but can't get you the link at the moment. > > > >In general, if you will be authenticating based on IP, you should leave > >username/secret out. > > > >Some more advanced users can correct me if I'm wrong. > > > > > > > > > >On Sun, Feb 14, 2010 at 7:35 PM, Joseph wrote: > > > >> I'm using "insecure=invite" with two different dial plans, it it > working > >> with one dial plan but not with the other. > >> What other parameters could influence "insecure=invite" > >> > >> In sip.conf below "insecure=invite" is working OK > >> [pstn-1270] > >> type=friend > >> secret=spa3k > >> username=voice-1270 > >> mailbox=369 > >> host=dynamic > >> insecure=invite > >> canreinvite=no > >> disallow=all > >> allow=ulaw > >> allow=alaw > >> nat=no > >> context=incoming > >> callgroup=1 > >> pickupgroup=1 > >> > >> > >> In sip.conf below "insecure=invite" is NOT WORKING > >> > >> [pstn-] > >> type=friend > >> secret=256 > >> insecure=invite > >> username=voice- > >> mailbox=622 > >> context=incoming > >> host=dynamic > >> canreinvite=no > >> disallow=all > >> allow=ulaw > >> allow=alaw > >> nat=no > >> callgroup=1 > >> pickupgroup=1 > >> > >> Both dial plan loaded on the same asterisk using the same Audiocodes > MP-114 > >> What other variable would influence operation of "insecure=invite" ? > >> > >> With the dial plan that "insecure=invite" is not working, asterisk logs > >> show: > >> "... username mismatch, have <4>, digest has > >> handle_request_invite: Failed to authenticate user "KMIEC J" > >> > >> -- > >> Joseph > > -- > _ > -- 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 > -- _ -- 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
Re: [asterisk-users] insecure=invite - not working for different dial plan
Order of configurations does make a difference so you may want to try with the same order as the one that works. Saw it on the voip-info.org wiki somewhere, but can't get you the link at the moment. In general, if you will be authenticating based on IP, you should leave username/secret out. Some more advanced users can correct me if I'm wrong. On Sun, Feb 14, 2010 at 7:35 PM, Joseph wrote: > I'm using "insecure=invite" with two different dial plans, it it working > with one dial plan but not with the other. > What other parameters could influence "insecure=invite" > > In sip.conf below "insecure=invite" is working OK > [pstn-1270] > type=friend > secret=spa3k > username=voice-1270 > mailbox=369 > host=dynamic > insecure=invite > canreinvite=no > disallow=all > allow=ulaw > allow=alaw > nat=no > context=incoming > callgroup=1 > pickupgroup=1 > > > In sip.conf below "insecure=invite" is NOT WORKING > > [pstn-] > type=friend > secret=256 > insecure=invite > username=voice- > mailbox=622 > context=incoming > host=dynamic > canreinvite=no > disallow=all > allow=ulaw > allow=alaw > nat=no > callgroup=1 > pickupgroup=1 > > Both dial plan loaded on the same asterisk using the same Audiocodes MP-114 > What other variable would influence operation of "insecure=invite" ? > > With the dial plan that "insecure=invite" is not working, asterisk logs > show: > "... username mismatch, have <4>, digest has > handle_request_invite: Failed to authenticate user "KMIEC J" > > -- > Joseph > > -- > _ > -- 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 > -- _ -- 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
Re: [asterisk-users] SIP Registration Failure Logging
Try: core set verbose 4 >From the Asterisk CLI -uzzi PS: If you're not seeing any connection information, be sure to double-check the IP address is correct. Learned that lesson the hard way =\ On Sun, Jan 31, 2010 at 5:51 PM, Jim Rosenberg wrote: > Let's say I have two Asterisk boxes, A and B. I am trying to get A to do > SIP registration on B, so an extension for A can dial SIP phones covered by > B. If I examine the logs on B, if the registration succeeds, I am seeing a > notice to that effect on B. But if the registration *fails*, i'm not seeing > any message logged on B. Maybe I'm just not looking in the right place. Is > there a way to turn on logging or debugging so registration failures are > logged on the "target"? > > I'm doing something profoundly stupid, and seeing the notorious > > chan_sip.c:12009 handle_response_invite: Failed to authenticate on INVITE > > message, and trying to trace why. > > -Thanks, Jim > > -- > _ > -- 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 > -- _ -- 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
Re: [asterisk-users] disable comfort noise
On Fri, Jan 29, 2010 at 1:14 PM, wrote: > To get back to the original poster's possible situation, i've seen this > with my first IP phone, which was a cisco 7912 (SIP image). With that > phone, asterisk sometimes gave me this same error. I'm quite sure i've > asked the very same question here back then (probably i was a bit more > specific :). Since it is related to only this type of phone, i've gone > to different ip phone products. > > regards > adam > > -- > _ > -- 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 > Please correct me if I'm wrong As the error says, "Please turn off on client if possible." Comfort noise (aka silent suppression, or Voice Activity Detection (VAD)) is not supported by Asterisk. It needs to be turned off on the user (client) end. This may be a phone or another switch/PBX. See http://www.voip-info.org/wiki/view/RTP+Silence+Suppression for more details -- _ -- 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
