Hello, parameters in the Via header have nothing to do with authentication. It seems that the key log messages are in Asterisk:
[Jan 21 23:13:20] NOTICE[20785][C-00000001] acl.c: SIP Peer ACL: Rejecting '10.0.1.30' due to a failure to pass ACL '(BASELINE)' [Jan 21 23:13:20] NOTICE[20785][C-00000001] chan_sip.c: Failed to authenticate device <sip:95678@10.0.1.35 <mailto:sip%3A95678@10.0.1.35>>;tag=as4028dabf Is the 10.0.1.30 in the IP ACL white list for Asterisk? Cheers, Daniel On 22/01/16 16:15, DING MA wrote: > Hi, all > > We're trying to build a system that consists of pbx, kamailio and > asterisk in the following configuration. > > pbx (sip trunk) --- kamailio --- asterisk > > The kamailio and asterisk are integrated with same database. The > outgoing calls to pbx works. But there is a problem with incoming > calls from pbx. > If we make a consecutive calls from the same pbx user to the same user > registered with kamailio. The first would go through, but the second > call would be rejected by asterisk. We have insecure=invite set on the > trunk/peer, so asterisk is not supposed to auth the invite from > kamailio. But the pbx user (from in this case) is not in the database. > > The asterisk log says: > > [Jan 21 23:13:19] VERBOSE[20785] chan_sip.c: --- (16 headers 13 lines) --- > [Jan 21 23:13:19] VERBOSE[20785] chan_sip.c: Sending to 10.0.1.30:5061 > <http://10.0.1.30:5061> (no NAT) > [Jan 21 23:13:19] VERBOSE[20785][C-00000001] chan_sip.c: Sending to > 10.0.1.30:5061 <http://10.0.1.30:5061> (no NAT) > [Jan 21 23:13:19] VERBOSE[20785][C-00000001] chan_sip.c: Using INVITE > request as basis request - > 4aaa2dce75c60e8546994c3501dae9e7@10.0.1.35:5061 > <http://4aaa2dce75c60e8546994c3501dae9e7@10.0.1.35:5061> > [Jan 21 23:13:20] NOTICE[20785][C-00000001] acl.c: SIP Peer ACL: > Rejecting '10.0.1.30' due to a failure to pass ACL '(BASELINE)' > [Jan 21 23:13:20] NOTICE[20785][C-00000001] chan_sip.c: Failed to > authenticate device <sip:95678@10.0.1.35 > <mailto:sip%3A95678@10.0.1.35>>;tag=as4028dabf > [Jan 21 23:13:20] VERBOSE[20785][C-00000001] chan_sip.c: > <--- Reliably Transmitting (no NAT) to 10.0.1.30:5061 > <http://10.0.1.30:5061> ---> > SIP/2.0 403 Forbidden^M > Via: SIP/2.0/TLS > 10.0.1.30:5061;branch=z9hG4bK9c8e.5cd2c05f6a572312c7793abf5fe1183c.0;i=2;received=10.0.1.30^M > Via: SIP/2.0/TLS > 10.0.1.35:5061;received=10.0.1.35;branch=z9hG4bK249855c1;rport=59929^M > From: <sip:95678@10.0.1.35 > <mailto:sip%3A95678@10.0.1.35>>;tag=as4028dabf^M > To: <sip:16317@10.0.1.30 <mailto:sip%3A16317@10.0.1.30>>;tag=as35f47241^M > Call-ID: 4aaa2dce75c60e8546994c3501dae9e7@10.0.1.35:5061 > <http://4aaa2dce75c60e8546994c3501dae9e7@10.0.1.35:5061>^M > CSeq: 102 INVITE^M > Server: Asterisk PBX 13.6.0^M > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, > INFO, PUBLISH, MESSAGE^M > Supported: replaces, timer^M > Content-Length: 0^M > > Comparing the two invites from kamailio to asterisk, it seems the only > difference is that the second invite has an "i=2" in the Via header > while the first one has "i=1". Not sure what the "i=1" is for. Would > appreciate some insights on how kamailio is adding/handling the "i=#" > in Via header. > > Thanks. > > Ding Ma > SPG, Motorola Solutions > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com http://miconda.eu
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