[SR-Users] Fwd: Kamailio - Asterisk Integration

---------- Forwarded message ---------- From: Bugaian A. Vitalie <bugaian(a)gmail.com> Date: Thu, Oct 16, 2014 at 10:50 PM Subject: Kamailio - Asterisk Integration To: sr-users(a)lists.sip-router.org Hello list, thanks for creating kamailio and good documentation for it. Also thanks to Daniel for this guide http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb I managed to integrate kamailio with asterisk and a2billing and I do have some small issues with some UA's. Generally all soft phones are OK but from some I can not generate calls. Especially it is true for Linksys SPA and Grandstream gxv3140. From 10 phones I have one with this issue, all config is the same on device side and on server/config side, but some are failing to generate calls. These UA will receive calls but will not be able to call. They are registered OK and I see these devices as all other ones in location table. One work around I have found is to update manually the cc_sip_buddy table(from asterisk realtime) with IP address and port and big part of these US's will start working. But still I have some that will not work even after that. What I see when capturing with ngrep is this: 1) Invite with SDP 2) 407 Proxy Authentication Required 3) ACK 4) Invite with SDP 5) 100 trying -- your call is important to us 6) 401 Unauthorized 7) ACK Not all captured calls have all 7 steps some of these drop at 4th step and I mention that all UA-s are showing OK as registered. I use sip.Server == "kamailio (4.1.6 (x86_64/linux))" ; Please suggest some steps to increase that authentication time(i suspect that these UA's do not have sufficient time to authenticate) and help me to understand why this is happening. Thank you in andvance. Vitalie Alex Bugaian.