Thanks all for inputs. In doing more testing, I find that the delayed- pickup behavior happens when plugged into one particular ethernet switch, but not others. Not sure yet what to make of that, but I am digging into it.
Tony, I did find the symmetrical RTP setting in the SIPx phone config screens, but could not find a corresponding setting on the ATA's web interface itself--odd. Anyway, I set it, but it had no effect on this problem. Jeff On Feb 9, 2010, at 11:45 AM, Dale Worley wrote: > On Tue, 2010-02-09 at 08:53 -0500, Jeff Gilmore wrote: >> I have attached a trace of a call that went like this: >> >> 1. Placed call from an outside phone (i.e not attached to my switch). >> 2. The sipx extension which has the called DID as an alias rings >> immediately. >> 3. I pick up after 1 ring. >> 4. I hear dead air on both phones (and no ringing, or ring tone on >> either). >> 5. After about 10 seconds, I hear a brief ring tone on the calling >> phone. >> 6. The audio connects, >> 7. I stay on the line for about 10 seconds. >> 8. I hang up the sipx extension > > Note that you have not provided any of the extension numbers, nor > described the expected call behavior of the DID, which makes things > harder for us. > > Looking at the trace, it appears to be a normal call. The number > called > is 6078829030. It gets mapped into sip:[email protected]. > sip:1...@... gets mapped into sip:[email protected]:5060, > sip:1...@ev..., and sip:~~vm~...@ev... (with priority 0.1). The > call is > sent to sip:[email protected]:5060. sip:1...@... has no contacts. > > The interesting bit is the INVITE to sip:[email protected]:5060. > It is > sent by TCP at 00:20:53Z, and appears to get an immediate ICMP reject. > So it is resent by UDP within the same second. The phone responds 100 > and 180 within the same second. The 180 is delivered to the > originating > UA promptly. But the UA (Linksys/SPA2102-5.2.10) responds 200 at > 00:21:07Z, which is 14 seconds after ringing starts. > > Based on your description, it appears that the UA took about 10 > seconds > after you answered the phone to generate the 200. > > So it looks like the problem is with the UAs on each end of the call. > > Dale > > _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users sipXecs IP PBX -- http://www.sipfoundry.org/
