I found the problem, although I still don't understand exactly what is happening. In the switch, a Zyzel ES-1528, there is a "feature" called "autoVOIP". The help says:
"Select IP Phone to give the highest priority to SIP, MGCP and SCCP packets passing through the switch." Seemed harmless, but they must be doing something squirrely. Disabling it eliminated the problem. Thanks again for all the help. Time to buy a better switch... Jeff On Feb 9, 2010, at 2:30 PM, Jeff Gilmore wrote: > 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/
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