I could be wrong but this sounds like a NAT issue rather SIP related packet issue. You are not receiving a response back is what I get a lot of times when my NAT is not setup properly. Call goes on for 10 or 20 second (I try the echo application and it hangs up before I get to talk) and then cuts off.
-Bruce On Mon, Dec 19, 2011 at 7:41 PM, William Scott <[email protected]>wrote: > > It seems quite unlikely that the presence of > > an 'a=rtpmap' line in the SDP for G.729 is going to cause a call to have > any > > problems. > > Thanks for the reply. > > I'll expand on the scenario... > > This particular ATA does not send 'a=rtpmap' for any codec. > > When talking to a Asterisk PBX everything works fine. > > When talking to a VSP that sends an INVITE with "User-Agent: Sippy" > the call is setup then drops after 32 seconds. > > Packet captures shows that no ACK is received after the ATA sends the > 200 OK (missing rtpmap). After sending 200 OK about 6 times it then > sends BYE and the call disconnects. > > Every other ATA I have sends rtpmap and works fine. > > The idea was to manipulate Asterisk into not sending rtpmap for the > codec to confirm what happens. > > I'll now look for another solution. > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
