On Mon, Mar 1, 2010 at 9:12 PM, Pizza Napoletana <[email protected]> wrote:
> I'm sorry if this is a repeat mail. But I think my previous mail with the > trace xml file was dropped because it was too big. Here is the trace file, > zipped. Thanks for helping. > > > > On Mar 1, 2010, at 5:53 PM, Tony Graziano wrote: > > > > On Mon, Mar 1, 2010 at 8:50 PM, Matt White <[email protected]>wrote: > >> The default is to send UDP invites. I would suspect your getting TCP >> invites because your packet is too big. >> >> Check and see if the invite is close to the MTU size. I bet it is. >> >> If so, you can try pruning the list of allowed codecs to reduce the >> invite. The other option is to see if patton has a compact sip header >> format. >> >> -M >> >> >> >>> On 3/1/2010 at 07:43 PM, in message < >> [email protected]>, Pizza Napoletana < >> [email protected]> wrote: >> >> In working with Patton support on debugging an issue with their M-ATA >> device (1 port FXS) where it ignores INVITEs from sipx, we have a theory. We >> think that the device is getting confused by sipx's TCP based iNVITEs which >> precede the UDP based INVITEs. >> >> How can I prevent sipx from using TCP as the initial transport for INVITEs >> before it tries UDP for this device? >> >> I thought this was based on the device's registration. But the device's >> registration doesn't say transport=tcp. >> I don't know if the DNS NAPTR record has any relevance here. But I have >> also set it to give priority to UDP over TCP. >> I don't know what else to do to force sipx to not try TCP transport so we >> can prove / disprove our theory. >> >> How can I force sipx to send out only UDP INVITEs? >> >> Thanks >> >> btw, the device registers fine and can make outgoing calls too. The >> problem is only with the ignoring of INVITEs. >> >> >> >> _______________________________________________ >> 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/ >> > > Look at Matt showing his stuff. I was gonna say that! A sample call trace > would indicate a whole lot here though. > _______________________________________________ > 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/ > > > > It came through first time. Have you tried calling it from something other than a SNOM handset? Did you configure the ATA to only use g711u/a (since it's on an ATA, using other codecs for HD and such is not going to produce better audio anyway)? Can you register a softphone (x-lite) and call it successfully?
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