Holger Schurig wrote:
thanks for the heads-up about grandstream, but as i stated in the original message, i'm using xten lite softphones. hopefully this is the approproriate forum for this question; i believe this is not an xten configuration issue because i can connect to a ser/rtproxy/nathelper server without problems and i can connect directly to a voicepulse account, which leads me to believe that this is an * configuration problem on my part. less likely, i suppose, is the chance that * isn't as robust in handling nat than ser or whatever voicepulse is running.i'm new to asterisk and am having trouble placing outbound calls. i
Bug Grandstream so that they finally fix their buggy software.
The GS phone sends occassional SIP packets to port 0, not to port 5060, as tcpdump or (better) ethereal will show you.
There's a page on this at voip-info.org.
given the configuration files that i posted in the original message, are there any changes that i should make? certainly the asterisk faq makes the solution seems straighforward [1]:
"Most likely you have a SIP client behind NAT that is trying to communicate with Asterisk without having the "nat=yes" setting in place in sip.conf. Another cause for this could be related to a user device that has an sip entry but has been physically removed (switched off or LAN-disconnected)."
but as my original message showed, i do have nat=yes in my sip.conf and i don't believe the latter scenario is true.
any help is greatly appreciated.
[1] http://www.voip-info.org/wiki-Asterisk+FAQ _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
