Packet sniffer found the problem. RTP was firewalled on the Asterisk box. Fixed it using the Asterisk firewall rules page on the wiki <http://www.voip-info.org/wiki-Asterisk+firewall+rules>.
The 30 second lag on the dialing has something to do with using the domain name instead of the IP address of the asterisk server in the SIP config on X-Lite. The call goes immediately when I set the domain to the IP address of the asterisk box. Thanks for your help. Rob Schall wrote: > This typically happens when the phone is natting or there is a firewall > between the phone and the asterisk server. The connection is made via > sip (5060), but the voice is over ports 10000-20000 (RTP). Most likely, > the sip connection is succeeding, since you are connecting, but the > actual voice is failing to transfer over RTP. > > if this is the case, I would aim to use IAX since it was made for this > type of use. > > If the phone is on the same network as the asterisk server, and you are > still having issues, use a packet sniffer and watch the traffic on both > ends. You should be able to receive every packet that is sent. Most > likely in this case though, you will only see those 5060 packets making it. > > Rob > > > Andrew Stewart wrote: >> I'm trying to setup my first Asterisk setup on a CentOS 5 installation >> on VMWare Workstation 6. Got two Linksys SPA941s working fine. But >> X-Lite softphones can't answer phone calls, and when one of them calls >> on of the Linksys phones they "connect" but neither party can hear hear >> the other. I noticed that the Linksys phones are connected via Native >> bridging while the X-Lite ones are connected via Packet2Packet bridging. >> >> Also, on the X-Lite phones there is a about a 30 second lag between when >> the X-Lite client hits dial/call and when the called party starts ringing. >> >> >> ::Asterisk setup:: >> Asterisk 1.4.4 >> Zaptel 1.4.3 (only ztdummy compiled) >> Asterisk Addons 1.4.1 >> CentOS 5 >> VMWare Workstation 6 >> >> >> ::sip.conf:: >> [Linksys01] >> type=friend >> secret=ledzep >> context=default >> host=dynamic >> mailbox=6445 >> >> [X-Lite01] >> type=friend >> secret=rammerjammer >> context=default >> host=dynamic >> dtmfmode=rfc2833 >> mailbox=2070 >> canreinvite=yes >> nat=no >> >> [Linksys02] >> type=friend >> secret=bigben >> context=default >> host=dynamic >> mailbox=6368 >> qualify=yes >> >> >> ::extenstions.conf:: >> [default] >> include => demo >> >> exten => 6445,1,Dial(SIP/Linksys01,20) >> exten => 6445,n,Voicemail(u6445) >> >> exten => 2070,1,Dial(SIP/X-Lite01,20) >> exten => 2070,n,Voicemail(u2070) >> exten => 2070,n,HangUp() >> >> exten => 6368,1,Answer >> exten => 6368,n,Ringing >> exten => 6368,n,Dial(SIP/Linksys02,20) >> exten => 6368,n,Voicemail(u6368) >> exten => 6368,n,HangUp() >> >> >> >> >> ------------------- >> Andrew Stewart >> >> >> >> _______________________________________________ >> --Bandwidth and Colocation provided by Easynews.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- ------------------- Andrew Stewart [EMAIL PROTECTED] (205) 585-2980 - cell _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users