search for: sip_incoming

...er gets dial tone - User dials 1002 - IVR says "No such extension - please try again" ??? It seems that the 1st digit gets canceled out? Debugging the server output I get (tried twice): snip -------------------------- Goto (incoming,s,70) -- Executing Goto("Zap/1-1", "sip_incoming|s|1") in new stack -- Goto (sip_incoming,s,1) -- Executing Dial("Zap/1-1", "SIP/1001||rtT") in new stack -- Called 1001 ioctl(ZT_LOADZONE) failed: Inappropriate ioctl for device Failed to register zone 'United States / North America': No data available...

...support TCP. http://bugs.digium.com/view.php?id=4903 Step1: configure LCS 2005 to let sip uri: *@pstngw.domain to route to next hop: pstngw ip address Step2: patch your asterisk chan_sip.c to support TCP Step3: configure your Asterisk sip.conf, extensions.conf simple example :-) sip.conf context=sip_incoming extensions.conf [sip_incoming] exten => _XX.,1,Answer exten => _XX.,2,Noop(do trust ip check or some authentication) exten => _XX.,3,Dial(Zap/${EXTEN}&SIP/${EXTEN}) I still find out how to let LCS 2005 accept SIP invite from Asterisk, Need more help. 2005/8/13, bubuk <bubuk@ish....

...d Cone NAT' as per STUN RFC's terminology. All my UAs are XLite-on-Windows. My Asterisk is running on Fedora Core 6. I have the following flags set in the [general] section of my sip.conf [general] nat=yes qualify=yes rtpkeepalive=60 rtptimeout=90 rtpholdtimeout=300 canreinvite=no context=sip_incoming (... among others ...) Following is the relevant portion of my extensions.conf [sip_incoming] exten => _.,1,GotoIf($[${SIPDOMAIN}=mydomain.com]?4) exten => _.,2,Dial(SIP/${EXTEN}@${SIPDOMAIN}) exten => _.,3,HangUp() exten => _.,4,Dial(SIP/${EXTEN}) exten => _.,5,HangUp() exten =>...