Alcatel Unleashed

All,
i have trouble in SIP Trunk ...any help...below sip process and attached the log file.

(1)lab> ps -edf|grep sip
mtcl 1854 751 0 13:07 ? 00:00:00 /DHS3bin/servers/sipmotor
mtcl 1855 1854 0 13:07 ? 00:00:11 /DHS3bin/servers/sipmotor
mtcl 2835 1854 0 14:30 ? 00:00:00 /DHS3bin/servers/sipmotor
mtcl 2836 1854 0 14:30 ? 00:00:00 /DHS3bin/servers/sipmotor
mtcl 5397 5283 0 18:11 ttyS0 00:00:00 grep sip

Instead of log and process - describe release in your PBX, management, for what you use SIP (SIP trunking, used signaling in SIP TG), another party (OXO, Asterisk...).

system info : R9.0-h1.301-48-mv-c80s1/ SIP Trunk is used instead of ISDN.what do you mean "another party (OXO, Asterisk...)."??

HI, any suggestions on attached traced log....
issues are: cannot to make calls & cannot recive.

hi all, any suggestions on one way calls...attached the log file

try creating SIP extension and then check internal communication !

regards.

Hi all, is their any way that SIP trunk can be configure in OXE without the NAT if IP ranges are different from the SIP provider range?

No,
NAT has nothing to do with SIP Trunk in OXE. You have to solve the NAT question at you router or firewall. Or you can use a SBC at our side see ALU doc "CPSA27_SBC_for_Enterprise_Voice&Applications_Solutions_ed3"

Hi all, I have problm on SIP trunk call transfer...if any call transfered by on ringing are ok, but if i transfered call after the annoucement the call going to disconnect after 4-5 seconds....anu suggestions on this??
i have attached the log files....
calling from 7782334 to 6800100 and transfered to 6339.