i tried to do debug (put messages in /tmp/call.log) but there is no Invite having only one zero(like 06999786) all call were 0097937223 or 46364736. this means that only third else part is always active but if third part is active then there must be some invite starting with only one zero. it means second time invite call is not comming here .
My from main invites are only comming here route(3). i think all messages are going to asterisk because there is only one statement in ser.cfg having port 5970 (this is for asterisk) and all my register messages are also going there to asterisk.
there are too many register messages on my asterisk. i dont know why they are comming to asterisk as this port is not available for any user nobody uses this port only ser is routing calls to asterisk
route { if (uri==myself) { if (method=="INVITE") { route(3); break; } } } route[3] { exec_msg("cat >> /tmp/call.log"); if(uri=="^sip:00[1-9].*@.*") { route(4); break; } else if (uri=="^sip:0[1-9].*@.*") { strip(1); route(5); break; } else { route(4); break; }
} --- Iqbal iqbal@gigo.co.uk wrote:
dump a call trace showing your uri
Iqbal
Kamran Ahmad wrote:
hello
i m using ser-0.9.0 on 5060 and asterisk-1.0.9 on
5970
but problem is that i cannot send my ser back to asterisk. the problem is that it always goto route(4) second else if is not properly checked how to check this condition. i m adding 0 when asterisk is sending
call
back.
if(uri=="^sip:00[1-9]+@.*") { route(4); break; } else if (uri=="^sip:0[1-9]+@.*") { strip(1); route(5); break; } else { route(4); break; }__________________________________ Do you Yahoo!? Read only the mail you want - Yahoo! Mail
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