<div dir="ltr">Hello everyone,<div>I have a server with OpenSIPS 1.10 installed and I want my customer can be able to make call to landline, so I relay all the landline request to my ITSP (which is net2phone). Everything is working perfectly except that when landline phone ends call, instead of sending BYE like</div>
<blockquote style="margin:0px 0px 0px 40px;border:none;padding:0px"><div>BYE sip:0912167@mydomain:56566;transport=tcp SIP/2.0</div></blockquote>they sends<div><blockquote style="margin:0px 0px 0px 40px;border:none;padding:0px">
<div>BYE sip:0912167@mydomain:56566 SIP/2.0<br></div></blockquote>which is invalid according to dialog information, so I using fix_route_dialog() to fix Request-URI. But the problem is after fixing it, OpenSIPS doesn't send BYE to my customer and crash after several tries.</div>
<div><br></div><div>To verify if fix_route_dialog() is the culprit, I install OpenSIPS 1.9 with default script and using fix_route_dialog() before relay BYE and everything run fine</div><div><br></div><div><div> if (is_method("BYE")) {</div>
<div> setflag(ACC_DO); # do accounting ...</div><div> setflag(ACC_FAILED); # ... even if the transaction fails</div></div><div><div> fix_route_dialog();</div>
<div> route(relay);</div></div><div> }<br></div><div><br></div><div>But if I do this to OpenSIPS 1.10, it can't relay BYE to client in the other side. Is it a bug or not?</div>
<div><br></div><div>Thank for reading</div></div>