May someone help with the sourcecode, trying find where can I manually send response on Received INFO request in PJSIP
ASTERISK-24986 issues opened already more the 2 month and calls from customers still drops. very annoying :( maybe some one could help me figure out where Received INFO request dies in source so I could patch it to response 200 OK ? > On 20 Apr 2015, at 15:08, Nick Awesome <[email protected]> wrote: > > Hi guys, have really annoying problem with trunks when I calling over voip > provider.. > > > after awhile provider sends INFO packages but for some reason Asterisk > doesn’t answer on it. > after 8 packagers provider just drops the call, here is the package > > <--- Received SIP request (555 bytes) from UDP:192.168.53.1:5060 ---> > INFO sip:[email protected]:5060 SIP/2.0 > Max-Forwards: 69 > To: <sip:[email protected]>;tag=b3769af4-118b-4467-8c95-042247ff1776 > From: <sip:[email protected]>;tag=3638518512-132845 > Call-ID: cfe34652-14c2-4072-9dea-f0b0c30cb15e > CSeq: 2 INFO > Allow: CANCEL, ACK, INVITE, BYE, OPTIONS, REGISTER, NOTIFY, INFO, REFER, > SUBSCRIBE, PRACK, UPDATE, MESSAGE, PUBLISH > Via: SIP/2.0/UDP > 192.168.53.1:5060;branch=z9hG4bK8da58cede20b91eb54dec15ad27f866c > Contact: <sip:[email protected]:5060> > Content-Length: 0 > > 192.168.53.1 - operator IP > 192.168.53.9 - asterisk IP > > > Any idea how to fix this? > > > have 2 Ethernet interfaces: > 192.168.1.4 - local network > 192.168.53.9 - VOIP Provider network > > Im using PJSIP, here is config: > > [udp] > type=transport > protocol=udp > bind=192.168.1.4 > local_net=10.0.0.0/24 > local_net=10.0.1.0/24 > local_net=192.168.1.0/24 > > external_media_address=195.239.8.122 > external_signaling_address=195.239.8.122 > > [udp_B] > type=transport > protocol=udp > bind=192.168.53.9 > > [10000] > type=endpoint > aors=10000 > context=dialmap > disallow=all > allow=alaw,ulaw > transport=udp_B > > [10000] > type=aor > contact=sip:192.168.53.1:5060 > max_contacts=4 > -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
