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
