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:c2e74184-4d23-43a3-8fd9-26ba2c5ef6c9 at 192.168.53.9:5060 SIP/2.0 Max-Forwards: 69 To: <sip:4959810128 at 192.168.53.9>;tag=b3769af4-118b-4467-8c95-042247ff1776 From: <sip:84957774888 at 192.168.53.1>;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:84957774888 at 192.168.53.1:5060> 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
Nick Awesome 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:c2e74184-4d23-43a3-8fd9-26ba2c5ef6c9 at 192.168.53.9:5060 SIP/2.0 > Max-Forwards: 69 > To:<sip:4959810128 at 192.168.53.9>;tag=b3769af4-118b-4467-8c95-042247ff1776 > From:<sip:84957774888 at 192.168.53.1>;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:84957774888 at 192.168.53.1:5060> > Content-Length: 0Looks like they are using INFO as a keepalive mechanism. Since we don't answer this it'd be a bug. File an issue on the issue tracker[1]. [1] https://issues.asterisk.org/jira -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org
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 <jleed at me.com> 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:c2e74184-4d23-43a3-8fd9-26ba2c5ef6c9 at 192.168.53.9:5060 SIP/2.0 > Max-Forwards: 69 > To: <sip:4959810128 at 192.168.53.9>;tag=b3769af4-118b-4467-8c95-042247ff1776 > From: <sip:84957774888 at 192.168.53.1>;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:84957774888 at 192.168.53.1:5060> > 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 >
Nick Awesome wrote:> 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 ?INFO support is currently implemented only for DTMF in the res_pjsip_dtmf_info module. This module is located at res/res_pjsip_dtmf_info.c It could be used as a base to implement this, or extended to support it. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org