Nick Cameo
2014-Aug-12 11:49 UTC
[asterisk-users] Asterisk seding 2 INVITEs all of a sudden
Hello Everyone, Today we observed asterisk sending two invites for the initial call before the call was established (ie, not re-invites). There were no changes made to the configuration for a very long time, and was kind of confused when seeing this action. Can someone please suggest where to look to remove this behaviour? U 2014/08/12 07:34:20.405029 192.168.2.10:5060 -> 192.168.2.20:5080 INVITE sip:873359633037 at 192.168.2.20:5080 SIP/2.0. Via: SIP/2.0/UDP 192.168.2.10:5060;branch=z9hG4bK541d5594;rport. Max-Forwards: 70. From: "555955599" <sip:555955599 at victoria.example.com>;tag=as285d2896. To: <sip:873359633037 at 192.168.2.20:5080>. Contact: <sip:555955599 at 192.168.2.10:5060>. Call-ID: 5a51eef8064a0d360009f64e34c7007a at victoria.example.com. CSeq: 102 INVITE. User-Agent: EXAMPLE Systems. Date: Tue, 12 Aug 2014 11:34:20 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Content-Type: application/sdp. Content-Length: 279. . v=0. o=root 1631923320 1631923320 IN IP4 192.168.2.10. s=EXAMPLE Systems. c=IN IP4 192.168.2.10. t=0 0. m=audio 52034 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. U 2014/08/12 07:34:20.903830 192.168.2.10:5060 -> 192.168.2.20:5080 INVITE sip:873359633037 at 192.168.2.20:5080 SIP/2.0. Via: SIP/2.0/UDP 192.168.2.10:5060;branch=z9hG4bK541d5594;rport. Max-Forwards: 70. From: "555955599" <sip:555955599 at victoria.example.com>;tag=as285d2896. To: <sip:873359633037 at 192.168.2.20:5080>. Contact: <sip:555955599 at 192.168.2.10:5060>. Call-ID: 5a51eef8064a0d360009f64e34c7007a at victoria.example.com. CSeq: 102 INVITE. User-Agent: EXAMPLE Systems. Date: Tue, 12 Aug 2014 11:34:20 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Content-Type: application/sdp. Content-Length: 279. . v=0. o=root 1631923320 1631923320 IN IP4 192.168.2.10. s=EXAMPLE Systems. c=IN IP4 192.168.2.10. t=0 0. m=audio 52034 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. Thanks in Advance, Nick
Scott Griepentrog
2014-Aug-12 19:41 UTC
[asterisk-users] Asterisk seding 2 INVITEs all of a sudden
?There is right at 500 ms between the two invites. You are seeing a retransmission due to a lack of response to the first INVITE in time. This is normal, correct, and expected behavior. The retransmission can occur even sooner in the case where QUALIFY is used to determine that the endpoint usually responds faster. ? On Tue, Aug 12, 2014 at 6:49 AM, Nick Cameo <symack at gmail.com> wrote:> Hello Everyone, > > Today we observed asterisk sending two invites for the initial call before > the call was established (ie, not re-invites). There were no changes made > to the configuration for a very long time, and was kind of confused when > seeing this action. Can someone please suggest where to look to remove > this behaviour? > > U 2014/08/12 07:34:20.405029 192.168.2.10:5060 -> 192.168.2.20:5080 > INVITE sip:873359633037 at 192.168.2.20:5080 SIP/2.0. > Via: SIP/2.0/UDP 192.168.2.10:5060;branch=z9hG4bK541d5594;rport. > Max-Forwards: 70. > From: "555955599" <sip:555955599 at victoria.example.com>;tag=as285d2896. > To: <sip:873359633037 at 192.168.2.20:5080>. > Contact: <sip:555955599 at 192.168.2.10:5060>. > Call-ID: 5a51eef8064a0d360009f64e34c7007a at victoria.example.com. > CSeq: 102 INVITE. > User-Agent: EXAMPLE Systems. > Date: Tue, 12 Aug 2014 11:34:20 GMT. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, > INFO, PUBLISH. > Supported: replaces, timer. > Content-Type: application/sdp. > Content-Length: 279. > . > v=0. > o=root 1631923320 1631923320 IN IP4 192.168.2.10. > s=EXAMPLE Systems. > c=IN IP4 192.168.2.10. > t=0 0. > m=audio 52034 RTP/AVP 18 101. > a=rtpmap:18 G729/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=silenceSupp:off - - - -. > a=ptime:20. > a=sendrecv. > > > U 2014/08/12 07:34:20.903830 192.168.2.10:5060 -> 192.168.2.20:5080 > INVITE sip:873359633037 at 192.168.2.20:5080 SIP/2.0. > Via: SIP/2.0/UDP 192.168.2.10:5060;branch=z9hG4bK541d5594;rport. > Max-Forwards: 70. > From: "555955599" <sip:555955599 at victoria.example.com>;tag=as285d2896. > To: <sip:873359633037 at 192.168.2.20:5080>. > Contact: <sip:555955599 at 192.168.2.10:5060>. > Call-ID: 5a51eef8064a0d360009f64e34c7007a at victoria.example.com. > CSeq: 102 INVITE. > User-Agent: EXAMPLE Systems. > Date: Tue, 12 Aug 2014 11:34:20 GMT. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, > INFO, PUBLISH. > Supported: replaces, timer. > Content-Type: application/sdp. > Content-Length: 279. > . > v=0. > o=root 1631923320 1631923320 IN IP4 192.168.2.10. > s=EXAMPLE Systems. > c=IN IP4 192.168.2.10. > t=0 0. > m=audio 52034 RTP/AVP 18 101. > a=rtpmap:18 G729/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=silenceSupp:off - - - -. > a=ptime:20. > a=sendrecv. > > Thanks in Advance, > > Nick > > -- > _____________________________________________________________________ > -- 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 >-- [image: Digium logo] Scott Griepentrog Digium, Inc ? Software Developer 445 Jan Davis Drive NW ? Huntsville, AL 35806 ? US direct/fax: +1 256 428 6239 ? mobile: +1 317 507 4029 Check us out at: http://digium.com ? http://asterisk.org -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140812/6c357476/attachment.html>