What causes an SPA112 to send SIP 410 in response to an invite?

Steven Wheeler · ‎06-14-2012

One of our customers had an issue this morning where the ATA appeared to be online (responsive to SIP Options requests) but would always send a SIP 410 response when it received an invite. The problem was ultimately resolved when they rebooted the device, but the user was very upset that their phone wasn't working. We have been unable to reproduce the problem in our lab.

Here is the SIP output captured with tcpdump:

11:23:39.375199 IP (tos 0x0, ttl  64, id 57660, offset 0, flags [none], proto: UDP (17), length: 1092) [SERVER IP].sip > [PHONE IP].sip: SIP, length: 1064
        INVITE sip:[PHONE USERNAME]@[PHONE IP]:5060 SIP/2.0
        Via: SIP/2.0/UDP [SERVER IP]:5060;branch=z9hG4bK67f78aa6;rport
        Max-Forwards: 70
        From: "[CALLER]" <sip:[CALLER]@[SERVER IP]>;tag=as17df6e99
        To: <sip:[PHONE USERNAME]@[PHONE IP]:5060>
        Contact: <sip:[CALLER]@[SERVER IP]:5060>
        Call-ID: 048c60e348832ebb64aefffe4f4cd4e4@[SERVER IP]:5060
        CSeq: 102 INVITE
        User-Agent: Asterisk PBX 1.8.9.2
        Date: Thu, 14 Jun 2012 16:23:39 GMT
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
        Supported: replaces, timer
        Content-Type: application/sdp
        Content-Length: 436
        v=0
        o=root 872600236 872600236 IN IP4 [SERVER IP]
        s=Asterisk PBX 1.8.9.2
        c=IN IP4 [SERVER IP]
        b=CT:384
        t=0 0
        m=audio 19422 RTP/AVP 0 9 8 3 101
        a=rtpmap:0 PCMU/8000
        a=rtpmap:9 G722/8000
        a=rtpmap:8 PCMA/8000
        a=rtpmap:3 GSM/8000
        a=rtpmap:101 telephone-event/8000
        a=fmtp:101 0-16
        a=ptime:20
        a=sendrecv
        m=video 19050 RTP/AVP 34 98 99
        a=rtpmap:34 H263/90000
        a=rtpmap:98 h263-1998/90000
        a=rtpmap:99 H264/90000
        a=sendrecv
11:23:39.485648 IP (tos 0x68, ttl  53, id 0, offset 0, flags [DF], proto: UDP (17), length: 349) [PHONE IP].sip > [SERVER IP].sip: SIP, length: 321
        SIP/2.0 410 Gone
        To: <sip:[PHONE USERNAME]@[PHONE IP]:5060>;tag=97169c6f1f663328i0
        From: "[CALLER]" <sip:[CALLER]@[SERVER IP]>;tag=as17df6e99
        Call-ID: 048c60e348832ebb64aefffe4f4cd4e4@[SERVER IP]:5060
        CSeq: 102 INVITE
        Via: SIP/2.0/UDP [SERVER IP]:5060;branch=z9hG4bK67f78aa6
        Content-Length: 0
11:23:39.486392 IP (tos 0x0, ttl  64, id 57661, offset 0, flags [none], proto: UDP (17), length: 492) [SERVER IP].sip > [PHONE IP].sip: SIP, length: 464
        ACK sip:[PHONE USERNAME]@[PHONE IP]:5060 SIP/2.0
        Via: SIP/2.0/UDP [SERVER IP]:5060;branch=z9hG4bK67f78aa6;rport
        Max-Forwards: 70
        From: "[CALLER]" <sip:[CALLER]@[SERVER IP]>;tag=as17df6e99
        To: <sip:[PHONE USERNAME]@[PHONE IP]:5060>;tag=97169c6f1f663328i0
        Contact: <sip:[CALLER]@[SERVER IP]:5060>
        Call-ID: 048c60e348832ebb64aefffe4f4cd4e4@[SERVER IP]:5060
        CSeq: 102 ACK
        User-Agent: Asterisk PBX 1.8.9.2
        Content-Length: 0

Daniele Giordano · ‎06-15-2012

I've done a fast search in google about "linksys sip 410".

Seems to be linked to this option "Ans Call Without Reg".

If setted to Yes fixes the problem.

Sounds strange.

Regards.

Steven Wheeler · ‎06-15-2012

Thanks for the quick response. I will try that setting. Are there any security risks associated with allowing inbound calls without registration?

Daniele Giordano · ‎06-15-2012

I think no but I suggest you to enable the "Restrict Source IP" option.

Regards.

Steven Wheeler · ‎06-15-2012

Great, we actually already have that option enabled. Thanks for your help.