SIP 420

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Asterisk Users 4 Comments

Hi;

I am running asterisk 1.6 from Fonality (Trixbox PRO).

I am trying to initiate a call FROM a softphone client to asterisk (either
an internal 4 digit extension call) or an outside line via a SIP trunk.

In both cases, asterisk rejects the call with a 420.

In this case, it’s a call from x3992 to x4415

Does this require a change on the softphone for x-call-detail?

<--- SIP read from UDP://x.x.x.x:5060 http://10.247.1.126:5060 —>

INVITE sip:4415@x.x.x.x:5060;transport=udp
SIP/2.0

To:
>

From:
>;tag=4f5cb549

Via: SIP/2.0/UDP
x.x.x.x:5060;branch=z9hG4bK-d87543-62412f606c62824d-1–d87543-;rport

Call-ID: 350da2493d160e6f@ZHQtZGVsbDN2ZHIwZjEuQU0uSURUQ09SUC5ORVQ.

CSeq: 1 INVITE

Contact:
>

Max-Forwards: 70

Session-Expires: 1800

Min-SE: 90

Accept-Language: en

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY

Content-Type: application/sdp

*Require: x-call-detail*

Supported: timer

User-Agent: xxx PBX Phone 1.1.0.10434 ORIG/IDCB01S110 SN/001d09048211
(Windows NT 5.1)

Content-Length: 426

v=0

o=SIP 1292608808 1292608808 IN IP4 x.x.x.x

s=SIP

c=IN IP4 x.x.x.x

t=1292608808 0

m=audio 10000 RTP/AVP 97 103 100 127 0 8 102 18 4 101

a=rtpmap:97 IPCMWB/16000

a=rtpmap:103 ISAC/16000

a=rtpmap:100 EG711U/8000

a=rtpmap:127 EG711A/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:102 iLBC/8000

a=fmtp:102 mode=30

a=rtpmap:18 G729/8000

a=rtpmap:4 G723/8000

a=rtpmap:101 telephone-event/8000

<------------->

4 thoughts on - SIP 420

  • Yes. The softphone is requiring x-call-detail, which Asterisk does
    not support. The softphone either needs to drop that requirement
    completely, or change it to a Supported header so it can be processed
    by other SIP servers.

    -Jonathan

  • This is pretty clear… your softphone is requiring support for a
    private SIP extension called ‘call-detail’, and since Asterisk does not
    support it, it cannot process the INVITE request.

  • Thanks Kevin.

    Did it work with Asterisk 1.2 because it ignored it?

    Why now?
    x.x.x.x:5060;branch=z9hG4bK-d87543-62412f606c62824d-1–d87543-;received=x.x.x.x;rport=5060
    ;transport=udp>>;tag=as34f3ff9f
    href=”mailto:kfleming@digium.com”>kfleming@digium.com

  • I don’t know specifically that Asterisk 1.2 ignored “Required” headers,
    but it’s certainly possible.