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Olli Heiskanen

2014-Aug-06 10:28 UTC

[asterisk-users] From and To headers contain same account in INVITEs

Hello,

I noticed a strange thing while testing my Asterisk-Kamailio Realtime
setup. In an INVITE the From and To headers contain the same number when
calling through a Realtime integration setup. This happens when the INVITE
leaves Asterisk.

Can you guys tell me what might be causing this? I have 660 at testers.com as
a websocket client and 700 at testers.com (caller) using a Zoiper client (db
output below). The call itself works, audio and all, only those headers are
puzzling to me. I noticed this when I tried to add a label saying '700
calling' on my web page. The same thing happens when I call from 660 to
700.

My Asterisk is 11.11.0 running on CentOS 6.5.

An INVITE is sent from my client to Kamailio and then to Asterisk:
(both Kamailio and Asterisk are at 1.1.1.1)

        INVITE sip:660 at testers.com;transport=UDP SIP/2.0
        Record-Route: <sip:1.1.1.1;lr=on;ftag=fd070807>
        Via: SIP/2.0/UDP
1.1.1.1;branch=z9hG4bKf6e9.339dda0648d95af665c91db701754d98.0
        Via: SIP/2.0/UDP 2.2.2.2:37730
;rport=37730;branch=z9hG4bK-d8754z-7f27c9fc35574abb-1---d8754z-
        Max-Forwards: 16
        Contact: <sip:700 at 2.2.2.2:37730;transport=UDP>
        To: <sip:660 at testers.com;transport=UDP>
        From: <sip:700 at testers.com;transport=UDP>;tag=fd070807
        Call-ID: ZDc0YjU1ZjNmMWI5YjUyYzY0YWNjN2NjN2NkODg2OTk.
        CSeq: 2 INVITE
        Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS,
INFO, SUBSCRIBE
        Content-Type: application/sdp
        Supported: replaces, norefersub, extended-refer, timer,
X-cisco-serviceuri
        User-Agent: Z 3.2.21357 r21367
        Allow-Events: presence, kpml
        Content-Length: 239

        v=0
        o=Z 0 0 IN IP4 2.2.2.2
        s=Z
        c=IN IP4 2.2.2.2
        t=0 0
        m=audio 8000 RTP/AVP 3 110 8 0 98 101
        a=rtpmap:110 speex/8000
        a=rtpmap:98 iLBC/8000
        a=fmtp:98 mode=20
        a=rtpmap:101 telephone-event/8000
        a=fmtp:101 0-15
        a=sendrecv

... and Asterisk responds with Trying:

        SIP/2.0 100 Trying
        Via: SIP/2.0/UDP
1.1.1.1;branch=z9hG4bKf6e9.339dda0648d95af665c91db701754d98.0;received=1.1.1.1;rport=5060
        Via: SIP/2.0/UDP 2.2.2.2:37730
;rport=37730;branch=z9hG4bK-d8754z-7f27c9fc35574abb-1---d8754z-
        Record-Route: <sip:1.1.1.1;lr=on;ftag=fd070807>
        From: <sip:700 at testers.com;transport=UDP>;tag=fd070807
        To: <sip:660 at testers.com;transport=UDP>
        Call-ID: ZDc0YjU1ZjNmMWI5YjUyYzY0YWNjN2NjN2NkODg2OTk.
        CSeq: 2 INVITE
        Server: I Am the Devil
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
        Supported: replaces, timer
        Session-Expires: 1800;refresher=uas
        Contact: <sip:660 at 1.1.1.1:5070>
        Content-Length: 0

And when Asterisk sends out the INVITE, From and To headers both have the
same number:

        INVITE sip:660 at 1.1.1.1:5060 SIP/2.0
        Via: SIP/2.0/UDP 1.1.1.1:5070;branch=z9hG4bK75de61d0;rport
        Max-Forwards: 70
        From: <sip:660 at testers.com>;tag=as7b7c32a5
        To: <sip:660 at 1.1.1.1:5060>
        Contact: <sip:660 at 1.1.1.1:5070>
        Call-ID: 7240b8a011890ec677f185f4548583f4 at testers.com
        CSeq: 102 INVITE
        User-Agent: I Am the Devil
        Date: Wed, 06 Aug 2014 09:54:35 GMT
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
        Supported: replaces, timer
        Content-Type: application/sdp
        Content-Length: 801

        v=0
        o=root 969416519 969416519 IN IP4 1.1.1.1
        s=Asterisk PBX 11.11.0
        c=IN IP4 1.1.1.1
        t=0 0
        m=audio 18740 RTP/SAVPF 0 3 8 101
        a=rtpmap:0 PCMU/8000
        a=rtpmap:3 GSM/8000
        a=rtpmap:8 PCMA/8000
        a=rtpmap:101 telephone-event/8000
        a=fmtp:101 0-16
        a=ptime:20
        a=ice-ufrag:50d777041673316422560b90281fcd2e
        a=ice-pwd:0093fdde724f8a411742661c31c90f21
        a=candidate:H5bdd423d 1 UDP 2130706431 1.1.1.1 18740 typ host
        a=candidate:S5bdd423d 1 UDP 1694498815 1.1.1.1 18740 typ srflx
        a=candidate:H5bdd423d 2 UDP 2130706430 1.1.1.1 18741 typ host
        a=candidate:S5bdd423d 2 UDP 1694498814 1.1.1.1 18742 typ srflx
        a=connection:new
        a=setup:actpass
        a=fingerprint:SHA-256
CE:EE:D9:28:EA:B0:6E:D0:CE:4F:5A:9A:FB:53:66:74:83:47:18:37:2F:76:C1:6D:10:C0:EE:FF:A4:56:F4:05
        a=sendrecv


Here's the dialplan, nothing special:

exten => _XXX,1,NoOp(general : Dialed ${EXTEN})
 same => n,Dial(SIP/${EXTEN},3600,rt)
 same => n,Hangup


And here's how the clients are set in my db:

            id: 4
          name: 660
        ipaddr: 1.1.1.1
          port: 5060
    regseconds: 1407320692
   defaultuser: 660
   fullcontact: sip:660 at 1.1.1.1:5060
     regserver:
     useragent:
        lastms: 0
          host: dynamic
          type: friend
       context: default
          deny: 0.0.0.0/0.0.0.0
        permit: 1.1.1.1
        secret: NULL
     md5secret: NULL
          avpf: yes
     force_avp: yes
    icesupport: yes
   directmedia: no
    encryption: yes
           nat: force_rport,comedia
     callgroup: NULL
   pickupgroup: NULL
      language: NULL
      disallow: NULL
         allow: NULL
        setvar: NULL
      callerid: NULL
      amaflags: NULL
  videosupport: no
maxcallbitrate: NULL
       mailbox: NULL
      regexten: NULL
    fromdomain: testers.com
      fromuser: 660
       qualify: NULL
     defaultip: NULL
 outboundproxy: 1.1.1.1
 contactpermit: NULL
   contactdeny: NULL
      fullname: NULL
    cid_number: NULL
   callingpres: NULL
  mohinterpret: NULL
    mohsuggest: NULL
  hasvoicemail: NULL
  subscribemwi: NULL
       vmexten: NULL
  rtpkeepalive: NULL
directrtpsetup: yes
    dtlsenable: yes
    dtlsverify: no
dtlsprivatekey: /etc/asterisk/keys/asterisk.pem
     dtlssetup: actpass
  dtlscertfile: /etc/asterisk/keys/asterisk.pem
    dtlscafile: /etc/asterisk/keys/ca.crt
     sippasswd: a84a4ddcda13d13c9573d5294047b6a2
          rpid: NULL
        domain: testers.com
    sippasswd2: NULL

            id: 8
          name: 700
        ipaddr: 1.1.1.1
          port: 5060
    regseconds: 1407323638
   defaultuser: 700
   fullcontact: sip:700 at 1.1.1.1:5060
     regserver:
     useragent:
        lastms: 0
          host: dynamic
          type: friend
       context: default
          deny: 0.0.0.0/0.0.0.0
        permit: 1.1.1.1
        secret: NULL
     md5secret: NULL
          avpf: no
     force_avp: NULL
    icesupport: NULL
   directmedia: NULL
    encryption: NULL
           nat: force_rport,comedia
     callgroup: NULL
   pickupgroup: NULL
      language: NULL
      disallow: NULL
         allow: NULL
        setvar: NULL
      callerid: NULL
      amaflags: NULL
  videosupport: yes
maxcallbitrate: NULL
       mailbox: NULL
      regexten: NULL
    fromdomain: testers.com
      fromuser: 700
       qualify: NULL
     defaultip: NULL
 outboundproxy: 1.1.1.1
 contactpermit: NULL
   contactdeny: NULL
      fullname: NULL
    cid_number: NULL
   callingpres: NULL
  mohinterpret: NULL
    mohsuggest: NULL
  hasvoicemail: NULL
  subscribemwi: NULL
       vmexten: NULL
  rtpkeepalive: NULL
directrtpsetup: NULL
    dtlsenable: NULL
    dtlsverify: NULL
dtlsprivatekey: NULL
     dtlssetup: NULL
  dtlscertfile: NULL
    dtlscafile: NULL
     sippasswd: 2ef16ba6cda5dcd34088f4127b90048b
          rpid: NULL
        domain: testers.com
    sippasswd2: NULL




cheers,
Olli
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Joshua Colp

2014-Aug-06 12:26 UTC

head link

[asterisk-users] From and To headers contain same account in INVITEs

Olli Heiskanen wrote:> Hello,
Kia ora,
> I noticed a strange thing while testing my Asterisk-Kamailio Realtime
> setup. In an INVITE the From and To headers contain the same number when
> calling through a Realtime integration setup. This happens when the
> INVITE leaves Asterisk.
>
> Can you guys tell me what might be causing this? I have 660 at testers.com
> <mailto:660 at testers.com> as a websocket client and 700 at
testers.com
> <mailto:700 at testers.com> (caller) using a Zoiper client (db output
> below). The call itself works, audio and all, only those headers are
> puzzling to me. I noticed this when I tried to add a label saying '700
> calling' on my web page. The same thing happens when I call from 660 to
> 700.
Your configuration has "fromuser" set which explicitly sets the user 
portion of the From header to what you specify. This is commonly used 
for ITSPs as they use that to determine who you are trying to 
authenticate as. If you require this to be set then caller id 
information has to be transported in a different manner (RPID or PAI).

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

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