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[ohjelmistoarkkitehti a...]

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@testers.com (660@testers.com) as a websocket client and 700@testers.com (700@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@testers.com ([email]sip%3A660@testers.com[/email]);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@2.2.2.2:37730;transport=UDP>
        To: <sip:660@testers.com ([email]sip%3A660@testers.com[/email]);transport=UDP>
        From: <sip:700@testers.com ([email]sip%3A700@testers.com[/email]);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@testers.com ([email]sip%3A700@testers.com[/email]);transport=UDP>;tag=fd070807
        To: <sip:660@testers.com ([email]sip%3A660@testers.com[/email]);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@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@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@testers.com ([email]sip%3A660@testers.com[/email])>;tag=as7b7c32a5
        To: <sip:660@1.1.1.1:5060>
        Contact: <sip:660@1.1.1.1:5070>
        Call-ID: 7240b8a011890ec677f185f4548583f4@testers.com (7240b8a011890ec677f185f4548583f4@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@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@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

[jcolp at digium.com]

Olli Heiskanen wrote:

[ohjelmistoarkkitehti a...]

Hi,

There we go, that was it. Thank you Joshua! 


cheers,
Olli



2014-08-06 15:26 GMT+03:00 Joshua Colp <jcolp@digium.com (jcolp@digium.com)>: