[asterisk-dev] Help with outbound dialing problems

[Daniel Corbe]

Dear Asterisk Community.

I am using asterisk realtime.

I'm having an issue with outbound calls from my Asterisk box to a PSTN
gateway in my network.  The asterisk box is sticking its IP address in
the username field in the From: header of the SIP message.  This is
causing all sorts of issues (including caller ID not being delivered,
the PSTN gateway not recognizing which account the asterisk box is
trying to impersonate, etc)

This problem exists regardless of what I set callerid= in my sip
configuration.  Any help is greatly appriciated.  Thanks.

This is the Dial() command that gets called in my dialplan and the
corresponding SIP entry:

 id     context         exten          priority     app
appdata
 -----  --------------  -------------  -----------  -------------
--------------------------------------
16     1892-outbound   _91NXXNXXXXXX  1            dial
SIP/${EXTEN:1}@1892-outbound

 id     name           accountcode     amaflags     callgroup
callerid       canreinvite     context        defaultip     dtmfmode
  fromuser     fromdomain     host         insecure     language
mailbox     md5secret     nat     permit     deny     mask
pickupgroup     port     qualify     restrictcid     rtptimeout
rtpholdtimeout     secret     type     username     disallow     allow
          musiconhold     regseconds     ipaddr     regexten
cancallforward     fullcontact
 -----  -------------  --------------  -----------  ------------
-------------  --------------  -------------  ------------
-----------  -----------  -------------  -----------  -----------
-----------  ----------  ------------  ------  ---------  -------
-------  --------------  -------  ----------  --------------
-------------  -----------------  ---------  -------  -----------
-----------  --------------  --------------  -------------  ---------
-----------  -----------------  --------------
 3      1892-outbound  (null)          (null)       (null)
1892-outbound  yes             1892-outbound  (null)        (null)
  (null)       (null)         64.49.129.5  (null)       (null)
(null)      (null)        no      (null)     (null)   (null)   (null)
                 (null)      (null)          (null)         (null)
        testing    friend                all          g729;ulaw;alaw
(null)          0                                      yes
   (null)




INVITE sip:19549212400 at 64.49.129.5 SIP/2.0
Via: SIP/2.0/UDP 12.109.47.235:5060;branch=z9hG4bK641d7bbb;rport
From: "Daniel Corbe" <sip:1210947235 at 12.109.47.235>;tag=as23956372
To: <sip:19549212400 at 64.49.129.5>
Contact: <sip:1210947235 at 12.109.47.235>
Call-ID: 5a5e089016b9b347492898fd583674d8 at 12.109.47.235
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 05 Jul 2006 15:37:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 287

v=0
o=root 5422 5422 IN IP4 12.109.47.235
s=session
c=IN IP4 12.109.47.235
t=0 0
m=audio 14544 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -