[asterisk-dev] SIPFROMDOMAIN

[Philip A. Prindeville]

On 04/13/2010 11:02 PM, Olle E. Johansson wrote:
> 13 apr 2010 kl. 22.43 skrev Philip A. Prindeville:
>
>   
>> On 04/11/2010 10:09 AM, Olle E. Johansson wrote:
>>     
>>> 10 apr 2010 kl. 23.04 skrev Philip A. Prindeville:
>>>
>>>       
>>>> Olle,
>>>>
>>>> Does SIPFROMDOMAIN take effect on outbound calls?  I'm trying to set it
>>>> to explicitly set my domain when making ISN/Freenum calls.
>>>>         
>>> Not in any release. I have a patch in a branch to set the domain of the From: in the dialplan.
>>> I believe it's in trunk.
>>>
>>> /O
>>>       
>> svn blame | grep SIPFROMUSER
>>     
> The From: Username is the caller ID.
>
> /O
>   


Well, I've been using 1.6.2.6 with the following dialplan fragment:

https://issues.asterisk.org/file_download.php?file_id=25739&type=bug

and even though I'm setting the SIPFROMDOMAIN as:

same => n,GotoIf($["${GLOBAL(FREENUMDOMAIN)}" == ""]?dial:)
same => n,Set(SIPFROMDOMAIN=${GLOBAL(FREENUMDOMAIN)})
same => n(dial),Dial(SIP/${isnresult},40)


I'm seeing:

    -- Executing [278*1122 at outbound-freenum2:7] Set("SIP/guest_1-00000036", "SIPFROMUSER=112") in new stack
    -- Executing [278*1122 at outbound-freenum2:8] GotoIf("SIP/guest_1-00000036", "0?dial:") in new stack
    -- Executing [278*1122 at outbound-freenum2:9] Set("SIP/guest_1-00000036", "SIPFROMDOMAIN=redfish-solutions.com") in new stack
    -- Executing [278*1122 at outbound-freenum2:10] Dial("SIP/guest_1-00000036", "SIP/278 at djhsolutions.net,40") in new stack

  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
       > ast_get_srv: SRV lookup for '_sip._UDP.djhsolutions.net' mapped to host sip.djhsolutions.net, port 5060
Audio is at 66.232.79.143 port 10642
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 99.6.101.6:5060:
INVITE sip:278 at djhsolutions.net SIP/2.0
Via: SIP/2.0/UDP 66.232.79.143:5060;branch=z9hG4bK293422b7;rport
Max-Forwards: 70
From: "Redfish Solutions" <sip:112 at 66.232.79.143>;tag=as2e9ac034
To: <sip:278 at djhsolutions.net>
Contact: <sip:112 at 66.232.79.143>
Call-ID: 53ee2b46432107a475d669df5d74d0b4 at 66.232.79.143
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.6
Date: Wed, 14 Apr 2010 23:46:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 1955298719 1955298719 IN IP4 66.232.79.143
s=Asterisk PBX 1.6.2.6
c=IN IP4 66.232.79.143
t=0 0
m=audio 10642 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---




Note the "From:" line is wrong.

What needs to be fixed?

Thanks,

-Philip