[OpenSIPS-Users] Problems with dialplan configuration --- strange invite response

[thomas fitzgibbon]

*Hello, I am quite new to opensips.*
*My current task requires complete transformation of DIDs.*
*I am using opensips 1.6 and the dialplan module.*
*The DID is transformed correctly but I get a strange response to the sip
invite (see below)*
*
*
*I assume its a simple config issue, I havn't been able to find many
drouting examples online and most of them involve variables which I dont
understand. *
*
*
*Any hints appreciated*
*
*
*relevant configs:*

modparam ("dialplan", "db_url", "mysql://opensips:opensipsrw@localhost
/opensips")
modparam ("dialplan", "table_name", "dialplan")

*
*
*This is how I call dp_translate *

 if (src_ip==4.4.4.4) || (src_ip==5.5.5.5) {
        dp_translate ("1");                                 * ##Do I need
more parameters here?*
        rewritehostport( "6.6.6.6:5061");
        route(1);

        }

*dialplan rule:*

'1', '1', '0', '0', '13129245555', '11', '', '16155555555', ''

*the calls is then passed to an asterisk pbx*
*
*
*The main function seems to be working properly, and the invite to asterisk
looks like this **(the dialplan has replaced the DID)*

U 207.182.132.xxx:5060 -> 206.222.7.xxx:5061
INVITE sip:16155555555 at 206.222.7.xxx:5061;user=phone SIP/2.0.
Record-Route: <sip:207.182.132.xxx;lr=on>.
Record-Route: <sip:216.66.79.xx;lr;ftag=gK0f629023;did=012.00d5527>.
Via: SIP/2.0/UDP 207.182.132.xxx;branch=z9hG4bK59c8.843a2614.0.
Via: SIP/2.0/UDP 216.66.79.xx;branch=z9hG4bK59c8.4a5dcc82.0.
Via: SIP/2.0/UDP 74.120.95.xxx:5060;branch=z9hG4bK0fB2d616c8782204e36.
From: "ttmmff11" <sip:+16617480xxx at 74.120.95.xxx;user=phone>;tag=gK0f629023.
To: <sip:+13129245555 at 216.66.79.xx;user=phone>.
Call-ID: 1963953326_77183542 at 74.120.95.xxx.
CSeq: 29022 INVITE.
Max-Forwards: 64.
Allow:
INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS.
Accept: application/sdp, application/isup, application/dtmf,
application/dtmf-relay,  multipart/mixed.
Contact: "ttmmff11" <sip:+16617480240 at 74.120.95.xxx:5060>.
Supported: timer,100rel,replaces.
Session-Expires: 1800.
Min-SE: 90.
Content-Length:  234.
Content-Disposition: session; handling=required.
Content-Type: application/sdp.
.
v=0.
o=Sonus_UAC 9864 544 IN IP4 74.120.95.195.
s=SIP Media Capabilities.
c=IN IP4 74.120.95.199.
t=0 0.
m=audio 8786 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=sendrecv.
a=maxptime:20.

*But the response from asterisk has strange formatting and the invite from
opensips keeps looping*
*
*
I 206.222.7.xxx -> 207.182.132.xxx 3:3
....E..... at .=.................b.INVITE
sip:16155555555 at 206.222.7.xxx:5061;user=phone
SIP/2.0.
Record-Route: <sip:207.182.132.xxx;lr=on>.
Record-Route: <sip:216.66.79.xx;lr;ftag=gK0f629023;did=012.00d5527>.
Via: SIP/2.0/UDP 207.182.132.xxx;branch=z9hG4bK59c8.843a2614.0.
Via: SIP/2.0/UDP 216.66.79.xx;branch=z9hG4bK59c8.4a5dcc82.0.
Via: SIP/2.0/UDP 74.120.95.xxx:5060;branch=z9hG4bK0fB2d616c8782204e36.
From: "ttmmff11" <sip:+16617480240 at 74.120.95.xxx;user=phone>;tag=gK0f629023.
To: <sip:+13129245555 at 216.66.79.xx;user=phone>.
Call-ID: 1963953326_77183
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