to clarify please translate "6.6.6.6" to "206.222.7.xxx"
On Sun, Jun 9, 2013 at 1:41 AM, thomas fitzgibbon < [email protected]> wrote: > *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:[email protected]: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:[email protected] > ;user=phone>;tag=gK0f629023. > To: <sip:[email protected];user=phone>. > Call-ID: [email protected]. > 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:[email protected]: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.....@.=.................b.INVITE > sip:[email protected]: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:[email protected] > ;user=phone>;tag=gK0f629023. > To: <sip:[email protected];user=phone>. > Call-ID: 1963953326_77183 > > > >
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