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

[thomas fitzgibbon]

to clarify please translate "6.6.6.6" to "206.222.7.xxx"


On Sun, Jun 9, 2013 at 1:41 AM, thomas fitzgibbon <
anning.marketing at gmail.com> 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: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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