[SR-Users] Problem with to: header

[Daniel-Constantin Mierla]

Hello,

look at the uac module for uac_replace_from() and uac_replace_to() 
functions.

Btw, rfc3261 mandates a tag parameter for From header, which is missing 
on the INVITE you pasted here, so it is rather broken and many UA may 
reject it.

Cheersm
Daniel

On 9/23/13 7:09 PM, julian arsanches wrote:
> Can someone advise me on how to change the to header to show the host 
> that we are sending the call to an not the servers ip.
>
> I am using dispatcher on my setup .
>
> i am getting this
>
>
> U 2013/09/23 12:57:54.576312 10.0.1.206:5060 <http://10.0.1.206:5060> 
> -> 2.2.2.2:5060 <http://2.2.2.2:5060>
> INVITE sip:+42123333235 at 2.2.2.2:5060 
> <http://sip:+42123333235@2.2.2.2:5060> SIP/2.0.
> Record-Route: <sip:1.1.1.1;lr=on;ftag=as1f3df8b3>.
> Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bKb29d.7399c2b3.0.
> Via: SIP/2.0/UDP  1.1.1.5:5060;branch=z9hG4bK42a1ecaf;rport=5060.
> Max-Forwards: 16.
> From:<sip:unavailable at 1.1.1.1 <mailto:sip%3Aunavailable at 1.1.1.1>>.
> To: <sip:+4212333323 at 1.1.1.1 <mailto:sip%3A%2B4212333323 at 1.1.1.1>>.
> Contact: <sip:anonymous at 1.1.1.5:5060 <http://sip:anonymous@1.1.1.5:5060>>.
> Call-ID: 0df8db614d45bae27035443c35166ba6 at 1.1.1.5:5060 
> <http://0df8db614d45bae27035443c35166ba6@1.1.1.5:5060>.
> CSeq: 102 INVITE.
> User-Agent: Asterisk PBX 1.8.15-cert2.
> Date: Mon, 23 Sep 2013 16:58:25 GMT.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
> INFO, PUBLISH.
> Supported: replaces, timer.
> Cisco-Guid: 7128745-3588944267-852064 at msc1
> Content-Type: application/sdp.
> Content-Length: 288.
> .
> v=0.
> o=root 1760548326 1760548326 IN IP4 54.236.97.30.
> s=Asterisk PBX 1.8.15-cert2.
> c=IN IP4 1.1.1.5.
> t=0 0.
> m=audio 39794 RTP/AVP 0 18 101.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:18 G729/8000.
> a=fmtp:18 annexb=no.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=ptime:20.
> a=sendrecv.
>
> I need to  sent header to a carrier like this
>
> To: <sip:+4212333323 at 2.2.2.2 <mailto:sip%3A%2B4212333323 at 2.2.2.2>> 
>  instead of 1.1.1.1.
>
> i am proxing calls from asterisk to a main carrier.
> Please help.
>
> here is my config.
>
>
>
> if (starts_with("$var(o)","anonymous")) {
>
> ds_select_domain("$var(z)", "4");#carrier dynamic
>
>
> xlog("here is anonymous call <$var(o)>77777\n");
>
>
> $var(n)=$(tU{s.substr,3,0});
>
>
> remove_hf("From");
> remove_hf("P-Asserted-Identity");
> remove_hf("Privacy");
>
> insert_hf("From:<sip:unavailable at 1.1.1.1.1>\r\n", "From");
> $tU=$var(n);
> xlog("out header CHECK ANONYMOUS BEFORE to $tu--$td - contact pai+++ 
> <<$ct>>++ from_uri=$fu;<$tU---=$var(n)> to_uri=$tu; 
> }pai<$ai>intid=$fU; type_call=$si; dst_ip=$ru; 
> carriercode=$var(z);callmode=$var(out)");
>
> if(!t_relay()){;
> sl_reply_error();
> exit;
> };
> ##ENDANONYMOUS
>
> exit;
>
> }
>
> Again thanks  a lot for any help.
>
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Trainings - Berlin, Oct 21-24; Miami, Nov 11-13, 2013
   - more details about Kamailio trainings at http://www.asipto.com -

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20130923/8893d64d/attachment-0001.html>