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 -> 2.2.2.2:5060 INVITE 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@1.1.1.1. To: sip:+4212333323@1.1.1.1. Contact: sip:anonymous@1.1.1.5:5060. Call-ID: 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@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@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@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.
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@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@1.1.1.1 mailto:sip%3Aunavailable@1.1.1.1>. To: <sip:+4212333323@1.1.1.1 mailto:sip%3A%2B4212333323@1.1.1.1>. Contact: <sip:anonymous@1.1.1.5:5060 http://sip:anonymous@1.1.1.5:5060>. Call-ID: 0df8db614d45bae27035443c35166ba6@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@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@2.2.2.2 mailto:sip%3A%2B4212333323@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@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.
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