here we are :)
88.88.88.84 is the asterisk generating the call 88.88.88.83 is SER 88.88.88.89 is a pstn gateway
we have 88.88.88.84 --> 88.88.88.83 --> 88.88.88.89 asterisk SER Pstn gateway
initial invite going to ser from asterisk
U 88.88.88.84:5060 -> 88.88.88.83:5060 INVITE sip:003227321073@toto.com SIP/2.0. Via: SIP/2.0/UDP 88.88.88.84:5060;branch=z9hG4bK6c6f6e3a;rport. From: "7321073" sip:123456@toto.com;tag=as66857880. To: sip:003227321073@finalcut.be. Contact: sip:123456@88.88.88.84. Call-ID: 79c32ec07d327db37ddab2fb211cbd4a@toto.com. CSeq: 103 INVITE. User-Agent: Phonext Pbx. Max-Forwards: 70. Proxy-Authorization: Digest username="123456", realm="finalcut.be", algorithm=MD5, uri="sip:003227321073@toto.com", nonce="4422b15f6ee5477116355cd454befe82f159112d", response="d6bf375d764e044e0b9ee6574fa2e650", opaque="". Date: Thu, 23 Mar 2006 14:26:25 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Content-Type: application/sdp. Content-Length: 310. . v=0. o=root 1042 1043 IN IP4 88.88.88.84. s=session. c=IN IP4 88.88.88.84. t=0 0. m=audio 18244 RTP/AVP 18 0 8 3 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:3 GSM/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -.
ngrep after transformation from SER going to the pstn gateway avp_write("$from/username", "$ocn"); avp_pushto("$X-From", "$ocn");
U 88.88.88.83:5060 -> 88.88.88.89:5060 INVITE sip:3227321073@88.88.88.89:5060 SIP/2.0. Record-Route: sip:003227321073@88.88.88.83:5060;nat=yes;ftag=as66857880;lr=on. Via: SIP/2.0/UDP 88.88.88.83;branch=z9hG4bKd636.284afc5.0. Via: SIP/2.0/UDP 88.88.88.84:5060;branch=z9hG4bK6c6f6e3a;rport=5060. From: "7321073" sip:123456@toto.com;tag=as66857880. To: sip:003227321073@toto.com. Contact: sip:123456@88.88.88.84. Call-ID: 79c32ec07d327db37ddab2fb211cbd4a@toto.com. CSeq: 103 INVITE. User-Agent: Phonext Pbx. Max-Forwards: 16. Date: Thu, 23 Mar 2006 14:26:25 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Content-Type: application/sdp. Content-Length: 330. X-Feature: WebCall. X-Forwarded: Yes. X-From: 123456. . v=0. o=root 1042 1043 IN IP4 88.88.88.84. s=session. c=IN IP4 88.88.88.84. t=0 0. m=audio 18244 RTP/AVP 18 0 8 3 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:3 GSM/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=direction:active.
Steve Blair a écrit :
Do you have a ngrep snapshot of this packet ?
olivier.taylor wrote:
Hi Steve,
first of all, thanks for a so fast answer :)
The problem is the result I get, after the avp_write, I use avp_pushto to create a new header (to be used for the billing).
avp_write("$from/username", "$ocn"); avp_pushto("$X-From", "$ocn");and I get : X-From: 69 as header, I 'dlike to have X-From: 45454558
Olivier
Steve Blair a écrit :
olivier.taylor wrote:
hi all,
Using ser 0.94 I have a sip message coming from an asterisk containing : From: "45454558" sip:69@toto.com;tag=as2524cec1. I need to extract the 45454558 and add it in a new header.
Did you try something like:
modparam("avpops", "avp_aliases","ocn=i:701") avp_write("$from/username", "$ocn");
Then use $ocn as an alias whereever you need to reference this username.
-Steve
I there a way to do that with avpops?
I tried avp_write("$from", "s:from"); and get sip:69@toto.com also avp_write("$from/username", "s:from"); and get 69
is there another way?
Cheers,
Olivier
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