[OpenSIPS-Users] Replace From Field

[Vlad Paiu]

Hello,

The uac_replace_from() just modifies the FROM header, is does not touch 
the Authorization header or any other header.

Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com  



Pe 3/2/2012 10:13 PM, Iulian Macare a scris:
> Hello
>
> My setup is the following:
>
>
> Serveral asterisk boxes (let's name them asteriskG) that register and
> send calls to opensips that balance the outbound calls to other 3 x
> asterisk servers (let's name them asteriskO) that  futher sends the
> calls to my SIP ITSP. ( I have 3 ITSP , each one connected to a
> asteriskO )
>
> So by having this setup , I can balance the calls generated by
> asteriskG to my 3 ITSP providers.
>
>
> What I would like to do is to get rid of the 3 asteriskO server that I
> use to connect to the ITSP and use MediaProxy for rtp relay.
>
> In order to do this I must replace the from field that contains my
> internal usernames&passwords that asteriskG connect to opensips with
> the username&password that my ITSP require
>
> I have tried doing this with uac_replace_from but I keep get SIP/2.0
> 403 Forbidden
>
>
> What I have found is the following:
>
> Let's say that asteriskG ( 192.168.20.66 ) connects to opensips (
> 192.168.4.250 ) with user 9000 and the user for ITSP ( 192.168.4.26 )
> is 30
>
> I tried the following: uac_replace_from("30","sip:30 at 192.168.4.250");
> and the sip log is:
>
> INVITE sip:05954543 at 192.168.4.26 SIP/2.0
> Record-Route:<sip:192.168.4.250;lr=on;ftag=ca42eb7f;vsf=AAAAAAoAcAF5AxcDGAkYDAAGGwI1MA-->
> Via: SIP/2.0/UDP 192.168.4.250;branch=z9hG4bK58a2.12e355c3.0
> Via: SIP/2.0/UDP
> 192.168.20.66:26838;received=192.168.20.66;branch=z9hG4bK-d8754z-2f50b813447d8503-1---d8754z-;rport=26838
> Max-Forwards: 69
> Contact:<sip:9000 at 192.168.20.66:26838>
> To: "05954543"<sip:05954543 at 192.168.4.250>
> From: "30"<sip:30 at 192.168.4.250>;tag=ca42eb7f
> Call-ID: OWNiODdiNTBiNTdiOWIxMDUxZDQwNzkwZTg4MjdjOGE.
> CSeq: 4 INVITE
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
> SUBSCRIBE, INFO
> Content-Type: application/sdp
> Proxy-Authorization: Digest
> username="9000",realm="asterisk",nonce="2fe5b437",uri="sip:05954543 at 192.168.4.250",response="38d11bf7db043a643500a74dfb29b933",algorithm=MD5
> Content-Length: 420
>
>
> Sending to 192.168.4.250 : 5060 (NAT)
> Using INVITE request as basis request -
> OWNiODdiNTBiNTdiOWIxMDUxZDQwNzkwZTg4MjdjOGE.
> Found user '30'
> <--- Reliably Transmitting (NAT) to 192.168.4.250:5060 --->
> SIP/2.0 403 Forbidden
> Via: SIP/2.0/UDP
> 192.168.4.250;branch=z9hG4bK58a2.12e355c3.0;received=192.168.4.250
> Via: SIP/2.0/UDP
> 192.168.20.66:26838;received=192.168.20.66;branch=z9hG4bK-d8754z-2f50b813447d8503-1---d8754z-;rport=26838
> From: "30"<sip:30 at 192.168.4.250>;tag=ca42eb7f
> To: "05954543"<sip:05954543 at 192.168.4.250>;tag=as388b4f7b
> Call-ID: OWNiODdiNTBiNTdiOWIxMDUxZDQwNzkwZTg4MjdjOGE.
> CSeq: 4 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 0
>
>
>
>
> Why do I get forbidden ? why in Proxy-Authorization: Digest username
> is still 9000 ?
>
> If I set user 30 on AsteriskG ( 192.168.20.66 ) works fine ....I use
> opensips 1.6.4
>
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