[asterisk-users] problem with setting outbound caller id when calling another asterisk

[Chris Mazuc]

Asterisk seems to have a bug which is not letting me set the caller id 
to another peer's caller id.

http://www.mail-archive.com/asterisk-dev@lists.digium.com/msg23230.html

I've sent this to the asterisk-users mailing list, hopefully I get a 
response soon if there is a workaround.

I'm going to see if there is a way to blindly accept calls from a known 
IP address, but I don't think there is a way that would retain CDR 
information.

Chris Mazuc wrote:
> I have an asterisk box at a remote location (which I will call remote), 
> which registers to my local asterisk box (I'll call that one local), and 
> uses that to route calls to the outside world. The problem I am having 
> is that the remote location needs to lie about it's callerid sometimes, 
> however if I set a callerid that matches the extension of another peer 
> that exists, "local" returns a 403 to "remote". I can set the callerid 
> to the did and it will work fine, or I can set the callerid to something 
> random and it will work fine.
> 
> What does * do with the proxy-authorization header, because it seems to 
> be ignoring the username part.
> 
> Any help is greatly appreciated.
> 
> Thanks,
> Chris Mazuc
> 
> <-- SIP read from REMOTE:1025:
> INVITE sip:1XXXXXX7257 at LOCAL SIP/2.0
> Via: SIP/2.0/UDP REMOTE:5060;branch=z9hG4bK1757eacd;rport
> From: "My Name" <sip:1XXXXXX1200 at REMOTE>;tag=as4f42dab4
> To: <sip:1XXXXXX7257 at LOCAL>
> Contact: <sip:1XXXXXX1200 at REMOTE>
> Call-ID: 571c518a0257e17916e6e27b4e4b9fed at REMOTE
> CSeq: 103 INVITE
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Proxy-Authorization: Digest username="1XXXXXX1205", realm="asterisk", 
> algorithm=MD5, uri="sip:1XXXXXX7257 at LOCAL", nonce="45a347bc", 
> response="934b409f19a0ebf28d1cf266db29f497", opaque=""
> Date: Tue, 24 Oct 2006 20:26:28 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Content-Type: application/sdp
> Content-Length: 240
> 
> v=0
> o=root 2238 2239 IN IP4 REMOTE
> s=session
> c=IN IP4 REMOTE
> t=0 0
> m=audio 15384 RTP/AVP 0 8 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> 
> --- (14 headers 11 lines)---
> Using INVITE request as basis request - 
> 571c518a0257e17916e6e27b4e4b9fed at REMOTE
> Sending to REMOTE : 5060 (NAT)
> Found user '1XXXXXX1200'
> Reliably Transmitting (NAT) to REMOTE:1025:
> SIP/2.0 403 Forbidden
> Via: SIP/2.0/UDP 
> REMOTE:5060;branch=z9hG4bK1757eacd;received=REMOTE;rport=1025
> From: "My Name" <sip:1XXXXXX1200 at REMOTE>;tag=as4f42dab4
> To: <sip:1XXXXXX7257 at LOCAL>;tag=as1f40e0ec
> Call-ID: 571c518a0257e17916e6e27b4e4b9fed at REMOTE
> CSeq: 103 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Contact: <sip:7257 at LOCAL>
> Content-Length: 0
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