Chris Mazuc
2006-Oct-24 13:41 UTC
[asterisk-users] problem with setting outbound caller id when calling another asterisk
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... or maybe I need to go read some RFCs. Any help is greatly appreciated. Thanks, Chris Mazuc <-- SIP read from REMOTE:1025: INVITE sip:1XXXXXX7257@LOCAL SIP/2.0 Via: SIP/2.0/UDP REMOTE:5060;branch=z9hG4bK1757eacd;rport From: "My Name" <sip:1XXXXXX1200@REMOTE>;tag=as4f42dab4 To: <sip:1XXXXXX7257@LOCAL> Contact: <sip:1XXXXXX1200@REMOTE> Call-ID: 571c518a0257e17916e6e27b4e4b9fed@REMOTE CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Proxy-Authorization: Digest username="1XXXXXX1205", realm="asterisk", algorithm=MD5, uri="sip:1XXXXXX7257@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@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@REMOTE>;tag=as4f42dab4 To: <sip:1XXXXXX7257@LOCAL>;tag=as1f40e0ec Call-ID: 571c518a0257e17916e6e27b4e4b9fed@REMOTE CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:7257@LOCAL> Content-Length: 0
Chris Mazuc
2006-Oct-25 10:34 UTC
[asterisk-users] problem with setting outbound caller id when calling another asterisk
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@LOCAL SIP/2.0 > Via: SIP/2.0/UDP REMOTE:5060;branch=z9hG4bK1757eacd;rport > From: "My Name" <sip:1XXXXXX1200@REMOTE>;tag=as4f42dab4 > To: <sip:1XXXXXX7257@LOCAL> > Contact: <sip:1XXXXXX1200@REMOTE> > Call-ID: 571c518a0257e17916e6e27b4e4b9fed@REMOTE > CSeq: 103 INVITE > User-Agent: Asterisk PBX > Max-Forwards: 70 > Proxy-Authorization: Digest username="1XXXXXX1205", realm="asterisk", > algorithm=MD5, uri="sip:1XXXXXX7257@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@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@REMOTE>;tag=as4f42dab4 > To: <sip:1XXXXXX7257@LOCAL>;tag=as1f40e0ec > Call-ID: 571c518a0257e17916e6e27b4e4b9fed@REMOTE > CSeq: 103 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Contact: <sip:7257@LOCAL> > Content-Length: 0 > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >