There is a user=phone issue within the 180's To header.  It is inside
the brackets for INVITE; it is outside for 180. 

> -----Original Message-----
> From: [email protected] 
> [mailto:[email protected]] On 
> Behalf Of Andrew Wood
> Sent: Monday, January 05, 2009 5:39 PM
> To: [email protected]
> Subject: [Sip-implementors] 180 being ignored by phone
> 
> Im trying to implement a simple forking proxy server.
> In the example below the calling phone (200) is on 
> 192.168.254.1 The called phone (201) is on 192.168.254.2 The 
> proxy is at 192.168.254.254
> 
> Following is the Invite received by the proxy from the 
> caller, and the 180 which the proxy sends back to the caller.
> The phone rings, and the 180 is modified and forwarded back 
> but  the caller doesnt get a ring tone.
> Ive checked them over and over and cant see whats wrong with them. 
> 
> Am I overlooking something here?
> 
> Thanks for any help
> 
> Andrew
> 
> 
> INVITE sip:[email protected];user=phone SIP/2.0
> Via: SIP/2.0/UDP 192.168.254.1:5060;branch=z9hG4bK1db88668c5d780ae
> Contact: <sip:[email protected]:5060;user=phone;transport=udp>
> To: <sip:[email protected];user=phone>
> From: <sip:[email protected];user=phone>;tag=4260414991
> Call-ID: [email protected]
> CSeq: 2 INVITE
> Expires: 300
> Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, 
> REGISTER, PRACK, UPDATE
> Content-Type: application/sdp
> Proxy-Authorization: Digest
> username="userA",realm="",nonce="1231192060:64976ad5abf3f2b1e8
> f9d72b7c7e71b9",uri="sip:[email protected]",response="b6e1d9
> faa7b9dbb897250f531c5cc092",qop=auth-int,nc=00000001,cnonce="f
> 9919a53" 
> 
> Supported: replaces, 100rel
> User-Agent: Cisco-CP7912/8.0.0-060111A
> Content-Length: 286
>  
> v=0
> o=200 10340144 10340144 IN IP4 192.168.254.1 s=Cisco 7912 SIP 
> Call c=IN IP4 192.168.254.1 t=0 0 m=audio 16384 RTP/AVP 0 18 
> 8 101 a=rtpmap:0 PCMU/8000/1
> a=rtpmap:18 G729/8000/1
> a=fmtp:18 annexb=yes
> a=rtpmap:8 PCMA/8000/1
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> 
> ==============================================================
> ===================
> 
> 
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP 192.168.254.1:5060;branch=z9hG4bK1db88668c5d780ae
> Contact: <sip:[email protected]:5060;user=phone;transport=udp>
> To: <sip:[email protected]:5060>;user=phone;tag=3440312132
> From: <sip:[email protected];user=phone>;tag=4260414991
> Call-ID: [email protected]
> CSeq: 2 INVITE
> Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, 
> REGISTER, PRACK, UPDATE
> Server: Cisco-CP7912/8.0.0-060111A
> Content-Length: 0
> 
> _______________________________________________
> Sip-implementors mailing list
> [email protected]
> https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
> 

_______________________________________________
Sip-implementors mailing list
[email protected]
https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors

Reply via email to