[Freeswitch-users] sip trunk question: why call through external profile is challenged?

Wed Jan 13 09:29:35 PST 2010

Look at how sipx sets up the users when they build the extensions and such for conferences, there was something special here, but I can't recall what.

Mike

On Jan 13, 2010, at 9:10 AM, Nikolay Kondratyev wrote:

> Hi all!
>  
> I’m brand new to FreeSwitch, but have some experience with SipX.
> Our company is evaluating FS.
> For test purposes I set up FS on a virtual machine (vmware esxi). I use CentOS.
> The FS version I use is 1.0.5-20100110-0400.
>  
> I have a question regarding sip trunk between FS and SipX.
> I created the following GW in external profile:
> [freeswitch at freeswitch external]$ cat sipx-lab.xml | grep -v '<!--'
> <include>
>   <gateway name="sipx4.lab.nstel.ru">
>   <param name="username" value="zxcv"/>
>   <param name="password" value="2007"/>
>   <param name="register" value="false"/>
>   </gateway>
> </include>
>  
> External.xml file is not modified after installation.
>  
> I see this gateway via fs_cli:
> freeswitch at internal> sofia status gateway sipx4.lab.nstel.ru
> =================================================================================================
> Name            sipx4.lab.nstel.ru
> Profile         external
> Scheme          Digest
> Realm           sipx4.lab.nstel.ru
> Username        zxcv
> Password        yes
> From            <sip:zxcv at sipx4.lab.nstel.ru;transport=udp>
> Contact         <sip:gw+sipx4.lab.nstel.ru at 172.23.22.49:5080;transport=udp;gw=sipx4.lab.nstel.ru>
> Exten           zxcv
> To              sip:zxcv at sipx4.lab.nstel.ru
> Proxy           sip:sipx4.lab.nstel.ru
> Context         public
> Expires         3600
> Freq            3600
> Ping            0
> PingFreq        0
> PingState       0/0/0
> State           NOREG
> Status          UP
> CallsIN         0
> CallsOUT        0
> =================================================================================================
>  
> I created new FS extension 2853. I registers (xlite) and I can call it.
>  
> Now I want to call FS user from sipx.
>  
> From the sipx side one can configure link to FS differently. There are two options:
> 1. Call FS directly trough sipxproxy (the core part of sipx works as sip proxy, not as b2bua)
> 2. Call trough embedded b2bua, named sipxbridge.
>  
> When a call is going trough sipxbridge, it is successfully landed at FS extension 2853.
> But when a call is going from the sipxproxy, it is challenged with Status 407 “proxy authentication required’, and then call fails.
> (I’m not sure if sipx should handle this challenge, but this is separate question for the sipx forum).
> At the default log level I see nothing in the freeswitch.log.
>  
> So the question is why one call through external profile is being challenged while the other is not?
> I suspect that the reason is in the difference in the two Invite messages:
>  
> Here is challenged Invite:
> INVITE sip:2853 at fs.lab.nstel.ru:5080;transport=udp SIP/2.0
> Record-Route: <sip:172.23.12.104:5060;lr;sipXecs-rs=%2Aauth%7E.%2Afrom%7EOTRiZjI3MjgtYWMxNzBjZTktMTNjNC0yZjZkZWMtNjNkNmUwN2UtMmY2ZGVj.900_ntap%2Aid%7EMjIyNjAtMQ%60%60%213ba6d85f6946c4c6001bee1d3b54474f>
> From: "testphone3857"<sip:3857 at lab.nstel.ru>;tag=94bf2728-ac170ce9-13c4-2f6dec-63d6e07e-2f6dec
> To: <sip:2853 at lab.nstel.ru>
> Call-Id: 94bed2a0-ac170ce9-13c4-2f6dec-68ffdb81-2f6dec at lab.nstel.ru
> Cseq: 2 INVITE
> Via: SIP/2.0/UDP 172.23.12.104;branch=z9hG4bK-sipXecs-000eaf48a36fe4029c7cde004a6f44424847
> Via: SIP/2.0/TCP 172.23.12.104;branch=z9hG4bK-sipXecs-000bf3008cfa3f3e70470789c75232ba9499~ac7fd729330fd563f83cacd941311e75;id=22260-1
> Via: SIP/2.0/UDP 172.23.12.233:5060;branch=z9hG4bK-2f6dec-b9456412-7ad07784
> Max-Forwards: 18
> Supported: replaces
> User-Agent: LG-Nortel LIP 6812 v1.2.38sp SN/00405A18B634
> Contact: <sip:3857 at 172.23.12.233:5060;x-sipX-nonat>
> Proxy-Authorization: Digest username="3857",realm="lab.nstel.ru",nonce="e2152722611af1bbe59f3a8eb31b8eb84b4db2ca",uri="sip:2853 at lab.nstel.ru",response="a09c34fbd897783c9b9af50bd044ccaa",algorithm=MD5
> Content-Type: application/sdp
> Content-Length: 301
> Date: Wed, 13 Jan 2010 11:47:22 GMT
> Expires: 60
> X-Sipx-Handled: X172.23.12.104-81.211.30.104
>  
> v=0
> o=LGEIPP 16246 16247 IN IP4 172.23.12.233
> s=SIP Call
> c=IN IP4 172.23.12.233
> t=0 0
> m=audio 23008 RTP/AVP 0 8 18 4 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=rtpmap:4 G723/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:18 annexb=no
> a=fmtp:101 0-11
> a=sendrecv
>  
>  
> And here is successful Invite:
> INVITE sip:2853 at fs.lab.nstel.ru;user=phone SIP/2.0
> Call-ID: 94bec3b8-ac170ce9-13c4-2f6d47-2fee09d0-2f6d47 at lab.nstel.ru.0
> CSeq: 1 INVITE
> From: "testphone3857" <sip:3857 at 172.23.12.104>;tag=816640414244159961
> To: <sip:2853 at fs.lab.nstel.ru;user=phone>
> Via: SIP/2.0/UDP 172.23.12.104:5080;branch=z9hG4bK885dde2c8680f5315845cd3350b8b605373534
> Max-Forwards: 70
> User-Agent: sipXecs/4.0.2 sipXecs/sipxbridge (Linux)
> P-Asserted-Identity: <sip:1004 at 172.23.12.104>
> Contact: <sip:1004 at 172.23.12.104:5080;transport=udp>
> Route: <sip:172.23.22.49:5080;transport=udp;lr>
> Session-Expires: 1800;refresher=uac
> Allow: INVITE,BYE,ACK,CANCEL,OPTIONS
> Content-Type: application/sdp
> Content-Length: 315
>  
> v=0
> o=sipxbridge 6640787141824452741 1 IN IP4 172.23.12.104
> s=SIP Call
> c=IN IP4 172.23.12.104
> t=0 0
> m=audio 30248 RTP/AVP 0 8 18 4 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=rtpmap:4 G723/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:18 annexb=no
> a=fmtp:101 0-11
> a=sendrecv
>  
> Can somebody please if it is a FS configuration problem or a software problem or is it a problem on the sipx side?
>  
> Thanks in advance,
> Nikolay.
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