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

Anthony Minessale anthony.minessale at gmail.com
Thu Jan 14 14:01:52 PST 2010 

try setting param accept-blind-auth to true in your sofia profile config
internal.xml
iirc it was made just for sipX who feels the need to send auth headers even
when nobody asked for them.
so even when auth-calls is false we will still try to parse the auth if one
is sent.


On Thu, Jan 14, 2010 at 11:21 AM, Michael Jerris <mike at jerris.com> wrote:

> if you look in the sample configs for the words blind and auth you will
> find all these settings, also you can setup acls for ip auth to not
> challenge.
>
> Mike
>
> On Jan 14, 2010, at 3:26 AM, Nikolay Kondratyev wrote:
>
>  Mike, thanks for the reply.
>
>
>
> Mmm… looks like I need more detailed instructions where to dig…
>
> Is there a way to turn off “challenging” completely?
>
> I thought that <param name="auth-calls" value="false"/> should do it, but
> alas…
>
> By the way should this parameter be visible in either “sofia status profile
> external” or “sofia status gateway sipx4.lab.nstel.ru” ? I don’t see it…
>
>
>
> I attached traces of failed and successful calls.
>
>
>
> Thanks and regards,
>
> Nikolay.
>   ------------------------------
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Jerris
> *Sent:* Wednesday, January 13, 2010 8:30 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] sip trunk question: why call
> throughexternal profile is challenged?
>
>
>
> 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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>
>   <call_from_sipx_via_unmanaged_gw.pcap>
> <call_from_sipx_via_siptrunk_sip-only.pcap>
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-- 
Anthony Minessale II

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