[Spce-user] spce caller name peered with Asterisk/FreeSwitch

Sat Aug 20 16:35:07 EDT 2011

 I think the name will have to be extracted from $fu in some way and
placed into its own pvar. Then passed along with $fU where required
because other parts of the script needs $fU like P-Asserted-Identity.

just guessing here, will continue to play with this more.

U 2011/08/20 16:21:00.479790 127.0.0.1:5062 -> 127.0.0.1:5080
INVITE sip:b2b-16048881212 at DEVICE_IP:5060 SIP/2.0.
Record-Route:
<sip:127.0.0.1:5062;lr=on;ftag=as3718f22a;did=365.85f57a43;vsf=SlZ3eVwQEC81NzMXCB1iHHQgYggBETZ3RGtGflp3dhhDDUA2OA-->.
Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=as3718f22a>.
Record-Route: <sip:SPCE_IP;r2=on;lr=on;ftag=as3718f22a>.
Via: SIP/2.0/UDP
127.0.0.1:5062;branch=z9hG4bK334b.955d874cef855147ed506b3bbca0001f.0.
Route: <sip:lb at 127.0.0.1:5060;lr>.
Via: SIP/2.0/UDP 127.0.0.1;rport=5060;branch=z9hG4bK334b.648f3247.0.
v: SIP/2.0/UDP ASTERISK_IP:5060;branch=z9hG4bK50a97a7b;rport=5060.
Max-Forwards: 68.
f: ""Testy Testerson"" <sip:"Testy
Testerson"@ASTERISK_IP>;tag=as3718f22a.
t: <sip:16048881212 at SPCE_IP>.
m: <sip:2274581515 at ASTERISK_IP>.
i: 106aceff672c108717cb791207b6feb1 at ASTERISK_IP.
CSeq: 102 INVITE.
User-Agent: ASTERISK_UPSTREAM.
Date: Sat, 20 Aug 2011 20:21:00 GMT.
x: 600.
Min-SE: 90.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO.
k: replaces, timer.
c: application/sdp.
l: 544.
P-Asserted-Identity: <sip:"Testy Testerson"@ASTERISK_IP>.
P-R-Uri: sip:16048881212 at DEVICE_IP:5060.
P-D-Uri: sip:lb at 127.0.0.1:5060;lr

On Sat, 2011-08-20 at 13:13 -0700, Skyler wrote:
> Still looking to remove "" from the $fn pseudo var. I think this is good
> start, next there may be extra <> also but one thing at a time I
> guess :)
> 
> 
> INFO: <script>: Setting From to 'sip:"Testy Testerson"@ASTERISK_IP' -
> M=INVITE 
> <script>: NAT-Reply - S=400 - Bad From header F=<null>
> T=sip:16048881212 at SPCE_IP
> ERROR: pv [pv_core.c:418]: cannot parse From header
> ERROR: <core> [parser/parse_from.c:79]: ERROR:parse_from_header: bad
> from header
> ERROR: <core> [parser/parse_to.c:737]: ERROR: parse_to : unexpected char
> ["] in status 6: <<"Testy Testerson" <sip:>> .
> 
> 
> On Sat, 2011-08-20 at 21:52 +0200, Vladimir Broz wrote:
> > 
> > On 08/20/2011 09:51 PM, Vladimir Broz wrote:
> > > especially if it is in proxy config, then it could be somewhere here:
> > >
> > > ...
> > > if($fn != $null && avp_check("$fn", "re/^\"?\+?[0-9]+\"?$"))
> > > {
> > > $avp(s:caller_cli_userprov) = $fn;
> > > avp_subst("$avp(s:caller_cli_userprov)", "/^\"?([^\"]*)\"?$/\1/");
> > > }
> > > else
> > > {
> > > $avp(s:caller_cli_userprov) = $fU;
> > > }
> > > ....
> > >
> > > I guess (really my guess!!!), that the last line could be set to "$fn"...
> > >
> > > else
> > > {
> > > $avp(s:caller_cli_userprov) = $fU;
> > $avp(s:caller_cli_userprov) = $fn;
> > > }
> > indeed...
> > 
> > -Vlada B.
> > >
> > > I'm really not sure! and don't know SPCE in details, it may cause some
> > > other problems...
> > >
> > > Sorry if I'm wrong!
> > >
> > > Regards,
> > > -Vlada B.
> > >
> > > On 08/20/2011 09:42 PM, Vladimir Broz wrote:
> > >> Hi,
> > >>
> > >> On 08/20/2011 08:40 PM, Skyler wrote:
> > >>> Hi,
> > >>>
> > >>> My appologies. I missed the INVITE from spce proxy to sems. The missing
> > >>> INVITE is below, this shows that the From HF is replace by proxy prior
> > >>> to sending to sems. Now looking into proxy config ...
> > >>
> > >> try to find "$fn" which is the reference to display name in From
> > >> header...
> > >>
> > >> -Vlada B.
> > >>>
> > >>>
> > >>> U 2011/08/20 14:20:31.790504 127.0.0.1:5062 -> 127.0.0.1:5080
> > >>> INVITE sip:b2b-16048881212 at DEVICE_IP:5060 SIP/2.0.
> > >>> Record-Route:
> > >>> <sip:127.0.0.1:5062;lr=on;ftag=as6c6ae522;did=982.3dde1881;vsf=SlZ3eUh2S29GYmhXTExOWVUzZk1KVnd5SHZLb0ZiaFdMTE4->.
> > >>>
> > >>>
> > >>> Record-Route:<sip:127.0.0.1;r2=on;lr=on;ftag=as6c6ae522>.
> > >>> Record-Route:<sip:SPCE_IP;r2=on;lr=on;ftag=as6c6ae522>.
> > >>> Via: SIP/2.0/UDP
> > >>> 127.0.0.1:5062;branch=z9hG4bK8513.86716cf040064692b88f0029bb3fc8c7.0.
> > >>> Route:<sip:lb at 127.0.0.1:5060;lr>.
> > >>> Via: SIP/2.0/UDP 127.0.0.1;rport=5060;branch=z9hG4bK8513.9dd8fe07.0.
> > >>> v: SIP/2.0/UDP ASTERISK_IP:5060;branch=z9hG4bK3bb6032e;rport=5060.
> > >>> Max-Forwards: 68.
> > >>> f: "2725846431"<sip:2725846431 at ASTERISK_IP>;tag=as6c6ae522.
> > >>> t:<sip:16048881212 at SPCE_IP>.
> > >>> m:<sip:2725846431 at ASTERISK_IP>.
> > >>> i: 7fc3bcbf5bf0e6ef19907a8d5dc0e179 at ASTERISK_IP.
> > >>> CSeq: 102 INVITE.
> > >>> User-Agent: voxcentral.
> > >>> Date: Sat, 20 Aug 2011 18:20:31 GMT.
> > >>> x: 600.
> > >>> Min-SE: 90.
> > >>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
> > >>> INFO.
> > >>> k: replaces, timer.
> > >>> c: application/sdp.
> > >>> l: 544.
> > >>> P-Asserted-Identity:<sip:2725846431 at ASTERISK_IP>.
> > >>> P-R-Uri: sip:16048881212 at DEVICE_IP:5060.
> > >>> P-D-Uri: sip:lb at 127.0.0.1:5060;lr
> > >>> .
> > >>>
> > >>> On Sat, 2011-08-20 at 01:51 -0700, Skyler wrote:
> > >>>> Hi,
> > >>>>
> > >>>>>
> > >>>>> I see. I'll check it out in our dev systems.
> > >>>>>
> > >>>>
> > >>>> I can confirm that this is indeed an issue on the SPCE side of things
> > >>>> (trace below). Notice how the caller name in from field is replaced in
> > >>>> reply from SEMS? This is related to B2BUA functionality I think.
> > >>>>
> > >>>> Hope this helps out. I don't know anything about SEMS but wanting to
> > >>>> fix
> > >>>> this also so will be looking into it more over the weekend here.
> > >>>>
> > >>>> Cheers,
> > >>>>
> > >>>> S.
> > >>>>
> > >>>>
> > >>>> U 2011/08/20 04:31:02.882004 ASTERISK_IP:5060 -> SPCE_IP:5060
> > >>>> INVITE sip:16048881212 at SPCE_IP SIP/2.0.
> > >>>> v: SIP/2.0/UDP ASTERISK_IP:5060;branch=z9hG4bK78bc993b;rport.
> > >>>> Max-Forwards: 70.
> > >>>> f: "Testy Testerson"<sip:2725846431 at ASTERISK_IP>;tag=as17b832a2.
> > >>>> t:<sip:16048881212 at SPCE_IP>.
> > >>>> m:<sip:2725846431 at ASTERISK_IP>.
> > >>>> i: 470166fd32c0a2e840d226450f2ebd6c at ASTERISK_IP.
> > >>>> CSeq: 102 INVITE.
> > >>>> User-Agent: Asterisk (1.6.2).
> > >>>> Date: Sat, 20 Aug 2011 08:31:02 GMT.
> > >>>> x: 600.
> > >>>> Min-SE: 90.
> > >>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
> > >>>> INFO.
> > >>>> k: replaces, timer.
> > >>>> c: application/sdp.
> > >>>> l: 524.
> > >>>> .
> > >>>> v=0.
> > >>>> o=root 1615613317 1615613317 IN IP4 ASTERISK_IP.
> > >>>> s=Asterisk (1.6.2).
> > >>>> c=IN IP4 ASTERISK_IP.
> > >>>> t=0 0.
> > >>>> m=audio 15224 RTP/AVP 0 4 3 8 112 5 10 7 18 111 9 101.
> > >>>> a=rtpmap:0 PCMU/8000.
> > >>>> a=rtpmap:4 G723/8000.
> > >>>> a=fmtp:4 annexa=no.
> > >>>> a=rtpmap:3 GSM/8000.
> > >>>> a=rtpmap:8 PCMA/8000.
> > >>>> a=rtpmap:112 G726-32/8000.
> > >>>> a=rtpmap:5 DVI4/8000.
> > >>>> a=rtpmap:10 L16/8000.
> > >>>> a=rtpmap:7 LPC/8000.
> > >>>> a=rtpmap:18 G729/8000.
> > >>>> a=fmtp:18 annexb=no.
> > >>>> a=rtpmap:111 G726-32/8000.
> > >>>> a=rtpmap:9 G722/8000.
> > >>>> a=rtpmap:101 telephone-event/8000.
> > >>>> a=fmtp:101 0-16.
> > >>>> a=ptime:20.
> > >>>> a=sendrecv.
> > >>>>
> > >>>> U 2011/08/20 04:31:02.882371 SPCE_IP:5060 -> ASTERISK_IP:5060
> > >>>> SIP/2.0 100 Trying.
> > >>>> v: SIP/2.0/UDP ASTERISK_IP:5060;branch=z9hG4bK78bc993b;rport=5060.
> > >>>> f: "Testy Testerson"<sip:2725846431 at ASTERISK_IP>;tag=as17b832a2.
> > >>>> t:<sip:16048881212 at SPCE_IP>.
> > >>>> i: 470166fd32c0a2e840d226450f2ebd6c at ASTERISK_IP.
> > >>>> CSeq: 102 INVITE.
> > >>>> Server: kamailio (3.1.3 (x86_64/linux)).
> > >>>> Content-Length: 0.
> > >>>> .
> > >>>>
> > >>>> U 2011/08/20 04:31:02.882689 127.0.0.1:5060 -> 127.0.0.1:5062
> > >>>> INVITE sip:16048881212 at SPCE_IP SIP/2.0.
> > >>>> Record-Route:<sip:127.0.0.1;r2=on;lr=on;ftag=as17b832a2>.
> > >>>> Record-Route:<sip:SPCE_IP;r2=on;lr=on;ftag=as17b832a2>.
> > >>>> Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bK9658.0d9f2ae1.0.
> > >>>> v: SIP/2.0/UDP ASTERISK_IP:5060;branch=z9hG4bK78bc993b;rport=5060.
> > >>>> Max-Forwards: 69.
> > >>>> f: "Testy Testerson"<sip:2725846431 at ASTERISK_IP>;tag=as17b832a2.
> > >>>> t:<sip:16048881212 at SPCE_IP>.
> > >>>> m:<sip:2725846431 at ASTERISK_IP>.
> > >>>> i: 470166fd32c0a2e840d226450f2ebd6c at ASTERISK_IP.
> > >>>> CSeq: 102 INVITE.
> > >>>> User-Agent: Asterisk (1.6.2).
> > >>>> Date: Sat, 20 Aug 2011 08:31:02 GMT.
> > >>>> x: 600.
> > >>>> Min-SE: 90.
> > >>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
> > >>>> INFO.
> > >>>> k: replaces, timer.
> > >>>> c: application/sdp.
> > >>>> l: 524.
> > >>>> P-NGCP-Src-Ip: ASTERISK_IP.
> > >>>> P-NGCP-Src-Port: 5060.
> > >>>> P-NGCP-Src-Proto: udp.
> > >>>> .
> > >>>> v=0.
> > >>>> o=root 1615613317 1615613317 IN IP4 ASTERISK_IP.
> > >>>> s=Asterisk (1.6.2).
> > >>>> c=IN IP4 ASTERISK_IP.
> > >>>> t=0 0.
> > >>>> m=audio 15224 RTP/AVP 0 4 3 8 112 5 10 7 18 111 9 101.
> > >>>> a=rtpmap:0 PCMU/8000.
> > >>>> a=rtpmap:4 G723/8000.
> > >>>> a=fmtp:4 annexa=no.
> > >>>> a=rtpmap:3 GSM/8000.
> > >>>> a=rtpmap:8 PCMA/8000.
> > >>>> a=rtpmap:112 G726-32/8000.
> > >>>> a=rtpmap:5 DVI4/8000.
> > >>>> a=rtpmap:10 L16/8000.
> > >>>> a=rtpmap:7 LPC/8000.
> > >>>> a=rtpmap:18 G729/8000.
> > >>>> a=fmtp:18 annexb=no.
> > >>>> a=rtpmap:111 G726-32/8000.
> > >>>> a=rtpmap:9 G722/8000.
> > >>>> a=rtpmap:101 telephone-event/8000.
> > >>>> a=fmtp:101 0-16.
> > >>>> a=ptime:20.
> > >>>> a=sendrecv.
> > >>>>
> > >>>> U 2011/08/20 04:31:02.882854 127.0.0.1:5062 -> 127.0.0.1:5060
> > >>>> SIP/2.0 100 Trying.
> > >>>> Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bK9658.0d9f2ae1.0;rport=5060.
> > >>>> v: SIP/2.0/UDP ASTERISK_IP:5060;branch=z9hG4bK78bc993b;rport=5060.
> > >>>> f: "Testy Testerson"<sip:2725846431 at ASTERISK_IP>;tag=as17b832a2.
> > >>>> t:<sip:16048881212 at SPCE_IP>.
> > >>>> i: 470166fd32c0a2e840d226450f2ebd6c at ASTERISK_IP.
> > >>>> CSeq: 102 INVITE.
> > >>>> Server: kamailio (3.1.3 (x86_64/linux)).
> > >>>> Content-Length: 0.
> > >>>> .
> > >>>>
> > >>>> U 2011/08/20 04:31:02.956841 127.0.0.1:5080 -> 127.0.0.1:5060
> > >>>> INVITE sip:b2b-16048881212 at 207.216.253.26:5060 SIP/2.0.
> > >>>> Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bKQJ~PjayQ;rport.
> > >>>> From: "2725846431"
> > >>>> <sip:2725846431 at ASTERISK_IP>;tag=24AD3534-4E4F70C6000E9916-5C705700.
> > >>>> To:<sip:16048881212 at SPCE_IP>.
> > >>>> CSeq: 10 INVITE.
> > >>>> Call-ID: 470166fd32c0a2e840d226450f2ebd6c at ASTERISK_IP_b2b-1.
> > >>>> Contact:<sip:127.0.0.1:5080>.
> > >>>> Route:<sip:lb at 127.0.0.1:5060;lr;lr>.
> > >>>> P-Asserted-Identity:<sip:2725846431 at ASTERISK_IP>.
> > >>>> P-R-Uri: sip:16048881212 at 207.216.253.26:5060.
> > >>>> P-D-Uri: sip:lb at 127.0.0.1:5060;lr.
> > >>>> Supported: timer.
> > >>>> Session-Expires: 300.
> > >>>> Min-SE: 90.
> > >>>> Content-Type: application/sdp.
> > >>>> Content-Length: 544.
> > >>>> v=0.
> > >>>> o=root 1615613317 1615613317 IN IP4 SPCE_IP.
> > >>>> s=Asterisk (1.6.2).
> > >>>> c=IN IP4 SPCE_IP.
> > >>>> t=0 0.
> > >>>> m=audio 30076 RTP/AVP 0 4 3 8 112 5 10 7 18 111 9 101.
> > >>>> a=rtpmap:0 PCMU/8000.
> > >>>> a=rtpmap:4 G723/8000.
> > >>>> a=fmtp:4 annexa=no.
> > >>>> a=rtpmap:3 GSM/8000.
> > >>>> a=rtpmap:8 PCMA/8000.
> > >>>> a=rtpmap:112 G726-32/8000.
> > >>>> a=rtpmap:5 DVI4/8000.
> > >>>> a=rtpmap:10 L16/8000.
> > >>>> a=rtpmap:7 LPC/8000.
> > >>>> a=rtpmap:18 G729/8000.
> > >>>> a=fmtp:18 annexb=no.
> > >>>> a=rtpmap:111 G726-32/8000.
> > >>>> a=rtpmap:9 G722/8000.
> > >>>> a=rtpmap:101 telephone-event/8000.
> > >>>> a=fmtp:101 0-16.
> > >>>> a=ptime:20.
> > >>>> a=sendrecv.
> > >>>> a=nortpproxy:yes.
> > >>>>
> > >>>>
> > >>>>
> > >>>>
> > >>>
> > >>>
> > >>> _______________________________________________
> > >>> Spce-user mailing list
> > >>> Spce-user at lists.sipwise.com
> > >>> http://lists.sipwise.com/listinfo/spce-user
> > >>
> > >> _______________________________________________
> > >> Spce-user mailing list
> > >> Spce-user at lists.sipwise.com
> > >> http://lists.sipwise.com/listinfo/spce-user
> > >
> > > _______________________________________________
> > > Spce-user mailing list
> > > Spce-user at lists.sipwise.com
> > > http://lists.sipwise.com/listinfo/spce-user