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

Sat Aug 20 16:03:11 EDT 2011

Hi Vlad,

 Funny, I closed my email client while working and went the exact path
you were going.

 indeed $avp(s:caller_cli_userprov) = $fn; is a part of the problem.
Right now there are errors with from header as "Testy Testerson" (in
quotes) is invalid.

 just now I am looking into how to remove the ""

S.

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.
> >>>>
> >>>>
> >>>>
> >>>>
> >>>
> >>>
> >>> _______________________________________________
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> >>> Spce-user at lists.sipwise.com
> >>> http://lists.sipwise.com/listinfo/spce-user
> >>
> >> _______________________________________________
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> >
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