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

Sat Aug 20 14:40:53 EDT 2011

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 ...

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.
> 
> 
> 
> 