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

Sat Aug 20 15:42:58 EDT 2011

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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