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

Sat Aug 20 15:52:49 EDT 2011


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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>>
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