[Spce-user] spce caller name peered with Asterisk/FreeSwitch - [FOR REVIEW/TESTING]
Skyler
skchopperguy at gmail.com
Sat Aug 20 18:14:12 EDT 2011
Thanks Vlad !
My thinking was too complicated, I needed to start from beginning and
follow the flow. The only change that was needed is to comment out the
uac_replace_from on line 2146 of kamailio.proxy.cfg
Not sure if that affects anything else in the script so *caution*
Definitely in the right direction though, caller ID with name + number
is now working as expected. I'll be running more tests on
inbound/outbound, local and external to see if this maybe 'breaks'
something else.
New ngrep showing that it works...
U 2011/08/20 17:54:08.970601 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=as487a2f1f;did=96a.b272e736>.
Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=as487a2f1f>.
Record-Route: <sip:SPCE_IP;r2=on;lr=on;ftag=as487a2f1f>.
Via: SIP/2.0/UDP
127.0.0.1:5062;branch=z9hG4bKa39d.7c0b42a2c96b928e04764d8ad01f31eb.0.
Route: <sip:lb at 127.0.0.1:5060;lr>.
Via: SIP/2.0/UDP 127.0.0.1;rport=5060;branch=z9hG4bKa39d.13573013.0.
v: SIP/2.0/UDP ASTERISK_IP:5060;branch=z9hG4bK582f5c8a;rport=5060.
Max-Forwards: 68.
f: "Testy Testerson" <sip:2725846431 at ASTERISK_IP>;tag=as487a2f1f.
t: <sip:16048881212 at SPCE_IP>.
m: <sip:2725846431 at ASTERISK_IP>.
i: 32bc91cb58535e116bdeac620d55a97f at ASTERISK_IP.
CSeq: 102 INVITE.
User-Agent: Asterisk (1.6.2).
Date: Sat, 20 Aug 2011 21:54:08 GMT.
x: 600.
Min-SE: 90.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO.
k: replaces, timer.
c: application/sdp.
l: 542.
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
.
v=0.
o=root 580118314 580118314 IN IP4 SPCE_IP.
s=Asterisk (1.6.2).
c=IN IP4 SPCE_IP.
t=0 0.
m=audio 30136 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.
U 2011/08/20 17:54:08.971051 127.0.0.1:5080 -> 127.0.0.1:5062
SIP/2.0 100 Connecting.
Record-Route:
<sip:127.0.0.1:5062;lr=on;ftag=as487a2f1f;did=96a.b272e736>.
Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=as487a2f1f>.
Record-Route: <sip:SPCE_IP;r2=on;lr=on;ftag=as487a2f1f>.
Via: SIP/2.0/UDP
127.0.0.1:5062;branch=z9hG4bKa39d.7c0b42a2c96b928e04764d8ad01f31eb.0;received=127.0.0.1.
Via: SIP/2.0/UDP 127.0.0.1;rport=5060;branch=z9hG4bKa39d.13573013.0.
v: SIP/2.0/UDP ASTERISK_IP:5060;branch=z9hG4bK582f5c8a;rport=5060.
f: "Testy Testerson" <sip:2725846431 at ASTERISK_IP>;tag=as487a2f1f.
t: <sip:16048881212 at SPCE_IP>;tag=2CBF3F78-4E502D00000ED055-5CB09700.
i: 32bc91cb58535e116bdeac620d55a97f at ASTERISK_IP.
CSeq: 102 INVITE.
Server: Sipwise NGCP Application Server.
Contact: <sip:b2b-16048881212 at 127.0.0.1:5080>.
Content-Length: 0.
.
On Sat, 2011-08-20 at 22:45 +0200, Vladimir Broz wrote:
> I do not think that's correct way. You are trying to replace the URI,
> not the display name.
>
> from the log you posted before:
> INFO:<script>: Setting From to 'sip:"Testy Testerson"@ASTERISK_IP' -
> >> M=INVITE
>
> ...
> {
> xlog("L_INFO", "Setting From to '$var(caller_cli_uri)' - M=$rm R=$ru
> F=$fu T=$tu IP=$avp(s:ip):$avp(s:port) ($si:$sp) ID=$ci\n");
> uac_replace_from("$var(caller_cli)", "$var(caller_cli_uri)");
> }
>
> the uac_replace_from function should do:
>
> # replace both display and uri
> uac_replace_from("$avp(s:display)","$avp(s:uri)");
> # replace only display and do not touch uri
> uac_replace_from("batman","");
> # remove display and replace uri
> uac_replace_from("","sip:robin at gotham.org");
> # remove display and do not touch uri
> uac_replace_from("","");
>
> so I guess you should look at the first parameter "$var(caller_cli)".
> But it is only my guess, I'm not familiar with kamailio config into
> details, sorry.
>
> My 2 cents,
> -Vlada B
>
> On 08/20/2011 10:35 PM, Skyler wrote:
> >
> > 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
> >
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