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

Vladimir Broz vladiksip at centrum.cz
Sat Aug 20 18:45:53 EDT 2011


great! you managed to get it working ;-) It may happen that sipwise guys 
will propose different, more sophisticated or elegant solution, but 
we'll see...

-Vlada B.

On 08/21/2011 12:14 AM, Skyler wrote:
> 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.
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
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