It looks good but in the capture file I saw
FQNDIP in RR and not FQNDDNS
This post by Henning may help you:
This is a response from my Kamailio to Teams. Maybe it can be useful for
you:
tag: snd
pid: 1394
process: 1
time: 1599126436.582012
date: Thu Sep 3 11:47:16 2020
proto: tls ipv4
srcip: SBC-IP-ADDR
srcport: 5061
dstip: 52.114.75.24
dstport: 5061
~~~~~~~~~~~~~~~~~~~~
SIP/2.0 200 OK
Via: SIP/2.0/TLS 52.114.75.24:5061;branch=z9hG4bK8ac0b4eb
Record-Route: <sip:SBC-DNS-DOMAIN:5060;r2=on;lr>
Record-Route: <sip:SBC-DNS-DOMAIN:5061;transport=tls;r2=on;lr>
Record-Route: <sip:sip-du-a-eu.pstnhub.microsoft.com:5061
;transport=tls;lr>
From: Pepelux <sip:+34XXXXXXXXX@sip.pstnhub.microsoft.com:5061
;user=phone>;tag=3a6ca98c0a9a46c98ad781c82f389c4d
To: <sip:+34YYYYYYYYY@SBC-DNS-DOMAIN:5061;user=phone>;tag=as524dd8d6
Call-ID: 99ac64b5ad3455be9fd8c838cbdd4c6c
CSeq: 1 INVITE
Server: Asterisk PBX 16.2.1~dfsg-1+deb10u1
Allow:sINVITE, ACK, CANCEL, OPTIONS, BYE, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces
Contact: <sip:+34YYYYYYYYY@SBC-IP-ADDR:5080>
Content-Type: application/sdp
Content-Length: 532
v=0
o=root 11212956 11212956 IN IP4 SBC-IP-ADDR
s=Asterisk PBX 16.2.1~dfsg-1+deb10u1
c=IN IP4 SBC-IP-ADDR
t=0 0
m=audio 30444 RTP/SAVP 8
a=maxptime:150
a=mid:1
a=rtpmap:8 PCMA/8000
a=sendrecv
a=rtcp:30445
a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:13aivX1dzg2Pbon6bNcvyzNenxMbtvCEh65mnL0t
a=ptime:20
a=ice-ufrag:oysP7oty
a=ice-pwd:Vsv8LeF21onN8NFIKhOvpF53zL
a=candidate:MLkUNexgYqowLxtY 1 UDP 2130706431 SBC-IP-ADDR 30444 typ host
a=candidate:MLkUNexgYqowLxtY 2 UDP 2130706430 SBC-IP-ADDR 30445 typ host
~~~~~~~~~~~~~~~~~~~~
tag: rcv
pid: 1412
process: 19
time: 1599126436.612972
date: Thu Sep 3 11:47:16 2020
proto: tls ipv4
srcip: 52.114.75.24
srcport: 6209
dstip: SBC-IP-ADDR
dstport: 5061
~~~~~~~~~~~~~~~~~~~~
ACK sip:+34YYYYYYYYY@SBC-IP-ADDR:5080 SIP/2.0
FROM: Pepelux <sip:+34XXXXXXXXX@sip.pstnhub.microsoft.com:5061
;user=phone>;tag=3a6ca98c0a9a46c98ad781c82f389c4d
TO: <sip:+34YYYYYYYYY@SBC-DNS-DOMAIN:5061>;user=phone;tag=as524dd8d6
CSEQ: 1 ACK
CALL-ID: 99ac64b5ad3455be9fd8c838cbdd4c6c
MAX-FORWARDS: 70
VIA: SIP/2.0/TLS 52.114.75.24:5061;branch=z9hG4bK693bb042
ROUTE:
<sip:SBC-DNS-DOMAIN:5061;transport=tls;r2=on;lr>,<sip:SBC-DNS-DOMAIN:5060;r2=on;lr>
CONTACT: <sip:api-du-c-euwe.pstnhub.microsoft.com:443
;x-i=21167d9c-8cf9-4253-b059-2d987fadae5a;x-c=99ac64b5ad3455be9fd8c838cbdd4c6c/d/8/90df5d0ec92048de868e268fa57ac0f1>
CONTENT-LENGTH: 0
USER-AGENT: Microsoft.PSTNHub.SIPProxy v.2020.9.1.3 i.EUWE.7
ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY
Regards
On Thu, 3 Sep 2020 at 12:34, sip user <sipuser404(a)gmail.com> wrote:
Hi Pepelux,
I have this one:
remove_hf("Route");
if (is_method("INVITE|SUBSCRIBE")) {
if($src_ip != "IP ASTERISK"){
record_route();
xlog("L_INFO", "***********ROUTE
PSTN***********");
$rU="1005";
} else {
xlog("L_INFO","LLamada desde $si con puerto
$sp");
record_route_preset("FQNDDNS:5061;transport=tls", "FQNDIP:5060");
add_rr_param(";r2=on");
route(DISPATCH);
route(RELAY);
}
}
When the call is from Teams (src_ip != "IP ASTERISK"), incoming calls,
I send the call to 1005 extension. Is here where I have to make the change?
Or where?
Thanks
El jue., 3 sept. 2020 a las 12:14, Pepelux (<pepeluxx(a)gmail.com>)
escribió:
> Hi
>
> Kamailio doesn't receive any ACK from Teams. I think the problem is
> the '200 Ok' that you send to Teams is not what he expected. Maybe this is
> wrong:
> Record-Route: <sip:FQNDIP;r2=on;lr>
> Record-Route: <sip:FQNDIP:5061;transport=tls;r2=on;lr>
>
> Try to put the registered domain (FQNDDNS) and not de IP address
>
> Regards
>
>
>
> On Thu, 3 Sep 2020 at 10:56, sip user <sipuser404(a)gmail.com> wrote:
>
>> Sorry.. Yes, I need to load sipdump.so module..
>>
>> I attach the result..
>>
>> Thanks
>>
>> El mar., 1 sept. 2020 a las 14:03, Pepelux (<pepeluxx(a)gmail.com>)
>> escribió:
>>
>>> Hi
>>>
>>> Have you loaded the module?
>>>
>>> loadmodule "sipdump.so"
>>>
>>> On Tue, 1 Sep 2020 at 13:56, sip user <sipuser404(a)gmail.com> wrote:
>>>
>>>> Hi pepelux.. When I set:
>>>>
>>>> modparam("sipdump", "enable", 1)
>>>>
>>>>
>>>> Error, Kamailio not start, error bad config..
>>>>
>>>> Thanks
>>>>
>>>> El mar., 1 sept. 2020 a las 13:45, Pepelux (<pepeluxx(a)gmail.com>)
>>>> escribió:
>>>>
>>>>> Sorry, I've sent last mail without finishing :)
>>>>>
>>>>>
https://kamailio.org/docs/modules/5.5.x/modules/sipdump.html
>>>>>
>>>>> You only have to load the module and set:
>>>>>
>>>>> modparam("sipdump", "enable", 1)
>>>>>
>>>>>
>>>>> Also you can enable or disable using RPC commands:
>>>>>
>>>>> kamcmd sipdump.enable
>>>>> kamcmd sipdump.enable 1
>>>>> kamcmd sipdump.enable 0
>>>>>
>>>>>
>>>>> Regards
>>>>>
>>>>> On Tue, 1 Sep 2020 at 13:37, Pepelux <pepeluxx(a)gmail.com>
wrote:
>>>>>
>>>>>> Hi
>>>>>>
>>>>>>
https://kamailio.org/docs/modules/5.5.x/modules/sipdump.html
>>>>>>
>>>>>> You only have to load the module and set:
>>>>>>
>>>>>> modparam("sipdump", "enable", 1)
>>>>>>
>>>>>> kamcmd sipdump.enable 1
>>>>>> kamcmd sipdump.enable 0
>>>>>>
>>>>>> modparam("sipdump", "enable", 1)
>>>>>>
>>>>>>
>>>>>> On Tue, 1 Sep 2020 at 13:23, sip user
<sipuser404(a)gmail.com>
>>>>>> wrote:
>>>>>>
>>>>>>> Hi Daniel..
>>>>>>>
>>>>>>> And how load sipdump?
>>>>>>> I'm using kamailio 5.2.1-1 and I think sipdump module is
not
>>>>>>> available, right?
>>>>>>>
>>>>>>> Thanks
>>>>>>>
>>>>>>> El mar., 1 sept. 2020 a las 12:27, Daniel-Constantin Mierla
(<
>>>>>>> miconda(a)gmail.com>) escribió:
>>>>>>>
>>>>>>>> Hello,
>>>>>>>>
>>>>>>>> it seems that the ACK comes in, but my guess is that the
R-URI
>>>>>>>> is not properly set. From the logs it looks like same
value as for To
>>>>>>>> header URI, while it should be the address in Contact
header of 200ok for
>>>>>>>> INVITE.
>>>>>>>>
>>>>>>>> Load the sipdump module and that will save all the sip
traffic
>>>>>>>> in a text file, making it easier to see what comes/goes
on both directions,
>>>>>>>> no matter is over tls or not. If you use kamailio devel
version (master
>>>>>>>> branch), then sipdump module can also store traffic in
pcap file (tls
>>>>>>>> traffic saved as udp for simplicity, but it is easy to
spot from headers or
>>>>>>>> meta data extra header).
>>>>>>>>
>>>>>>>> You can send the sipdump file here for investigation, so
we can
>>>>>>>> see if some headers or r-uri are not correct.
>>>>>>>>
>>>>>>>> Cheers,
>>>>>>>> Daniel
>>>>>>>> On 01.09.20 11:15, sip user wrote:
>>>>>>>>
>>>>>>>> Hi Daniel, thanks for answered to me...
>>>>>>>>
>>>>>>>> With debug=3 I see that:
>>>>>>>>
>>>>>>>> kamailio[1096]: 9(1109) DEBUG: <core>
>>>>>>>> [core/parser/msg_parser.c:610]: parse_msg(): SIP
Request:
>>>>>>>> kamailio[1096]: 9(1109) DEBUG: <core>
>>>>>>>> [core/parser/msg_parser.c:612]: parse_msg(): method:
<ACK>
>>>>>>>> kamailio[1096]: 9(1109) DEBUG: <core>
>>>>>>>> [core/parser/msg_parser.c:614]: parse_msg(): uri:
>>>>>>>> <sip:+34590@FQND:5061;user=phone;transport=tls>
>>>>>>>> kamailio[1096]: 9(1109) DEBUG: <core>
>>>>>>>> [core/parser/msg_parser.c:616]: parse_msg(): version:
<SIP/2.0>
>>>>>>>> kamailio[1096]: 9(1109) DEBUG: <core>
>>>>>>>> [core/parser/parse_addr_spec.c:185]: parse_to_param():
add param:
>>>>>>>> tag=92e2fd8688a9d17b927d9be2f84faa55-8079
>>>>>>>> kamailio[1096]: 9(1109) DEBUG: <core>
>>>>>>>> [core/parser/parse_addr_spec.c:864]: parse_addr_spec():
end of header
>>>>>>>> reached, state=29
>>>>>>>> kamailio[1096]: 9(1109) DEBUG: <core>
>>>>>>>> [core/parser/msg_parser.c:171]: get_hdr_field():
<TO> [94]; uri=[
>>>>>>>> sip:+34590@FQND:5061;user=phone]
>>>>>>>> kamailio[1096]: 9(1109) DEBUG: <core>
>>>>>>>> [core/parser/msg_parser.c:174]: get_hdr_field(): to body
[
>>>>>>>> <sip:+34590@FQND:5061;user=phone>], to tag
>>>>>>>> [92e2fd8688a9d17b927d9be2f84faa55-8079]
>>>>>>>> kamailio[1096]: 9(1109) DEBUG: <core>
>>>>>>>> [core/parser/msg_parser.c:152]: get_hdr_field(): cseq
<CSEQ>: <1> <ACK>
>>>>>>>> kamailio[1096]: 9(1109) DEBUG: <core>
>>>>>>>> [core/parser/parse_via.c:1303]: parse_via_param(): Found
param type 232,
>>>>>>>> <branch> = <z9hG4bKf4784e39>; state=16
>>>>>>>> kamailio[1096]: 9(1109) DEBUG: <core>
>>>>>>>> [core/parser/parse_via.c:2639]: parse_via(): end of
header reached, state=5
>>>>>>>> kamailio[1096]: 9(1109) DEBUG: <core>
>>>>>>>> [core/parser/msg_parser.c:498]: parse_headers(): Via
found, flags=2
>>>>>>>> kamailio[1096]: 9(1109) DEBUG: <core>
>>>>>>>> [core/parser/msg_parser.c:500]: parse_headers(): this is
the first via
>>>>>>>> kamailio[1096]: 9(1109) DEBUG: <core>
[core/receive.c:240]:
>>>>>>>> receive_msg(): --- received sip message - request -
call-id:
>>>>>>>> [d3649f52dc0057768ec6c18733de8206] - cseq: [1 ACK]
>>>>>>>> kamailio[1096]: 9(1109) DEBUG: <core>
>>>>>>>> [core/parser/msg_parser.c:185]: get_hdr_field():
content_length=0
>>>>>>>> kamailio[1096]: 9(1109) DEBUG: <core>
>>>>>>>> [core/parser/msg_parser.c:89]: get_hdr_field(): found end
of header
>>>>>>>> kamailio[1096]: 9(1109) DEBUG: {1 1 ACK
>>>>>>>> d3649f52dc0057768ec6c18733de8206} <core>
[core/receive.c:295]:
>>>>>>>> receive_msg(): preparing to run routing scripts...
>>>>>>>> kamailio[1096]: 9(1109) DEBUG: {1 1 ACK
>>>>>>>> d3649f52dc0057768ec6c18733de8206} sl [sl_funcs.c:397]:
sl_filter_ACK(): too
>>>>>>>> late to be a local ACK!
>>>>>>>>
>>>>>>>> So, I understand that ACK comes from Teams, right? So
kamailio
>>>>>>>> routing problem?
>>>>>>>>
>>>>>>>> Thanks
>>>>>>>>
>>>>>>>> El mar., 25 ago. 2020 a las 15:32, Daniel-Constantin
Mierla (<
>>>>>>>> miconda(a)gmail.com>) escribió:
>>>>>>>>
>>>>>>>>> Hello,
>>>>>>>>>
>>>>>>>>> run with debug=3 in kamailio.cfg and see if the ACK
comes to
>>>>>>>>> Kamailio, if yes, then some routing issue in
kamailio.cfg. If does not
>>>>>>>>> come, you will have to check the headers to see if MS
Teams expects
>>>>>>>>> something else there, typically is about Record-Route
domains...
>>>>>>>>>
>>>>>>>>> Cheers,
>>>>>>>>> Daniel
>>>>>>>>> On 20.08.20 12:25, sip user wrote:
>>>>>>>>>
>>>>>>>>> Hi, I'm connecting Teams with kamailio server.
From Kamailio
>>>>>>>>> to teams I have no problems, but from teams to
Kamailio yes. Drop the call..
>>>>>>>>>
>>>>>>>>> With ngrep I see that:
>>>>>>>>>
>>>>>>>>> INVITE
>>>>>>>>>
sip:1005@CLIENT_IP:55766;transport=UDP;rinstance=d47edf336726e940
>>>>>>>>> SIP/2.0.
>>>>>>>>> Record-Route: <sip:FQND_IP;r2=on;lr>.
>>>>>>>>> Record-Route:
<sip:FQND_IP:5061;transport=tls;r2=on;lr>.
>>>>>>>>> FROM: "Javier Gonz..lez Mu..oz"
>>>>>>>>>
<sip:+324@sip.pstnhub.microsoft.com:5061;user=phone>
>>>>>>>>> ;tag=c17bb1eb7f8649d4a89d8d4a876ac32b.
>>>>>>>>> TO: <sip:+34560@FQND:5061;user=phone>.
>>>>>>>>> CSEQ: 1 INVITE.
>>>>>>>>> CALL-ID: c1364913e582553a9a9c2544c3583b0a.
>>>>>>>>> MAX-FORWARDS: 69.
>>>>>>>>> Via: SIP/2.0/UDP
>>>>>>>>>
92.222.217.64;branch=z9hG4bK7bb5.0fbae76a37829205e04279f220a26af2.0;i=a1.
>>>>>>>>> VIA: SIP/2.0/TLS
52.114.76.76:5061;branch=z9hG4bKd216a55.
>>>>>>>>> RECORD-ROUTE:
>>>>>>>>>
<sip:sip-du-a-eu.pstnhub.microsoft.com:5061;transport=tls;lr>.
>>>>>>>>> CONTACT:
>>>>>>>>>
<sip:api-du-a-euno.pstnhub.microsoft.com:443;x-i=b0b53fc5-76ef-4619-9a68-13e0a4eea92d;x-c=c1364913e582553a9a9c2544c3583b0a/d/8/6c25eb3789a14bf188ce6b05b5e27891>
>>>>>>>>> .
>>>>>>>>> CONTENT-LENGTH: 1091.
>>>>>>>>> MIN-SE: 300.
>>>>>>>>> SUPPORTED: timer.
>>>>>>>>> USER-AGENT: Microsoft.PSTNHub.SIPProxy v.2020.7.31.1
i.EUNO.0.
>>>>>>>>> CONTENT-TYPE: application/sdp.
>>>>>>>>> ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY.
>>>>>>>>> P-ASSERTED-IDENTITY: <tel:+324>
<+324>,<sip:EMAIL>.
>>>>>>>>> PRIVACY: id.
>>>>>>>>> SESSION-EXPIRES: 3600.
>>>>>>>>> .
>>>>>>>>> v=0.
>>>>>>>>> o=- 165103 0 IN IP4 127.0.0.1.
>>>>>>>>> s=session.
>>>>>>>>> c=IN IP4 52.113.44.8.
>>>>>>>>> b=CT:10000000.
>>>>>>>>> t=0 0.
>>>>>>>>> m=audio 50452 RTP/SAVP 104 9 103 111 18 0 8 97 101 13
118.
>>>>>>>>> c=IN IP4 52.113.44.8.
>>>>>>>>> a=rtcp:50453.
>>>>>>>>> a=ice-ufrag:FZTb.
>>>>>>>>> a=ice-pwd:yD3P7nr+xNq0VYdv7xpB1F+Y.
>>>>>>>>> a=rtcp-mux.
>>>>>>>>> a=candidate:1 1 UDP 2130706431 52.113.44.8 50452 typ
srflx
>>>>>>>>> raddr 10.0.33.240 rport 50
>>>>>>>>>
>>>>>>>>> U CLIENT_IP:55766 -> FQND_IP:5060 #2
>>>>>>>>> SIP/2.0 180 Ringing.
>>>>>>>>> Via: SIP/2.0/UDP
>>>>>>>>>
FQND_IP;branch=z9hG4bK7bb5.0fbae76a37829205e04279f220a26af2.0;i=a1.
>>>>>>>>> Via: SIP/2.0/TLS
52.114.76.76:5061;branch=z9hG4bKd216a55.
>>>>>>>>> Record-Route: <sip:FQND_IP;lr;r2=on>.
>>>>>>>>> Record-Route:
<sip:FQND_IP:5061;transport=tls;r2=on;lr>.
>>>>>>>>> Record-Route:
>>>>>>>>>
<sip:sip-du-a-eu.pstnhub.microsoft.com:5061;transport=tls;lr>.
>>>>>>>>> Contact:
>>>>>>>>>
<sip:1005@CLIENT_IP:55766;transport=UDP;rinstance=d47edf336726e940>
>>>>>>>>> .
>>>>>>>>> To:
<sip:+34560@FQND:5061;user=phone>;tag=de4e6b45.
>>>>>>>>> From: "Javier Gonz..lez Mu..oz"
>>>>>>>>>
<sip:+324@sip.pstnhub.microsoft.com:5061;user=phone>
>>>>>>>>> ;tag=c17bb1eb7f8649d4a89d8d4a876ac32b.
>>>>>>>>> Call-ID: c1364913e582553a9a9c2544c3583b0a.
>>>>>>>>> CSeq: 1 INVITE.
>>>>>>>>> User-Agent: 3CXPhone 6.0.26523.0.
>>>>>>>>> Content-Length: 0.
>>>>>>>>>
>>>>>>>>> U CLIENT_IP:55766 -> FQND_IP:5060 #3
>>>>>>>>> SIP/2.0 200 OK.
>>>>>>>>> Via: SIP/2.0/UDP
>>>>>>>>>
FQND_IP;branch=z9hG4bK7bb5.0fbae76a37829205e04279f220a26af2.0;i=a1.
>>>>>>>>> Via: SIP/2.0/TLS
52.114.76.76:5061;branch=z9hG4bKd216a55.
>>>>>>>>> Record-Route: <sip:FQND_IP;lr;r2=on>.
>>>>>>>>> Record-Route:
<sip:FQND_IP:5061;transport=tls;r2=on;lr>.
>>>>>>>>> Record-Route:
>>>>>>>>>
<sip:sip-du-a-eu.pstnhub.microsoft.com:5061;transport=tls;lr>.
>>>>>>>>> Contact:
>>>>>>>>>
<sip:1005@CLIENT_IP:55766;transport=UDP;rinstance=d47edf336726e940>
>>>>>>>>> .
>>>>>>>>> To:
<sip:+34560@FQND:5061;user=phone>;tag=de4e6b45.
>>>>>>>>> From: "Javier Gonz..lez Mu..oz"
>>>>>>>>>
<sip:+324@sip.pstnhub.microsoft.com:5061;user=phone>
>>>>>>>>> ;tag=c17bb1eb7f8649d4a89d8d4a876ac32b.
>>>>>>>>> Call-ID: c1364913e582553a9a9c2544c3583b0a.
>>>>>>>>> CSeq: 1 INVITE.
>>>>>>>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER,
SUBSCRIBE,
>>>>>>>>> NOTIFY, REFER, INFO, MESSAGE.
>>>>>>>>> Content-Type: application/sdp.
>>>>>>>>> Supported: replaces.
>>>>>>>>> User-Agent: 3CXPhone 6.0.26523.0.
>>>>>>>>> Content-Length: 1067.
>>>>>>>>> .
>>>>>>>>> v=0.
>>>>>>>>> o=3cxVCE 324945090 117647850 IN IP4 .
>>>>>>>>> s=3cxVCE Audio Call.
>>>>>>>>> t=0 0.
>>>>>>>>> m=audio 0 RTP/SAVP 104 9 103 111 18 0 8 97 101 13
118.
>>>>>>>>> c=IN IP4 52.113.44.8.
>>>>>>>>> a=rtpmap:104 SILK/16000.
>>>>>>>>> a=rtpmap:9 G722/8000.
>>>>>>>>> a=rtpmap:103 SILK/8000.
>>>>>>>>> a=rtpmap:111 SIREN/16000.
>>>>>>>>> a=fmtp:111 bitrate=16000.
>>>>>>>>> a=rtpmap:18 G729/8000.
>>>>>>>>> a=fmtp:18 annexb=no.
>>>>>>>>> a=rtpmap:0 PCMU/8000.
>>>>>>>>> a=rtpmap:8 PCMA/8000.
>>>>>>>>> a=rtpmap:97 RED/8000.
>>>>>>>>> a=rtpmap:101 telephone-event/8000.
>>>>>>>>> a=fmtp:101 0-16.
>>>>>>>>> a=rtpmap:13 CN/8000.
>>>>>>>>> a=rtpmap:118 CN/16000.
>>>>>>>>> a=rtcp:50453.
>>>>>>>>> a=ice-ufrag:FZTb.
>>>>>>>>> a=ice-pwd:yD3P7nr+xNq0VYdv7xpB1F+Y.
>>>>>>>>> a=rtcp-mux.
>>>>>>>>> a=candidate:1 1 UDP 213
>>>>>>>>>
>>>>>>>>> I never received ACK..
>>>>>>>>>
>>>>>>>>> In my configuration:
>>>>>>>>>
>>>>>>>>> Kamailio.cfg:
>>>>>>>>>
>>>>>>>>> #!KAMAILIO
>>>>>>>>> #!define WITH_TLS
>>>>>>>>>
>>>>>>>>> event_route[tm:local-request] {
>>>>>>>>>
>>>>>>>>> if(is_method("OPTIONS") &&
$ru =~ "
>>>>>>>>> pstnhub.microsoft.com") {
>>>>>>>>> append_hf("Contact:
>>>>>>>>> <sip:FQND:5061;transport=tls>\r\n");
>>>>>>>>> }
>>>>>>>>> xlog("L_INFO", "Sent out tm
request: $mb\n");
>>>>>>>>> }
>>>>>>>>>
>>>>>>>>> request_route{
>>>>>>>>>
>>>>>>>>> remove_hf("Route");
>>>>>>>>> if (is_method("INVITE|SUBSCRIBE"))
{
>>>>>>>>> xlog("L_INFO","$fU is
trying to call to $rU
>>>>>>>>> con valores $tu\n");
>>>>>>>>> $rU="1005";
>>>>>>>>> }
>>>>>>>>> }
>>>>>>>>>
>>>>>>>>> What I'm doing wrong?
>>>>>>>>>
>>>>>>>>> I don't understand why not received ACK..
>>>>>>>>>
>>>>>>>>> Could anyone help me?
>>>>>>>>>
>>>>>>>>> Thanks
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> Kamailio (SER) - Users Mailing
Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>>>>>>>>
>>>>>>>>> --
>>>>>>>>> Daniel-Constantin Mierla --
www.asipto.comwww.twitter.com/miconda --
www.linkedin.com/in/miconda
>>>>>>>>> Funding:
https://www.paypal.me/dcmierla
>>>>>>>>>
>>>>>>>>> --
>>>>>>>> Daniel-Constantin Mierla --
www.asipto.comwww.twitter.com/miconda --
www.linkedin.com/in/miconda
>>>>>>>> Funding:
https://www.paypal.me/dcmierla
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>> Kamailio (SER) - Users Mailing List
>>>>>>> sr-users(a)lists.kamailio.org
>>>>>>>
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>>>>>>
>>>>>> _______________________________________________
>>>>> Kamailio (SER) - Users Mailing List
>>>>> sr-users(a)lists.kamailio.org
>>>>>
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>>>>
>>>> _______________________________________________
>>>> Kamailio (SER) - Users Mailing List
>>>> sr-users(a)lists.kamailio.org
>>>>
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>>>
>>> _______________________________________________
>>> Kamailio (SER) - Users Mailing List
>>> sr-users(a)lists.kamailio.org
>>>
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>>
>> _______________________________________________
>> Kamailio (SER) - Users Mailing List
>> sr-users(a)lists.kamailio.org
>>
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>
> _______________________________________________
> Kamailio (SER) - Users Mailing List
> sr-users(a)lists.kamailio.org
>
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
_______________________________________________
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sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
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