Greetings list,
I am trying to get profile size with jsonrpc-s module. Below is jsonrpc-s
configuration and a curl command to get the profile size.
listen=tcp:0.0.0.0:5060
loadmodule "xhttp"
loadmodule "jsonrpc-s"
modparam("xhttp", "url_match", "^/rpc_path/")
modparam("jsonrpc-s", "pretty_format", 1)
event_route[xhttp:request] {
if($hu =~ "^/rpc_path")
jsonrpc_dispatch();
else
xhttp_reply("404", "Not Found", "text/html", "");
}
curl -v -H 'Content-Type: application/json' -H 'Call-ID: abc12' -d
'{"jsonrpc": "2.0", "method": "dlg.profile_get_size","params":{"accountno":
"1234567"}, "id": "0deadb33f"}' http://127.0.0.1:5060/rpc_path/
I am getting this in respose.
* Hostname was NOT found in DNS cache
* Trying 127.0.0.1...
* Connected to 127.0.0.1 (127.0.0.1) port 5060 (#0)
> POST /rpc_path/ HTTP/1.1
> User-Agent: curl/7.38.0
> Host: 127.0.0.1:5060
> Accept: */*
> Content-Type: application/json
> Call-ID: abc12
> Content-Length: 106
>
* upload completely sent off: 106 out of 106 bytes
*< HTTP/1.1 404 Profile not found: 1234567*
< Sia: SIP/2.0/TCP 127.0.0.1:49374
< Call-ID: abc12
< Content-Type: application/json
* Server kamailio (4.4.1 (x86_64/linux)) is not blacklisted
< Server: kamailio (4.4.1 (x86_64/linux))
< Content-Length: 106
<
{
"jsonrpc": "2.0",
"error": {
"code": -32000,
"message": "Execution Error"
},
"id": "0deadb33f"
* Connection #0 to host 127.0.0.1 left intact
Whereas fifo command gives me this.
root@debian:/usr/local/kamailio/sbin# ./kamctl fifo profile_get_size
accountno 1234567
profile:: name=accountno value=1234567 count=2
Why jsonrpc-s is search for profile 1234567 whereas profile should be
accountno.
Anyhelp what i am doing wrong here is much appreciated.
Best Regards.
Hello Daniel,
I really ask for help, here are configuration file
https://paste.fedoraproject.org/477652/88413891/
I spent quite a lot of time trying understand loose_route() /record_route() mix.
I can get signalling working, call is not disconnects, but no RTP. Or I can get rtp and signalling BYE is not routed properly.
My setup is just proxy all requests to freesiwtch boxes base on dispatcher selection where kamailio setup with 2 interfaces public and private.
I really appreciate on you time and help.
Slava.
From: "volga629" <volga629(a)skillsearch.ca>
To: "sr-users" <sr-users(a)lists.sip-router.org>
Cc: miconda(a)gmail.com
Sent: Thursday, 10 November, 2016 23:54:40
Subject: Re: [SR-Users] BYE dispatcher
Hello Daniel,
What
From: "volga629" <volga629(a)skillsearch.ca>
To: miconda(a)gmail.com
Cc: "sr-users" <sr-users(a)lists.sip-router.org>
Sent: Thursday, 10 November, 2016 11:25:19
Subject: Re: [SR-Users] BYE dispatcher
Hello Daniel,
My setup is proxy all requests to freeswitch via dispatcher.
Slava.
From: "Daniel-Constantin Mierla" <miconda(a)gmail.com>
To: "volga629" <volga629(a)skillsearch.ca>, "sr-users" <sr-users(a)lists.sip-router.org>
Sent: Thursday, 10 November, 2016 04:56:53
Subject: Re: [SR-Users] BYE dispatcher
Hello,
as I said before, the registrations have little to do with calls in sip, unless there is gruu in use.
Cheers,
Daniel
On 09/11/16 18:07, Slava Bendersky wrote:
Hello Everyone,
I cleared registrations and tried again and issue still present.
Client reply with 481.
IP (tos 0x0, ttl 52, id 7731, offset 0, flags [none], proto UDP (17), length 638)
client_pub_ip.49383 > proxy_pub_ip.llrp: [udp sum ok] UDP, length 610
E..~.3..4...c....E.\.....j..SIP/2.0 481 Call Leg/Transaction Does Not Exist
Via: SIP/2.0/UDP proxy_pub_ip :5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0
Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp
From: "Test Extension" <sip:4300@sip.company.tld> ;tag=SXt3DQQ90a0Dj
To: < sip:4300@ client_pub_ip :49383>;tag=719973534
Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb
CSeq: 99019404 BYE
Supported: replaces, path, eventlist
User-Agent: Grandstream Wave 1.2.2
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0
Slava.
From: "volga629" <volga629(a)skillsearch.ca>
To: miconda(a)gmail.com , "sr-users" <sr-users(a)lists.sip-router.org>
Sent: Wednesday, 9 November, 2016 12:28:32
Subject: Re: [SR-Users] BYE dispatcher
Hello Everyone,
I changed dispatcher algorithm from 0 to 1 and start working as expected. Yes group 0 is accepted.
route[DISPATCHER] {
if(!ds_select_dst("0", "1")) {
xlog("L_ERROR","ERROR: Proxy Mapping - Desitnation for $fd not found...request dropped \n");
sl_send_reply("404","Desitination Not Found \n");
drop();
} else {
$var(did) = 1;
}
if($var(did)) {
if (!t_relay()) {
sl_reply_error();
}
#forward();
}
t_on_failure("DISPATCHER_FAIL_ROUTE");
exit;
}
Slava.
From: "Daniel-Constantin Mierla" <miconda(a)gmail.com>
To: "sr-users" <sr-users(a)lists.sip-router.org>
Sent: Wednesday, 9 November, 2016 04:33:33
Subject: Re: [SR-Users] BYE dispatcher
Hello,
On 08/11/16 20:42, Slava Bendersky wrote:
BQ_BEGIN
Hello Everyone,
My setup is kamailio as proxy with few boxes of freeswitch in the LAN. Having issue with BYE when extensions register on different freeswitch boxes. Here are some trace of the call.
Not sure if this tag= miss match or routing.
Dispatcher use group 0 with option 4 (round robin).
is group value 0 accepted? I think this may create problems if a function returns the group in the config as return code -- iirc, this was changed maybe for lcr or permissions.
On the other hand, the registrations are quite independent in SIP in relation with calls. The BYE should be routed based on record-routing to the freeswitch that was involved in routing initial INVITE, with no relation to new registrations from end devices. Is the BYE sent to the freeswitch that got the initial BYE.
Cheers,
Daniel
--
Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
BQ_END
--
Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
I have the following call flow:
INVITE -> sip:gonzalo@sip.parzee.io -- TLS/TCP/UDP -> *KAMAILIO* - DB
Lookup -> INVITE sip:gonzalo58@test.external.com;transport=tls (Phone1)
If Phone1 is Busy or No answer, I want call to go to VM.
Phone1, is not registered to Kamailio, nor I'm using Realtime Integration.
This Phone1 is registered to an external PBX.
Currently in sample configuration script, seems to be that value: $avp(oexten)
is used to redirect to VM, but in my case this value is null.
I didnt find any documentation for this.
*Questions:*
a) What is $avp(oexten) ?
b) What is the best way to pass a Redirect number in SIP INVITE to
VoiceMail system (Asterisk or Freeswitch)
c) Is there a way to configure CFNA timer per alias/uri ?
*Example*:
route[TOVOICEMAIL] {
#!ifdef WITH_VOICEMAIL
if(!is_method("INVITE|SUBSCRIBE")) return;
# check if VoiceMail server IP is defined
if (strempty($sel(cfg_get.voicemail.srv_ip))) {
xlog("SCRIPT: VoiceMail routing enabled but IP not
defined\n");
return;
}
if(is_method("INVITE")) {
xdbg("SIP Request: method [$rm] from [$fu] to [$tu]\n");
xlog("VoiceMail routing enabled $avp(oexten)\n");
if($avp(oexten)==$null) return;
$ru = "sip:" + $avp(oexten) + "@" +
$sel(cfg_get.voicemail.srv_ip)
+ ":" + $sel(cfg_get.voicemail.srv_port);
xlog("SCRIPT: VoiceMail to $tu\n");
} else {
if($rU==$null) return;
$ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip)
+ ":" + $sel(cfg_get.voicemail.srv_port);
}
route(RELAY);
exit;
#!endif
return;
}
Hello Daniel,
My setup is proxy all requests to freeswitch via dispatcher.
Slava.
From: "Daniel-Constantin Mierla" <miconda(a)gmail.com>
To: "volga629" <volga629(a)skillsearch.ca>, "sr-users" <sr-users(a)lists.sip-router.org>
Sent: Thursday, 10 November, 2016 04:56:53
Subject: Re: [SR-Users] BYE dispatcher
Hello,
as I said before, the registrations have little to do with calls in sip, unless there is gruu in use.
Cheers,
Daniel
On 09/11/16 18:07, Slava Bendersky wrote:
Hello Everyone,
I cleared registrations and tried again and issue still present.
Client reply with 481.
IP (tos 0x0, ttl 52, id 7731, offset 0, flags [none], proto UDP (17), length 638)
client_pub_ip.49383 > proxy_pub_ip.llrp: [udp sum ok] UDP, length 610
E..~.3..4...c....E.\.....j..SIP/2.0 481 Call Leg/Transaction Does Not Exist
Via: SIP/2.0/UDP proxy_pub_ip :5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0
Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp
From: "Test Extension" <sip:4300@sip.company.tld> ;tag=SXt3DQQ90a0Dj
To: < sip:4300@ client_pub_ip :49383>;tag=719973534
Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb
CSeq: 99019404 BYE
Supported: replaces, path, eventlist
User-Agent: Grandstream Wave 1.2.2
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0
Slava.
From: "volga629" <volga629(a)skillsearch.ca>
To: miconda(a)gmail.com , "sr-users" <sr-users(a)lists.sip-router.org>
Sent: Wednesday, 9 November, 2016 12:28:32
Subject: Re: [SR-Users] BYE dispatcher
Hello Everyone,
I changed dispatcher algorithm from 0 to 1 and start working as expected. Yes group 0 is accepted.
route[DISPATCHER] {
if(!ds_select_dst("0", "1")) {
xlog("L_ERROR","ERROR: Proxy Mapping - Desitnation for $fd not found...request dropped \n");
sl_send_reply("404","Desitination Not Found \n");
drop();
} else {
$var(did) = 1;
}
if($var(did)) {
if (!t_relay()) {
sl_reply_error();
}
#forward();
}
t_on_failure("DISPATCHER_FAIL_ROUTE");
exit;
}
Slava.
From: "Daniel-Constantin Mierla" <miconda(a)gmail.com>
To: "sr-users" <sr-users(a)lists.sip-router.org>
Sent: Wednesday, 9 November, 2016 04:33:33
Subject: Re: [SR-Users] BYE dispatcher
Hello,
On 08/11/16 20:42, Slava Bendersky wrote:
BQ_BEGIN
Hello Everyone,
My setup is kamailio as proxy with few boxes of freeswitch in the LAN. Having issue with BYE when extensions register on different freeswitch boxes. Here are some trace of the call.
Not sure if this tag= miss match or routing.
Dispatcher use group 0 with option 4 (round robin).
is group value 0 accepted? I think this may create problems if a function returns the group in the config as return code -- iirc, this was changed maybe for lcr or permissions.
On the other hand, the registrations are quite independent in SIP in relation with calls. The BYE should be routed based on record-routing to the freeswitch that was involved in routing initial INVITE, with no relation to new registrations from end devices. Is the BYE sent to the freeswitch that got the initial BYE.
Cheers,
Daniel
--
Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
BQ_END
--
Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com
Hello
Thank You. In using Kamailio only as:
route {
if (is_method("INVITE"))
{
sql_query("cb", "insert into invites
(from_user,to_user,callid,src_ip,dst_ip,src_port,dst_port) values
('$fU','$rU','$ci','$si','$rd','$sp','$rp')");
}
}
I know that
sip_capture();
In some way get real DST port even if there is no port in r-URI.
Greetings
Hello,
I have several SIP vendors and I must present a different CLID for each of
them.
I am planning to use failure routers in Kamailio to route calls to a
different vendor in case of failure.
The problem I face is changing clid in each route.
I am planning to use a mysql database to select the clid based on a prefix
and sip vendor and in case of failure, select new clid base on new sip
vendor.
Is mysql the right way to do that, is there a module for this?
In order to change the clid I am thinking of
using: uac_replace_to(display,uri). Is this correct way to do that?
And finally, in case of using uac_replace_to(display,uri), do I have to
use uac_restore_from() to restone the correct from to avoid further issues?
Thank you in advance
Nelson.-
Hello Everyone,
Here are full trace call.
https://paste.fedoraproject.org/476607/14787290/
Slava.
From: "volga629" <volga629(a)skillsearch.ca>
To: "sr-users" <sr-users(a)lists.sip-router.org>
Sent: Wednesday, 9 November, 2016 13:17:34
Subject: Re: [SR-Users] BYE dispatcher
Based on this out put Freeswitch send BYE to kamailio and Route present then kamailio forward BYE to client and no routes. Then client reply 481. Do I need add it ? Is this tag= problem ?
24 is freeswtich and 27 kamailio.
IP (tos 0x0, ttl 64, id 56723, offset 0, flags [none], proto UDP (17), length 704)
10.18.130.24.5160 > 10.18.130.27.sip: [udp sum ok] UDP, length 676
E.......@..B
...
....(....8.BYE sip:4300@client_public_ip:49383 SIP/2.0
Via: SIP/2.0/UDP 10.18.130.24:5160;rport;branch=z9hG4bKm80c0USSKv5Bp
Route: <sip:10.18.130.27;r2=on;lr=on;ftag=SXt3DQQ90a0Dj>
Route: <sip:proxy_public_ip:5084;r2=on;lr=on;ftag=SXt3DQQ90a0Dj>
Max-Forwards: 70
From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj
To: <sip:4300@client_public_ip:49383>;tag=719973534
Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb
CSeq: 99019404 BYE
User-Agent: FreeSWITCH
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Reason: Q.850;cause=16;text="NORMAL_CLEARING"
Content-Length: 0
IP (tos 0x10, ttl 64, id 36705, offset 0, flags [none], proto UDP (17), length 700)
proxy_public_ip.llrp > client_public_ip.49383: [bad udp cksum 0x4d15 -> 0x34be!] UDP, length 672
E....a..@..d.E.\c.........M.BYE sip:4300@client_public_ip:49383 SIP/2.0
Via: SIP/2.0/UDP proxy_public_ip:5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0
Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp
Max-Forwards: 69
From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj
To: <sip:4300@client_public_ip:49383>;tag=719973534
Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb
CSeq: 99019404 BYE
User-Agent: FreeSWITCH
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Reason: Q.850;cause=16;text="NORMAL_CLEARING"
Content-Length: 0
IP (tos 0x0, ttl 52, id 7731, offset 0, flags [none], proto UDP (17), length 638)
client_public_ip.49383 > proxy_public_ip.llrp: [udp sum ok] UDP, length 610
E..~.3..4...c....E.\.....j..SIP/2.0 481 Call Leg/Transaction Does Not Exist
Via: SIP/2.0/UDP proxy_public_ip:5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0
Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp
From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj
To: <sip:4300@client_public_ip:49383>;tag=719973534
Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb
CSeq: 99019404 BYE
Supported: replaces, path, eventlist
User-Agent: Grandstream Wave 1.2.2
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0
Slava.
From: "volga629" <volga629(a)skillsearch.ca>
To: "sr-users" <sr-users(a)lists.sip-router.org>
Sent: Wednesday, 9 November, 2016 13:07:11
Subject: Re: [SR-Users] BYE dispatcher
Hello Everyone,
I cleared registrations and tried again and issue still present.
Client reply with 481.
IP (tos 0x0, ttl 52, id 7731, offset 0, flags [none], proto UDP (17), length 638)
client_pub_ip.49383 > proxy_pub_ip.llrp: [udp sum ok] UDP, length 610
E..~.3..4...c....E.\.....j..SIP/2.0 481 Call Leg/Transaction Does Not Exist
Via: SIP/2.0/UDP proxy_pub_ip :5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0
Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp
From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj
To: <sip:4300@ client_pub_ip :49383>;tag=719973534
Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb
CSeq: 99019404 BYE
Supported: replaces, path, eventlist
User-Agent: Grandstream Wave 1.2.2
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0
Slava.
From: "volga629" <volga629(a)skillsearch.ca>
To: miconda(a)gmail.com, "sr-users" <sr-users(a)lists.sip-router.org>
Sent: Wednesday, 9 November, 2016 12:28:32
Subject: Re: [SR-Users] BYE dispatcher
Hello Everyone,
I changed dispatcher algorithm from 0 to 1 and start working as expected. Yes group 0 is accepted.
route[DISPATCHER] {
if(!ds_select_dst("0", "1")) {
xlog("L_ERROR","ERROR: Proxy Mapping - Desitnation for $fd not found...request dropped \n");
sl_send_reply("404","Desitination Not Found \n");
drop();
} else {
$var(did) = 1;
}
if($var(did)) {
if (!t_relay()) {
sl_reply_error();
}
#forward();
}
t_on_failure("DISPATCHER_FAIL_ROUTE");
exit;
}
Slava.
From: "Daniel-Constantin Mierla" <miconda(a)gmail.com>
To: "sr-users" <sr-users(a)lists.sip-router.org>
Sent: Wednesday, 9 November, 2016 04:33:33
Subject: Re: [SR-Users] BYE dispatcher
Hello,
On 08/11/16 20:42, Slava Bendersky wrote:
Hello Everyone,
My setup is kamailio as proxy with few boxes of freeswitch in the LAN. Having issue with BYE when extensions register on different freeswitch boxes. Here are some trace of the call.
Not sure if this tag= miss match or routing.
Dispatcher use group 0 with option 4 (round robin).
is group value 0 accepted? I think this may create problems if a function returns the group in the config as return code -- iirc, this was changed maybe for lcr or permissions.
On the other hand, the registrations are quite independent in SIP in relation with calls. The BYE should be routed based on record-routing to the freeswitch that was involved in routing initial INVITE, with no relation to new registrations from end devices. Is the BYE sent to the freeswitch that got the initial BYE.
Cheers,
Daniel
--
Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Based on this out put Freeswitch send BYE to kamailio and Route present then kamailio forward BYE to client and no routes. Then client reply 481. Do I need add it ? Is this tag= problem ?
24 is freeswtich and 27 kamailio.
IP (tos 0x0, ttl 64, id 56723, offset 0, flags [none], proto UDP (17), length 704)
10.18.130.24.5160 > 10.18.130.27.sip: [udp sum ok] UDP, length 676
E.......@..B
...
....(....8.BYE sip:4300@client_public_ip:49383 SIP/2.0
Via: SIP/2.0/UDP 10.18.130.24:5160;rport;branch=z9hG4bKm80c0USSKv5Bp
Route: <sip:10.18.130.27;r2=on;lr=on;ftag=SXt3DQQ90a0Dj>
Route: <sip:proxy_public_ip:5084;r2=on;lr=on;ftag=SXt3DQQ90a0Dj>
Max-Forwards: 70
From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj
To: <sip:4300@client_public_ip:49383>;tag=719973534
Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb
CSeq: 99019404 BYE
User-Agent: FreeSWITCH
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Reason: Q.850;cause=16;text="NORMAL_CLEARING"
Content-Length: 0
IP (tos 0x10, ttl 64, id 36705, offset 0, flags [none], proto UDP (17), length 700)
proxy_public_ip.llrp > client_public_ip.49383: [bad udp cksum 0x4d15 -> 0x34be!] UDP, length 672
E....a..@..d.E.\c.........M.BYE sip:4300@client_public_ip:49383 SIP/2.0
Via: SIP/2.0/UDP proxy_public_ip:5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0
Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp
Max-Forwards: 69
From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj
To: <sip:4300@client_public_ip:49383>;tag=719973534
Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb
CSeq: 99019404 BYE
User-Agent: FreeSWITCH
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Reason: Q.850;cause=16;text="NORMAL_CLEARING"
Content-Length: 0
IP (tos 0x0, ttl 52, id 7731, offset 0, flags [none], proto UDP (17), length 638)
client_public_ip.49383 > proxy_public_ip.llrp: [udp sum ok] UDP, length 610
E..~.3..4...c....E.\.....j..SIP/2.0 481 Call Leg/Transaction Does Not Exist
Via: SIP/2.0/UDP proxy_public_ip:5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0
Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp
From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj
To: <sip:4300@client_public_ip:49383>;tag=719973534
Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb
CSeq: 99019404 BYE
Supported: replaces, path, eventlist
User-Agent: Grandstream Wave 1.2.2
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0
Slava.
From: "volga629" <volga629(a)skillsearch.ca>
To: "sr-users" <sr-users(a)lists.sip-router.org>
Sent: Wednesday, 9 November, 2016 13:07:11
Subject: Re: [SR-Users] BYE dispatcher
Hello Everyone,
I cleared registrations and tried again and issue still present.
Client reply with 481.
IP (tos 0x0, ttl 52, id 7731, offset 0, flags [none], proto UDP (17), length 638)
client_pub_ip.49383 > proxy_pub_ip.llrp: [udp sum ok] UDP, length 610
E..~.3..4...c....E.\.....j..SIP/2.0 481 Call Leg/Transaction Does Not Exist
Via: SIP/2.0/UDP proxy_pub_ip :5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0
Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp
From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj
To: <sip:4300@ client_pub_ip :49383>;tag=719973534
Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb
CSeq: 99019404 BYE
Supported: replaces, path, eventlist
User-Agent: Grandstream Wave 1.2.2
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0
Slava.
From: "volga629" <volga629(a)skillsearch.ca>
To: miconda(a)gmail.com, "sr-users" <sr-users(a)lists.sip-router.org>
Sent: Wednesday, 9 November, 2016 12:28:32
Subject: Re: [SR-Users] BYE dispatcher
Hello Everyone,
I changed dispatcher algorithm from 0 to 1 and start working as expected. Yes group 0 is accepted.
route[DISPATCHER] {
if(!ds_select_dst("0", "1")) {
xlog("L_ERROR","ERROR: Proxy Mapping - Desitnation for $fd not found...request dropped \n");
sl_send_reply("404","Desitination Not Found \n");
drop();
} else {
$var(did) = 1;
}
if($var(did)) {
if (!t_relay()) {
sl_reply_error();
}
#forward();
}
t_on_failure("DISPATCHER_FAIL_ROUTE");
exit;
}
Slava.
From: "Daniel-Constantin Mierla" <miconda(a)gmail.com>
To: "sr-users" <sr-users(a)lists.sip-router.org>
Sent: Wednesday, 9 November, 2016 04:33:33
Subject: Re: [SR-Users] BYE dispatcher
Hello,
On 08/11/16 20:42, Slava Bendersky wrote:
Hello Everyone,
My setup is kamailio as proxy with few boxes of freeswitch in the LAN. Having issue with BYE when extensions register on different freeswitch boxes. Here are some trace of the call.
Not sure if this tag= miss match or routing.
Dispatcher use group 0 with option 4 (round robin).
is group value 0 accepted? I think this may create problems if a function returns the group in the config as return code -- iirc, this was changed maybe for lcr or permissions.
On the other hand, the registrations are quite independent in SIP in relation with calls. The BYE should be routed based on record-routing to the freeswitch that was involved in routing initial INVITE, with no relation to new registrations from end devices. Is the BYE sent to the freeswitch that got the initial BYE.
Cheers,
Daniel
--
Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Hello Everyone,
I cleared registrations and tried again and issue still present.
Client reply with 481.
IP (tos 0x0, ttl 52, id 7731, offset 0, flags [none], proto UDP (17), length 638)
client_pub_ip.49383 > proxy_pub_ip.llrp: [udp sum ok] UDP, length 610
E..~.3..4...c....E.\.....j..SIP/2.0 481 Call Leg/Transaction Does Not Exist
Via: SIP/2.0/UDP proxy_pub_ip :5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0
Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp
From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj
To: <sip:4300@ client_pub_ip :49383>;tag=719973534
Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb
CSeq: 99019404 BYE
Supported: replaces, path, eventlist
User-Agent: Grandstream Wave 1.2.2
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0
Slava.
From: "volga629" <volga629(a)skillsearch.ca>
To: miconda(a)gmail.com, "sr-users" <sr-users(a)lists.sip-router.org>
Sent: Wednesday, 9 November, 2016 12:28:32
Subject: Re: [SR-Users] BYE dispatcher
Hello Everyone,
I changed dispatcher algorithm from 0 to 1 and start working as expected. Yes group 0 is accepted.
route[DISPATCHER] {
if(!ds_select_dst("0", "1")) {
xlog("L_ERROR","ERROR: Proxy Mapping - Desitnation for $fd not found...request dropped \n");
sl_send_reply("404","Desitination Not Found \n");
drop();
} else {
$var(did) = 1;
}
if($var(did)) {
if (!t_relay()) {
sl_reply_error();
}
#forward();
}
t_on_failure("DISPATCHER_FAIL_ROUTE");
exit;
}
Slava.
From: "Daniel-Constantin Mierla" <miconda(a)gmail.com>
To: "sr-users" <sr-users(a)lists.sip-router.org>
Sent: Wednesday, 9 November, 2016 04:33:33
Subject: Re: [SR-Users] BYE dispatcher
Hello,
On 08/11/16 20:42, Slava Bendersky wrote:
Hello Everyone,
My setup is kamailio as proxy with few boxes of freeswitch in the LAN. Having issue with BYE when extensions register on different freeswitch boxes. Here are some trace of the call.
Not sure if this tag= miss match or routing.
Dispatcher use group 0 with option 4 (round robin).
is group value 0 accepted? I think this may create problems if a function returns the group in the config as return code -- iirc, this was changed maybe for lcr or permissions.
On the other hand, the registrations are quite independent in SIP in relation with calls. The BYE should be routed based on record-routing to the freeswitch that was involved in routing initial INVITE, with no relation to new registrations from end devices. Is the BYE sent to the freeswitch that got the initial BYE.
Cheers,
Daniel
--
Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Hello Everyone,
I changed dispatcher algorithm from 0 to 1 and start working as expected. Yes group 0 is accepted.
route[DISPATCHER] {
if(!ds_select_dst("0", "1")) {
xlog("L_ERROR","ERROR: Proxy Mapping - Desitnation for $fd not found...request dropped \n");
sl_send_reply("404","Desitination Not Found \n");
drop();
} else {
$var(did) = 1;
}
if($var(did)) {
if (!t_relay()) {
sl_reply_error();
}
#forward();
}
t_on_failure("DISPATCHER_FAIL_ROUTE");
exit;
}
Slava.
From: "Daniel-Constantin Mierla" <miconda(a)gmail.com>
To: "sr-users" <sr-users(a)lists.sip-router.org>
Sent: Wednesday, 9 November, 2016 04:33:33
Subject: Re: [SR-Users] BYE dispatcher
Hello,
On 08/11/16 20:42, Slava Bendersky wrote:
Hello Everyone,
My setup is kamailio as proxy with few boxes of freeswitch in the LAN. Having issue with BYE when extensions register on different freeswitch boxes. Here are some trace of the call.
Not sure if this tag= miss match or routing.
Dispatcher use group 0 with option 4 (round robin).
is group value 0 accepted? I think this may create problems if a function returns the group in the config as return code -- iirc, this was changed maybe for lcr or permissions.
On the other hand, the registrations are quite independent in SIP in relation with calls. The BYE should be routed based on record-routing to the freeswitch that was involved in routing initial INVITE, with no relation to new registrations from end devices. Is the BYE sent to the freeswitch that got the initial BYE.
Cheers,
Daniel
--
Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
