sr-users November 2016

sr-users@lists.kamailio.org

57 participants
124 discussions

How to get Destination port UDP
by andrzej.ciupek-asterisk.edu.pl 11 Nov '16

11 Nov '16

Hello I have some UAC like Panasonic PBX, that send traffic to port 6060 that I am listening on, but that port isn't included to R-URI. I can see that only at tcpdump, or sip_trace from sip_capture module. Kamailio variables like $dp or $rp have default value of 5060. Thank You,

3 2

Execution Error while getting profile size with jsonrpc-s
by Aqs Younas 11 Nov '16

11 Nov '16

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.

2 4

Re: [SR-Users] BYE dispatcher
by Slava Bendersky 11 Nov '16

11 Nov '16

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

2 1

Kamailio 4.4 VoiceMail
by Gonzalo Gasca Meza 10 Nov '16

10 Nov '16

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; }

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Re: [SR-Users] BYE dispatcher
by Slava Bendersky 10 Nov '16

10 Nov '16

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

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How to get Destination port UDP
by andrzej.ciupek-asterisk.edu.pl 10 Nov '16

10 Nov '16

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

2 1

Change CLID
by Nelson Migliaro 10 Nov '16

10 Nov '16

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.-

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Re: [SR-Users] BYE dispatcher
by Slava Bendersky 10 Nov '16

10 Nov '16

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

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Re: [SR-Users] BYE dispatcher
by Slava Bendersky 10 Nov '16

10 Nov '16

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

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Re: [SR-Users] BYE dispatcher
by Slava Bendersky 10 Nov '16

10 Nov '16

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

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sr-users November 2016