- sr-users - kamailio.org

msilo
by Slava Bendersky 15 Nov '16

15 Nov '16

Hello Everyone, I asking question/request to add improvement for msilo module where will have ability set $avp(i:body) in mod params. That will allow use it in m_store() by specifying body message $avp. In my case registration is on B2BUA so kamailio pass through/proxy to destination. In order to check user status online/offline kamailio wait for reply from B2BUA on MESSAGE status if it return error 503 that mean user offline and m_store(0 should store message in db, but the issue that error 503 is in final stage and MESSAGE body is not available any more. The improvement will allow specify body $var in m_store() so new way will be m_store($tu, $avp(body)). Any help thank you. Slava.

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Re: [SR-Users] BYE dispatcher
by Slava Bendersky 14 Nov '16

14 Nov '16

Hello Daniel, I resolved all configuration issues and right now call goes both directions include rtp. Again huge thank you for help. Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "volga629" <volga629(a)skillsearch.ca> Cc: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 10 November, 2016 12:24:57 Subject: Re: [SR-Users] BYE dispatcher Hello, this logic is definitely wrong -- FreeSwitch can send also a request, it means that you send it back to it. Only the initial request of a dialog should be routed with rules like dispatcher/load balancer/least cost routing/... The requests within dialog should be routed based on loose routing. Of course, one can think of exceptions, but then you should be fully aware of what kind of routing you do. Cheers, Daniel On 10/11/16 16:25, Slava Bendersky wrote: 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: BQ_BEGIN 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. 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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Homer's kamailio config + enabling siptrace module
by Ján Füri 11 Nov '16

11 Nov '16

Hi guys, I'm using captagent on my all kamalios and freeswitches to capture and send sip to homer server. That works very well. But I would like to enable, on homer's kamailio, the module SipTrace (http://www.kamailio.org/docs/modules/4.4.x/modules/siptrace.html) and save captured data into mysql as well. I already set it up using this mysql structure : http://www.kamailio.org/docs/db-tables/kamailio-db-devel.html#idp1279328 Almost everything works except the column "DIRECTION". Each row is saved as "IN" direction. Do you guys think that can be solved ? If so, can you pleas direct me HOW ... Thx Jan F.

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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,

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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.

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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

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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

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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

2 1

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

2 1

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

10 Nov '16

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

2 1

usrloc module expires column issues
by Grant Bagdasarian 10 Nov '16

10 Nov '16

Hello, When saving in the location table, the expires and last_modified columns contain the value 000-00-00 00:00:00 which causes issues when Kamailio is restarted and the preload modparam is set to load the location table from db in memory at startup. I've already upgraded Kamailio to 4.4.3, and used the dbtables script from version 4.4 to recreate the location table, but without success. Anyone else having this issue? Any resolution for it? Regards, Grant Bagdasarian CM

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

10 Nov '16

Hello Daniel, Here more clear trace https://www.dropbox.com/s/2z6ck71ulidqelh/kamailio-BYE-flow.png?dl=0 BYE send by kamailio to wrong freeswitch box. Slava From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 20:09:53 Subject: Re: [SR-Users] BYE dispatcher Hello Daniel, Please check this sip call flow picture https://www.dropbox.com/s/itdewdg3ph7xcyx/kamailio-fs-BYE-flow.gif?dl=0 Kamilio send BYE to incorrect freeswitch which already responded to BYE from leg one. Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 19:11:56 Subject: Re: [SR-Users] BYE dispatcher 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 _______________________________________________ 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 Daniel, Please check this sip call flow picture https://www.dropbox.com/s/itdewdg3ph7xcyx/kamailio-fs-BYE-flow.gif?dl=0 Kamilio send BYE to incorrect freeswitch which already responded to BYE from leg one. Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 19:11:56 Subject: Re: [SR-Users] BYE dispatcher 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 _______________________________________________ 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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Kamailio v4.4.4 Released
by Daniel-Constantin Mierla 09 Nov '16

09 Nov '16

Hello, Kamailio SIP Server v4.4.4 stable release is out. This is a maintenance release of the latest stable branch, 4.4, that includes fixes since the release of v4.4.3. There is no change to database schema or configuration language structure that you have to do on previous installations of v4.4.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.4.4. For more details about version 4.4.4 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2016/11/kamailio-v4-4-4-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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Count number of early(Ringing) dialogs on gateway
by Aqs Younas 09 Nov '16

09 Nov '16

Greetings list, I am using dialog aur dispatcher modules for calls distribution across multiple gateways. I am able to get number of active dialogs against a gateway but could not get early dialogs going on particular gateway. Dialog information is inserted in database only for state 4. Also, I tried to use dlg_list command to get early dialogs. I see call statistics but "callee_contact::" is empty untill call is answered. for ringing I am getting this. root@debian:/usr/local/kamailio/sbin# ./kamctl fifo dlg_list dialog:: hash=3443:9422 state:: 2 ref_count:: 1 timestart:: 0 timeout:: 0 callid:: GGjl58SQ32LXQB1dV2-CBA.. from_uri:: sip:1015@192.168.10.39:5060;transport=UDP from_tag:: 652ee907 caller_contact:: sip:1015@192.168.10.41:55842;transport=UDP caller_cseq:: 1 caller_route_set:: caller_bind_addr:: udp:192.168.10.39:5060 callee_bind_addr:: to_uri:: sip:1010@192.168.10.39:5060;transport=UDP to_tag:: callee_contact:: callee_cseq:: callee_route_set:: callee_contact:: is empty. Is there any way to get early dialog info againts a gateway.? Or any way to insert early dialog into database with callee_contact not empty? Best Regards, Aqs Younas

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Planning the release of Kamailio v4.4.4
by Daniel-Constantin Mierla 09 Nov '16

09 Nov '16

Hello, I am considering to package a new release out of latest stable branch 4.4, respectively v4.4.4. The target date is one of the days between Nov 8-10, a matter of work load that needs to be done for getting it out. As usual, if you are aware of any issue not yet reported on github bug tracker, do it as soon as possible to give it a chance to be fixed: - https://github.com/kamailio/kamailio/issues 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

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Re: [SR-Users] fnmatch
by Slava Bendersky 09 Nov '16

09 Nov '16

Hello Daniel, Yes, reply was 202 not 200. Right now it work as expected. Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Monday, 7 November, 2016 04:51:11 Subject: Re: [SR-Users] fnmatch Hello, for clarification -- what's the actual problem here? fnmatch("$rs", "200") is matching the 202? Cheers, Daniel On 07/11/16 08:52, Serge S. Yuriev wrote: Hello, In debug you can see 202 Accepted but match configured on 200. Not matched -- Wbr, Serge via mobile 07.11.2016, 01:08, "Slava Bendersky" <volga629(a)skillsearch.ca> : BQ_BEGIN Hello Everyone, I trying fnmatch to find which sip status is returned by B2BUA. But some reason it doesn't work. Nov 6 17:02:15 cavprx00 kamailio: 2(3747) ERROR: *** cfgtrace:onreply_route=[MESSAGE_FORWARD] c=[/etc/kamailio/kamailio-asterisk.cfg] l=456 a=26 n=xlog Nov 6 17:02:15 cavprx00 kamailio: 2(3747) INFO: <script>: Message 1 Accepted by B2BUA --> 202 with Accepted Nov 6 17:02:15 cavprx00 kamailio: 2(3747) ERROR: *** cfgtrace:onreply_route=[MESSAGE_FORWARD] c=[/etc/kamailio/kamailio-asterisk.cfg] l=469 a=16 n=if Nov 6 17:02:15 cavprx00 kamailio: 2(3747) ERROR: *** cfgtrace:onreply_route=[MESSAGE_FORWARD] c=[/etc/kamailio/kamailio-asterisk.cfg] l=457 a=26 n=fnmatch Nov 6 17:02:15 cavprx00 kamailio: 2(3747) ERROR: *** cfgtrace:onreply_route=[DEFAULT_ROUTE] c=[/etc/kamailio/kamailio-asterisk.cfg] l=370 a=2 n=return if(fnmatch("$rs", "200") && (fnmatch("$rr", "Accepted"))) { xlog("L_INFO", "Message Accepted by B2BUA --> $rs with $rr \n"); $var(tu) = $tu; xlog("L_INFO", "New To --> $var(tu)\n"); } Is this fnmatch correct for this operation ? Slava. _______________________________________________ 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

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BYE dispatcher
by Slava Bendersky 09 Nov '16

09 Nov '16

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). client_public_ip.28972 > proxy_public_ip.llrp: [udp sum ok] UDP, length 660 E....2..5...c....E.\q,.....CBYE sip:mod_sofia@10.18.130.24:5160 SIP/2.0 Via: SIP/2.0/UDP client_public_ip:28972;branch=z9hG4bK1248337638;rport Route: <sip:proxy_public_ip:5084;r2=on;lr=on;ftag=01K2Q1BFa9pKH> Route: <sip:10.18.130.27;r2=on;lr=on;ftag=01K2Q1BFa9pKH> From: <sip:4300@client_public_ip:28972>;tag=487304085 To: <sip:4300@sip.company.tld>;tag=01K2Q1BFa9pKH Call-ID: 1443c4ad-2085-1235-b5ad-5254003e39bb CSeq: 98979026 BYE Contact: <sip:4300@client_public_ip:28972> Max-Forwards: 70 Supported: replaces, path, timer, eventlist User-Agent: Grandstream Wave 1.2.2 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 IP (tos 0x10, ttl 64, id 10813, offset 0, flags [none], proto UDP (17), length 930) 10.18.130.27.sip > 10.18.130.24.5160: [bad udp cksum 0x1bf7 -> 0xd85c!] UDP, length 902 E...*=..@.4. ... ......(....BYE sip:mod_sofia@10.18.130.24:5160 SIP/2.0 Record-Route: <sip:10.18.130.27;r2=on;lr=on;ftag=487304085> Record-Route: <sip:proxy_public_ip:5084;r2=on;lr=on;ftag=487304085> Via: SIP/2.0/UDP 10.18.130.27;branch=z9hG4bKa61.22917bc5fdd65f16e39bf09dcbc7316b.0 Via: SIP/2.0/UDP client_public_ip:28972;received=client_public_ip;branch=z9hG4bK1248337638;rport=28972 Route: <sip:proxy_public_ip:5084;r2=on;lr=on;ftag=01K2Q1BFa9pKH> Route: <sip:10.18.130.27;r2=on;lr=on;ftag=01K2Q1BFa9pKH> From: <sip:4300@client_public_ip:28972>;tag=487304085 To: <sip:4300@sip.company.tld>;tag=01K2Q1BFa9pKH Call-ID: 1443c4ad-2085-1235-b5ad-5254003e39bb CSeq: 98979026 BYE Contact: <sip:4300@client_public_ip:28972> Max-Forwards: 69 Supported: replaces, path, timer, eventlist User-Agent: Grandstream Wave 1.2.2 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 IP (tos 0x0, ttl 64, id 38202, offset 0, flags [none], proto UDP (17), length 587) 10.18.130.24.5160 > 10.18.130.27.sip: [udp sum ok] UDP, length 559 E..K.:..@... ... ....(...7i.SIP/2.0 200 OK Via: SIP/2.0/UDP 10.18.130.27;branch=z9hG4bKa61.22917bc5fdd65f16e39bf09dcbc7316b.0 Via: SIP/2.0/UDP client_public_ip:28972;received=client_public_ip;branch=z9hG4bK1248337638;rport=28972 From: <sip:4300@client_public_ip:28972>;tag=487304085 To: <sip:4300@sip.company.tld>;tag=01K2Q1BFa9pKH Call-ID: 1443c4ad-2085-1235-b5ad-5254003e39bb CSeq: 98979026 BYE User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Content-Length: 0 IP (tos 0x10, ttl 64, id 782, offset 0, flags [none], proto UDP (17), length 503) proxy_public_ip.llrp > client_public_ip.28972: [bad udp cksum 0x4c50 -> 0xd018!] UDP, length 475 E.......@.+}.E.\c.....q,..LPSIP/2.0 200 OK Via: SIP/2.0/UDP client_public_ip:28972;received=client_public_ip;branch=z9hG4bK1248337638;rport=28972 From: <sip:4300@client_public_ip:28972>;tag=487304085 To: <sip:4300@sip.company.tld>;tag=01K2Q1BFa9pKH Call-ID: 1443c4ad-2085-1235-b5ad-5254003e39bb CSeq: 98979026 BYE User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Content-Length: 0 IP (tos 0x0, ttl 64, id 10584, offset 0, flags [none], proto UDP (17), length 764) 10.18.130.26.5160 > 10.18.130.27.sip: [udp sum ok] UDP, length 736 E...)X..@.6@ ... ....(....Y^BYE sip:4200@client_public_ip:48147;rinstance=c4084e18f3f39974;alias=client_public_ip~48147~1 SIP/2.0 Via: SIP/2.0/UDP 10.18.130.26:5160;rport;branch=z9hG4bK33785247SrX3H Route: <sip:10.18.130.27;r2=on;lr=on;ftag=d2e84f4a> Route: <sip:proxy_public_ip:5084;r2=on;lr=on;ftag=d2e84f4a> Max-Forwards: 70 From: <sip:4300@sip.company.tld>;tag=5K0FX0ryeBBeH To: "volga629" <sip:4200@sip.company.tld>;tag=d2e84f4a Call-ID: 137861_rel44MTA1ZGMwNzViODg5NDMwYzQ1NDQzYjlmMGExMzlmMmI CSeq: 98979038 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 802, offset 0, flags [none], proto UDP (17), length 741) proxy_public_ip.llrp > client_public_ip.48147: [bad udp cksum 0x4d3e -> 0x4f2d!] UDP, length 713 E...."..@.*{.E.\c.........M>BYE sip:4200@client_public_ip:48147;rinstance=c4084e18f3f39974 SIP/2.0 Via: SIP/2.0/UDP proxy_public_ip:5084;branch=z9hG4bK8395.d46858df1d9e02b910446fca09f1f864.0 Via: SIP/2.0/UDP 10.18.130.26:5160;received=10.18.130.26;rport=5160;branch=z9hG4bK33785247SrX3H Max-Forwards: 69 From: <sip:4300@sip.company.tld>;tag=5K0FX0ryeBBeH To: "volga629" <sip:4200@sip.company.tld>;tag=d2e84f4a Call-ID: 137861_rel44MTA1ZGMwNzViODg5NDMwYzQ1NDQzYjlmMGExMzlmMmI CSeq: 98979038 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 810, offset 0, flags [DF], proto UDP (17), length 504) client_public_ip.48147 > proxy_public_ip.llrp: [udp sum ok] UDP, length 476 E....*@.4..oc....E.\........SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP proxy_public_ip:5084;branch=z9hG4bK8395.d46858df1d9e02b910446fca09f1f864.0 Via: SIP/2.0/UDP 10.18.130.26:5160;received=10.18.130.26;rport=5160;branch=z9hG4bK33785247SrX3H To: "volga629" <sip:4200@sip.company.tld>;tag=d2e84f4a From: <sip:4300@sip.company.tld>;tag=5K0FX0ryeBBeH Call-ID: 137861_rel44MTA1ZGMwNzViODg5NDMwYzQ1NDQzYjlmMGExMzlmMmI CSeq: 98979038 BYE Accept-Language: en Content-Length: 0 IP (tos 0x10, ttl 64, id 63040, offset 0, flags [none], proto UDP (17), length 413) 10.18.130.27.sip > 10.18.130.26.5160: [bad udp cksum 0x19f4 -> 0x6386!] UDP, length 385 E....@..@.j. ... ......(....SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.18.130.26:5160;received=10.18.130.26;rport=5160;branch=z9hG4bK33785247SrX3H To: "volga629" <sip:4200@sip.company.tld>;tag=d2e84f4a From: <sip:4300@sip.company.tld>;tag=5K0FX0ryeBBeH Call-ID: 137861_rel44MTA1ZGMwNzViODg5NDMwYzQ1NDQzYjlmMGExMzlmMmI CSeq: 98979038 BYE Accept-Language: en Content-Length: 0 Slava.

2 1

Count number of early(Ringing) dialogs on gateway
by Aqs Younas 09 Nov '16

09 Nov '16

Greetings list, I am using dialog aur dispatcher modules for calls distribution across multiple gateways. I am able to get number of active dialogs against a gateway but could not get early dialogs going on particular gateway. Dialog information is inserted in database only for state 4. Also, I tried to use dlg_list command to get early dialogs. I see call statistics but "callee_contact::" is empty untill call is answered. For ringing I am getting this. root@debian:/usr/local/kamailio/sbin# ./kamctl fifo dlg_list dialog:: hash=3443:9422 state:: 2 ref_count:: 1 timestart:: 0 timeout:: 0 callid:: GGjl58SQ32LXQB1dV2-CBA.. from_uri:: sip:1015@192.168.10.39:5060;transport=UDP from_tag:: 652ee907 caller_contact:: sip:1015@192.168.10.41:55842;transport=UDP caller_cseq:: 1 caller_route_set:: caller_bind_addr:: udp:192.168.10.39:5060 callee_bind_addr:: to_uri:: sip:1010@192.168.10.39:5060;transport=UDP to_tag:: callee_contact:: callee_cseq:: callee_route_set:: callee_contact:: is empty. Is there any way to get early dialog info against each gateway or every dialog stat insertion in database dialog table. Best Regards.

1 0

using set_body_multipart and append_body_part on on_reply routes
by david 08 Nov '16

08 Nov '16

hello all i'm using kam version 4.4.1 and i would like to generate an ISUP part on the 1xx,2xx responses to a remote carrier since the fsw i send the calls to cannot handle ISUP SDP. despite seeing in the doc that those functions cannot be used there and only in REQUEST_ROUTE, FAILURE_ROUTE, BRANCH_ROUTE, i tried and it works.... partially, because it seems it doesnt create the last boundary and it creates a next boundary with an empty "Encapsulated multipart part: " this the the route i call from onreply_route route[SIPISUP] { set_body_multipart(); msg_apply_changes(); $var(isup) = "\x01\x12\x49\x01\x0a\x03\x02\x0a\x08\x84\x90\x33\x41 \x72\x17\x01\x06\x0a\x08\x04\x13\x93\x70\x21\x73\x23\x10\x08\x01\x01 \x00"; append_body_part("$var(isup)","application/isup;version=itu-t92 +","signal;handling=optional"); msg_apply_changes(); sipt_destination($rU, 31, 4); sipt_set_calling($fU, 4, 0, 3); msg_apply_changes(); } am i doing something wrong? how can i set this to end the SDP with last boundary? thanks a lot and regards david

2 3

Re: [SR-Users] kamailio + farm of Elastix + rtp direct
by Kato 08 Nov '16

08 Nov '16

Any comments on this please? Thanks! De: Kato [mailto:elkato@gmail.com] Enviado el: domingo, 6 de noviembre de 2016 11:29 a. m. Para: 'sr-users(a)lists.sip-router.org' <sr-users(a)lists.sip-router.org> Asunto: kamailio + farm of Elastix + rtp direct Hi, I'm just starting with Kamailio, and trying to validate a possible solution design. I started offering hosted PBXs (FreePBX or Elastix as customer wish), each one with a public IP, connected with SIP trunks from Net2Phone provider. RTP from/to customers calls is going in and out through PBX, and not directly customer <----> sip trunk provider. Billing is provided by SIP trunk provider, so with just CDR from freepbx/Elastix is enough. But, I would like to introduce 2 main changes. First a sip router/proxy in front of PBXs like kamailio, and doing just a basic domain based routing to each PBX. Kamailio with 2 interfaces (1 public, 1 internal) and change PBXs to internal IPs. Second, to know if would be possible to redirect RTP from/to customer-net2phone provider directly, not going through PBXs. Net2phone has public IPs for RTP, customer normally in their homes behind NAT with a SOHO router. Elastix 2.5 using Asterisk 11, and FreePBX using Asterisk 12 (could be 11 or 13 too tbe o reach this). I would like to kamilio be as simple and static as possible, not handling REGISTER or LOCATION. Just to route all traffic based on each domain to each corresponding PBX. My idea will be to pre-provide all subdomains to each internal IP using subdomains like "customer0001.domain.com, customer0002.domain.com." and customers would login as 100(a)customer0001.domain.com <mailto:100@customer0001.domain.com> , 101(a)customer0001.domain.com <mailto:101@customer0001.domain.com> to PBX0001 and 100(a)customer0002.domain.com <mailto:100@customer0002.domain.com> , 101(a)customer0002.domain.com <mailto:101@customer0002.domain.com> to PBX0002 for example. Just using as it was a firewall doing NAT and Asterisk (freepbx/elastix) behinds handles everything as they are doing already. Maybe kamailio could check/filter some malicious packets like I'm doing now with fail2ban on PBXs. Can be this possible? In that case where do I need to setup SIP trunk? On kamailio or on internal PBXs as I have it now? For second part, just to know and understand if would be possible customer "talks" RTP directly to Net2phone SIP trunk provider on doing or receiving external calls. Not going to my DC to save BW. How I could do that? Thanks a lot in advance! Regards, elkato

1 0

Next Kamailio IRC Devel Meeting, Nov 09, 2016
by Daniel-Constantin Mierla 08 Nov '16

08 Nov '16

Hello, I am proposing a new IRC meeting to discuss the current major issues and the plans for next Kamailio release, on Wednesday, Nov 09, 2016, with alternatives for previous or next day. If many developers are not available, we can postpone it to another date in the near future (make proposals if that is the case for you). I created a wiki page for it: * https://www.kamailio.org/wiki/devel/irc-meetings/2016b Add there the topics that you want to be discussed. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 4

Contacts replacing each other when registering
by Sebastian Damm 08 Nov '16

08 Nov '16

Hi, we are running a Registrar server with 4.4.3, the registrar module is configured with the following parameters: modparam("registrar", "max_contacts", 50) modparam("registrar", "use_path", 1) modparam("registrar", "path_mode", 0) modparam("registrar", "path_use_received", 1) modparam("registrar", "received_avp", "$(avp(i:801))") modparam("registrar", "default_expires", 600) modparam("registrar", "min_expires", 60) modparam("registrar", "max_expires", 600) modparam("registrar", "gruu_enabled", 0) A customer has configured 4 phones with the same username. Normally, all 4 phones would be visible in the location database, and incoming calls would fork out to all 4 phones. However, only one of the devices is always online, the others always replace each other on each registration. Client details: * All phones are snom 715 devices with the same firmware version. * All phones sit behind the same router. Registrations: client 1: * Via: SIP/2.0/UDP 192.168.208.91:35812;branch=z9hG4bK-4q0mmxgu6gm7;rport * Contact: <sip:1794388e3@192.168.208.91:35812;line=gicode6v>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:0a14174a-77be-424c-864e-0004137F1C9D>" client 2: * Via: SIP/2.0/UDP 192.168.208.90:41331;branch=z9hG4bK-3cbxdvbmk0di;rport * Contact: <sip:1794388e3@192.168.208.90:41331;line=0j6veso8>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:0a14174a-77be-424c-864e-0004137F1C9D>" client 3: * Via: SIP/2.0/UDP 192.168.208.92:5360;branch=z9hG4bK-4drsogakez94;rport * Contact: <sip:1794388e3@192.168.208.92:5360;line=0n7jfzih>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:0a14174a-77be-424c-864e-0004137F1C9D>" client 4: * Via: SIP/2.0/UDP 192.168.208.93:5260;branch=z9hG4bK-1ci0qia3yxyl;rport * Contact: <sip:1794388e3@192.168.208.93:5260;line=mh9sgxa1>;reg-id=1;q=1.0 Only client 4 is reachable all the time. The other 3 clients replace other registrations. So after client 1 registers, the contacts returned in the 200 OK looks like this: Contact: <sip:1794388e3@192.168.208.91:35812;line=gicode6v>;q=1;expires=600;received="sip:1.2.3.4:6411";+sip.instance="<urn:uuid:0a14174a-77be-424c-864e-0004137F1C9D>";reg-id=1, <sip:1794388e3@192.168.208.93:5260;line=mh9sgxa1>;q=1;expires=303;received="sip:1.2.3.4:8923" And after client 2 registers, the contacts returned in the 200 OK looks like this: Contact: <sip:1794388e3@192.168.208.90:41331;line=0j6veso8>;q=1;expires=600;received="sip:1.2.3.4:49061";+sip.instance="<urn:uuid:0a14174a-77be-424c-864e-0004137F1C9D>";reg-id=1, <sip:1794388e3@192.168.208.93:5260;line=mh9sgxa1>;q=1;expires=301;received="sip:1.2.3.4:8923" So you see, one contact stays the same, the other one gets replaced. Does anybody know what could cause this kind of behavior? I think, it could be GRUU. I see that those three clients all send the same GRUU parameter. From my understanding, this UUID should be different from client to client. I don't know whether this is a bug in the snom firmware always sending the same GRUU UUID or whether som "intelligent" SIP aware router is putting this in the Contact header. However, since we have disabled gruu for the registrar module (see above), I think, Kamailio should ignore it. Ideas? Thanks in advance. Best Regards, Sebastian

2 4

TCP Keepalive for NAT'd Clients
by Colin Morelli 08 Nov '16

08 Nov '16

Hey all, Looking to figure out the best way to allow TCP connections to stay alive for NAT'd clients, however, to protect against people just opening TCP connections to the server, I'm hoping to only keepalive TCP connections for connections that have sent an INVITE and received a 200 OK. Does this type of per-socket connection lifetime setting exist? For a bit more background: this is specifically for clients that are using the TCP connection for an INVITE. They don't register with the server, and as a result there's no active flow timer for their connection. Is there a way to enable this type of behavior without registering? Best, Colin

2 2

How to get Destination port UDP
by andrzej.ciupek-asterisk.edu.pl 08 Nov '16

08 Nov '16

Hello I am using Kamailio to collect data about active calls. I write INVITEs to database, and delete it on BYE or any REL. I would like to write source and destination port too, but don't know how to get destination port. I have tried variables: $dp, $rp, $Rp, $op But when I don't have port in URI like INVITE IP:PORT, $rp have default value of 5060. I would like to get real UDP port, that INVITE does to. Is it possible using PSEUDO VARIABLES? Greetings Andrzej

2 1

Out of memory in UB 210: OOM killed process 12261 (kamailio) score 0 vm:1614768kB, rss:280200kB, swap:131408kB
by Jurijs Ivolga 08 Nov '16

08 Nov '16

Hi, Our Kamailio server is crashing once per week, with following error: Oct 1 06:25:06 kamailio kernel: [26982632.803789] Out of memory in UB 210: OOM killed process 12261 (kamailio) score 0 vm:1614768kB, rss:280200kB, swap:131408kB Core dump was never created, probably it is because of my environment, but I will try to get it. Server constantly eats memory, maybe some kind of memory leak? Any help is highly appreciated! Jurijs

3 25

NER and ASR
by Gholamreza Sabery 07 Nov '16

07 Nov '16

Are there any real data about number of concurrent calls, number of on-line users, NER (Network Effectiveness Ratio) and ASR (Answer Seizure Ratio) of a real VoIP system? I want this data for making comparisons. Can anyone provide a reference, article, etc..

3 4

cdr on remote POSTGRES server
by beer Ll 07 Nov '16

07 Nov '16

Hi all I have a kamailio server configured with accounting on a separate Postgres SQL server by an internal network with acc module and db_postgres module Everything works good but what can I do if I have problem on the internal network or on the postgres server ? Is it possible cache or save the data if the insert data is failed ?

2 1

fnmatch
by Slava Bendersky 07 Nov '16

07 Nov '16

Hello Everyone, I trying fnmatch to find which sip status is returned by B2BUA. But some reason it doesn't work. Nov 6 17:02:15 cavprx00 kamailio: 2(3747) ERROR: *** cfgtrace:onreply_route=[MESSAGE_FORWARD] c=[/etc/kamailio/kamailio-asterisk.cfg] l=456 a=26 n=xlog Nov 6 17:02:15 cavprx00 kamailio: 2(3747) INFO: <script>: Message 1 Accepted by B2BUA --> 202 with Accepted Nov 6 17:02:15 cavprx00 kamailio: 2(3747) ERROR: *** cfgtrace:onreply_route=[MESSAGE_FORWARD] c=[/etc/kamailio/kamailio-asterisk.cfg] l=469 a=16 n=if Nov 6 17:02:15 cavprx00 kamailio: 2(3747) ERROR: *** cfgtrace:onreply_route=[MESSAGE_FORWARD] c=[/etc/kamailio/kamailio-asterisk.cfg] l=457 a=26 n=fnmatch Nov 6 17:02:15 cavprx00 kamailio: 2(3747) ERROR: *** cfgtrace:onreply_route=[DEFAULT_ROUTE] c=[/etc/kamailio/kamailio-asterisk.cfg] l=370 a=2 n=return if(fnmatch("$rs", "200") && (fnmatch("$rr", "Accepted"))) { xlog("L_INFO", "Message Accepted by B2BUA --> $rs with $rr \n"); $var(tu) = $tu; xlog("L_INFO", "New To --> $var(tu)\n"); } Is this fnmatch correct for this operation ? Slava.

3 2

kamailio + farm of Elastix + rtp direct
by Kato 06 Nov '16

06 Nov '16

Hi, I'm just starting with Kamailio, and trying to validate a possible solution design. I started offering hosted PBXs (FreePBX or Elastix as customer wish), each one with a public IP, connected with SIP trunks from Net2Phone provider. RTP from/to customers calls is going in and out through PBX, and not directly customer <----> sip trunk provider. Billing is provided by SIP trunk provider, so with just CDR from freepbx/Elastix is enough. But, I would like to introduce 2 main changes. First a sip router/proxy in front of PBXs like kamailio, and doing just a basic domain based routing to each PBX. Kamailio with 2 interfaces (1 public, 1 internal) and change PBXs to internal IPs. Second, to know if would be possible to redirect RTP from/to customer-net2phone provider directly, not going through PBXs. Net2phone has public IPs for RTP, customer normally in their homes behind NAT with a SOHO router. Elastix 2.5 using Asterisk 11, and FreePBX using Asterisk 12 (could be 11 or 13 too tbe o reach this). I would like to kamilio be as simple and static as possible, not handling REGISTER or LOCATION. Just to route all traffic based on each domain to each corresponding PBX. My idea will be to pre-provide all subdomains to each internal IP using subdomains like "customer0001.domain.com, customer0002.domain.com." and customers would login as 100(a)customer0001.domain.com <mailto:100@customer0001.domain.com> , 101(a)customer0001.domain.com <mailto:101@customer0001.domain.com> to PBX0001 and 100(a)customer0002.domain.com <mailto:100@customer0002.domain.com> , 101(a)customer0002.domain.com <mailto:101@customer0002.domain.com> to PBX0002 for example. Just using as it was a firewall doing NAT and Asterisk (freepbx/elastix) behinds handles everything as they are doing already. Maybe kamailio could check/filter some malicious packets like I'm doing now with fail2ban on PBXs. Can be this possible? In that case where do I need to setup SIP trunk? On kamailio or on internal PBXs as I have it now? For second part, just to know and understand if would be possible customer "talks" RTP directly to Net2phone SIP trunk provider on doing or receiving external calls. Not going to my DC to save BW. How I could do that? Thanks a lot in advance! Regards, elkato

1 0

Kamailio DB lookup
by Gonzalo Gasca Meza 06 Nov '16

06 Nov '16

Hi Kamailions, Kamailio 4.3 will receive calls (SIP TLS) for users in this domain: *sip.parzee.io <http://sip.parzee.io>. *Example sip:gonzalo@sip.parzee.io When call is received by Kamailio I need to overwrite R-URI and transport. I should keep a mapping per user. ( gonzalo -> gonzalo58) Example. sip:gonzalo58@test.external.com;transport=tls I started looking into *dbaliases* table in LOCATION. #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) route(SIPOUT); #!endif In dbaliases table I see 4 columns. Is the following configuration correct? *alias_username*: gonzalo58 *alias_domain*: test.external.com;transport=tls *username*: gonzalo *domain*: sip.parzee.io Thanks -Gonzalo

4 5

Malformed To header, or?
by Tobias 04 Nov '16

04 Nov '16

Hi! Having some issues with Kamailio 4.3 reporting a malformed header, but I cannot seem to figure out what's wrong. The problem seen is: Oct 27 13:48:56 /usr/sbin/kamailio[13765]: ERROR: <core> [parser/parse_addr_spec.c:678]: parse_addr_spec(): ERROR: parse_to : unexpected char [#015] in status 6: <<+123456789 <sip:123456789@127.0.0.1;tag=as4aa27bd0>> . Oct 27 13:48:56 /usr/sbin/kamailio[13765]: ERROR: <core> [parser/msg_parser.c:165]: get_hdr_field(): ERROR: get_hdr_field: bad to header Oct 27 13:48:56 /usr/sbin/kamailio[13765]: INFO: <core> [parser/msg_parser.c:338]: parse_headers(): ERROR: bad header field [To: +123456789 <si] Oct 27 13:48:56 /usr/sbin/kamailio[13765]: ERROR: tm [t_lookup.c:1050]: t_check_msg(): ERROR: reply cannot be parsed Oct 27 13:48:56 /usr/sbin/kamailio[13765]: ERROR: <core> [parser/parse_addr_spec.c:678]: parse_addr_spec(): ERROR: parse_to : unexpected char [#015] in status 6: <<+123456789 <sip:123456789@127.0.0.1;tag=as4aa27bd0>> . Oct 27 13:48:56 /usr/sbin/kamailio[13765]: ERROR: <core> [parser/msg_parser.c:165]: get_hdr_field(): ERROR: get_hdr_field: bad to header Oct 27 13:48:56 /usr/sbin/kamailio[13765]: INFO: <core> [parser/msg_parser.c:338]: parse_headers(): ERROR: bad header field [To: +123456789 <si] Oct 27 13:48:56 /usr/sbin/kamailio[13765]: ERROR: <core> [msg_translator.c:1457]: adjust_clen(): error parsing content-length Oct 27 13:48:56 /usr/sbin/kamailio[13765]: ERROR: <core> [msg_translator.c:2214]: generate_res_buf_from_sip_res(): error while adjusting Content-Length Oct 27 13:48:56 /usr/sbin/kamailio[13765]: ERROR: <core> [forward.c:760]: do_forward_reply(): building failed And the To header seen at this point is: To: +123456789 <sip:123456789@127.0.0.1>;tag=as24ed5606. version: kamailio 4.3.4 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled with gcc 4.7.2 I obviously understand Kamailio views this To header as malformed, but what I cannot understand is why? Is it as simple as the number +123456789 is not enclosed with ", i.e. it should be sent as "+123456789"? A similar issue seen, which for some reason doesn't create an ERROR, is when Kamailio receives a reINVITE looking like this which it's supposed to proxy: To: +123456789 <sip:123456789@127.0.0.1>;tag=SD1sf9001--45026-530b969-77c7543f-530b969. which is rewritten into: To: +123456789 <sip:123456789@127*0-)'.:/)&7'...0;user=phone>;tag=SD1sf9001--45026-530b969-77c7543f-530b969. and sent out.. Thanks for your help, /Tobias

3 13

Configuring/Integrating Kamailio Open IMS with FHoSS
by Olvera, Victor 04 Nov '16

04 Nov '16

How does one configure Kamailio Open IMS and FHoSS? In other words, how does one configure IMS and FHoSS so that when soft calls are made they must be processed by the FHoSS before they go through the Open IMS server? Great regards, Victor Olvera

8 13

nathelper not changes IP
by Yuriy Gorlichenko 03 Nov '16

03 Nov '16

I trying to use fix_nated_sdp with 0x02 and 0x08 flags for changing ip addres and the SDP body part like fix_nated_sdp(10,"1.2.3.4") But for now it beaks SDP At the output i see next o=- 7300689428214760503 2 IN IP4 1.1.1.1 c=IN IP4 1.1.1.1 1.2.3.41.2.3.4 <- is just a line that added after fix_nated_sdp atthe end of SDP body kamailio -v version: kamailio 4.4.3 (x86_64/linux) e91aec flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: e91aec compiled on 07:26:14 Oct 18 2016 with gcc 4.9.2

3 4

Cannot send request to OpenSIPS
by Laura 02 Nov '16

02 Nov '16

Dear support, on my Kamailio server (version: kamailio 4.3.6 (x86_64/linux) ) i see lot of these messages.. Nov 2 11:24:42 vm-test media-dispatcher[25540]: error: cannot write request to `/var/run/opensips/socket': No such file or directory Nov 2 11:24:42 vm-test media-dispatcher[25540]: error: failed to end dialog: Cannot send request to OpenSIPS I have only configured Disptachet (mediaproxy) for write on db.. Any idea ? Thanks Laura

2 4

diff between location and kamctl ul show
by Никитенко Виталий 02 Nov '16

02 Nov '16

Hi all! I have kamailio 4.2.3 with options modparam ( "usrloc", "db_url", DBURL) modparam ( "usrloc", "db_mode", 2) But not all registred users displayed in table "location". When viewed "kamctl ul show" displays Domain :: location table = 1024 records = 1172 max_slot = 5 But "location" table has 651 rows. Why not all users saved to mysql table? Thank!

3 2

Kamailio TLS client config selection
by Sergey Basov 02 Nov '16

02 Nov '16

Hi Daniel. After some try to configure kamailio 4.4.3 to act as SPI TLS client for Cisco SIP TLS gateways I have found one issue. If I do client configuration for tls [client:10.1.23.19:5061] verify_certificate = yes ca_list = /etc/kamailio/CAs/ca1.pem [client:10.1.23.29:5061] verify_certificate = yes ca_list = /etc/kamailio/CAs/ca2.pem [client:default] verify_certificate = no require_certificate = no Kamailo always do default profile selection (I do configuration without server_name or server_id, with it kamailio works fine but there are some troubles to make selection of this parameters from config script, I need additional checks and queries) after some research in tls module source code I have added some debug information in file tls_server.c: if (c->flags & F_CONN_PASSIVE) { state=S_TLS_ACCEPTING; dom = tls_lookup_cfg(cfg, TLS_DOMAIN_SRV, &c->rcv.dst_ip, c->rcv.dst_port, 0, 0); } else { state=S_TLS_CONNECTING; sname = tls_get_connect_server_name(); srvid = tls_get_connect_server_id(); // ------------------------------------------------------------- DBG("Entered client config loockup (c->rcv.dst_port %d)\n", c->rcv.dst_port); DBG("Entered client config loockup (&c->rcv.dst_ip %s)\n", ip_addr2a(&c->rcv.dst_ip)); DBG("Entered client config loockup (c->rcv.src_port %d)\n", c->rcv.src_port); DBG("Entered client config loockup (&c->rcv.src_ip %s)\n", ip_addr2a(&c->rcv.src_ip)); // ------------------------------------------------------------- dom = tls_lookup_cfg(cfg, TLS_DOMAIN_CLI, &c->rcv.dst_ip, c->rcv.dst_port, sname, srvid); } After making Oct 26 09:23:56 sip1 /usr/sbin/kamailio[20712]: DEBUG: <core> [parser/msg_parser.c:597]: parse_msg(): method: <INVITE> Oct 26 09:23:56 sip1 /usr/sbin/kamailio[20712]: DEBUG: <core> [parser/msg_parser.c:599]: parse_msg(): uri: <sip:9098@10.1.23.19:5061;transport=TLS> Oct 26 09:23:56 sip1 /usr/sbin/kamailio[20712]: DEBUG: <core> [parser/msg_parser.c:601]: parse_msg(): version: <SIP/2.0> I see Oct 26 09:23:56 sip1 /usr/sbin/kamailio[20712]: DEBUG: <core> [ip_addr.c:229]: print_ip(): tcpconn_new: new tcp connection: 10.1.23.19 Oct 26 09:23:56 sip1 /usr/sbin/kamailio[20712]: DEBUG: <core> [tcp_main.c:985]: tcpconn_new(): on port 5061, type 3 Oct 26 09:23:56 sip1 /usr/sbin/kamailio[20712]: DEBUG: <core> [tcp_main.c:1295]: tcpconn_add(): hashes: 1394:0:0, 1 Oct 26 09:23:56 sip1 /usr/sbin/kamailio[20712]: DEBUG: tls [tls_server.c:197]: tls_complete_init(): completing tls connection initialization Oct 26 09:23:56 sip1 /usr/sbin/kamailio[20712]: DEBUG: tls [tls_server.c:160]: tls_get_connect_server_name(): xavp with outbound server name not found Oct 26 09:23:56 sip1 /usr/sbin/kamailio[20712]: DEBUG: tls [tls_server.c:140]: tls_get_connect_server_id(): xavp with outbound server id not found Oct 26 09:23:56 sip1 /usr/sbin/kamailio[20712]: DEBUG: tls [tls_server.c:219]: tls_complete_init(): Entered client config loockup (c->rcv.dst_port 40123) Oct 26 09:23:56 sip1 /usr/sbin/kamailio[20712]: DEBUG: tls [tls_server.c:220]: tls_complete_init(): Entered client config loockup (&c->rcv.dst_ip 10.1.23.23) Oct 26 09:23:56 sip1 /usr/sbin/kamailio[20712]: DEBUG: tls [tls_server.c:221]: tls_complete_init(): Entered client config loockup (c->rcv.src_port 5061) Oct 26 09:23:56 sip1 /usr/sbin/kamailio[20712]: DEBUG: tls [tls_server.c:222]: tls_complete_init(): Entered client config loockup (&c->rcv.src_ip 10.1.23.19) Where: &c->rcv.dst_ip 10.1.23.23 - it is my local kamailio tls socket ip address to make tls connect from c->rcv.dst_port 40123 - it is my local kamailio tls socket port &c->rcv.src_ip 10.1.23.19 - ip of my TLS device to make tls connection to c->rcv.src_port 5061 - port of my TLS device to make tls connection to so if I change line dom = tls_lookup_cfg(cfg, TLS_DOMAIN_CLI, &c->rcv.dst_ip, c->rcv.dst_port, sname, srvid); to dom = tls_lookup_cfg(cfg, TLS_DOMAIN_CLI, &c->rcv.src_ip, c->rcv.src_port, sname, srvid); I got correct client domain selection Oct 26 09:33:56 sip1 /usr/sbin/kamailio[20712]: DEBUG: tls [tls_server.c:233]: tls_complete_init(): Using initial TLS domain TLSc<10.1.23.19:5061> (dom 0x7fd2eefa3d68 ctx 0x7fd2ef7e70a8 sn []) Oct 26 09:33:56 sip1 /usr/sbin/kamailio[20712]: DEBUG: tls [tls_domain.c:703]: sr_ssl_ctx_info_callback(): SSL handshake started Can you look at this code? Thank you in advance. -- Best regards, Sergey Basov e-mail: sergey.v.basov(a)gmail.com tel: (+38067) 403-62-54

2 3

Kamailio + IMS + sipml5 call to softphone
by Serhat Guler 01 Nov '16

01 Nov '16

Hi, I have a setup as follows: IMS enabled on Kamailio and whereas websockets are enabled for PCSCF for webrtc calls. Calls(both audio and video) between to sipml5 clients using firefox web browser is possible. The session is setup for the calls from sipml5 to Mercuro, but then there isn't audio flow as the codecs are not compatible. Now I want to test it with Boghe which supports G.722, PCMA, PCMU, and OPUS codecs as firefox but this time the session isn't being setup. Boghe replies with "Reason: SIP; cause=488; text="Bad content" " I have seen a similar issue has been mentioned here: https://github.com/c00lz3r0/boghe/issues/157 but the initial invite request from sipml5 does have the SDP with media attributes.

3 8

ACC module and BYE attacks
by Gholamreza Sabery 01 Nov '16

01 Nov '16

If you enable acc module to log CDR data into a database a user can send an infinite number of BYE requests to the server and all of them will be logged into the database as "481 Call/transaction does not exist"! How can we prevent this?

3 5

/usr/sbin/kamailio - unresolvable destination error
by Ginti Saurabh 01 Nov '16

01 Nov '16

Hi, Please help to sort the problem. Getting "Unresolvable destination" error in /var/log/messages.

2 1

Terminating PCSCF unresolvable request URI (Kamailio IMS Websockets)
by Serhat Guler 01 Nov '16

01 Nov '16

Dear all, I am trying to make a call between 2 webrtc clients(sipml5) using my kamailio IMS setup. The invite request fails at the terminating PCSCF with error code "Status: 478 Unresolvable destination (478/TM)". I have traced the packages and saw that the package that arrives at the terminating PCSCF has an unresolvable request URI due to the websocket connection. The package being sent from terminating ICSCF to SCSCF has the "Request-URI: sip:bob@net1.test" The package from terminating SCSCF to terminating PCSCF has the "Request-URI: sip:bob@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws" I guess the terminating SCSCF replaces the request URI with the one it stored during the registration process, but the terminating PCSCF cannot resolve the host name&port number; therefore, it reports back "Unresolvable destination". Is that really the case happening in here ? How can I get over this problem ? Any ideas are welcome. Cheers, Serhat

2 1

Server error for SIP client
by Hermann Norpois 01 Nov '16

01 Nov '16

Hello, if my SIP client tries to connect ... a server error occured (see below). Could you please help me to figure out the reason? Kamailio is installed on an amazon instance with an elastic IP (52.208.249.214, hostname: synth.ddns.net). Thanks Hermann ubuntu@ip-172-31-32-10:~sudo /etc/init.d/kamailio restart * Stopping Kamailio SIP Server: kamailio [ OK ] * Starting Kamailio SIP Server: kamailio loading modules under config path: /usr/lib/x86_64-linux-gnu/kamailio/modules/ Listening on udp: 127.0.0.1:5060 udp: 172.31.32.10:5060 tcp: 127.0.0.1:5060 tcp: 172.31.32.10:5060 Aliases: tcp: ip-172-31-32-10.eu-west-1.compute.internal:5060 tcp: localhost:5060 udp: ip-172-31-32-10.eu-west-1.compute.internal:5060 udp: localhost:5060 *: synth.ddns.net:* ubuntu@ip-172-31-32-10:~$ kamctl db show subscriber +----+----------+----------------+----------+---------------+----------------------------------+----------------------------------+------+ | id | username | domain | password | email_address | ha1 | ha1b | rpid | +----+----------+----------------+----------+---------------+----------------------------------+----------------------------------+------+ | 1 | Johanna | synth.ddns.net | 12345 | | 4fa26cedd456d7105c01a48742e71e42 | 1e249ac341d95701f855fce18bec0fba | NULL | ubuntu@ip-172-31-32-10:~$ sudo ngrep -d any -qt -W byline . port 5060 Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.5ad567803b13610279818912da2fcd81.0. Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.6a2baefed6cf4f6b6d34c7563d829e44.0. Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.6d74cec9e5f5e46dd9cd67338143127f.0. Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.a3a37d6bd0cd5d68fc786568f587f06e.0. Via: SIP/2.0/UDP 188.174.185.176:5060 ;received=188.174.185.176;rport=5060;branch=z9hG4bKPjAgF18hfvv20KOQ9H8G-fqfIPMjjcu9wq. From: <sip:Johanna@synth.ddns.net>;tag=vzlAJNNNES67ir6nsLMjvyFh5D0Thy6s. To: <sip:Johanna@synth.ddns.net>;tag=b27e1a1d33761e85846fc98f5f3a7e58.3300. Call-ID: 9Uj4TSeQAVwaMs3UHf8UKIhuoA4xBEZt. CSeq: 39586 REGISTER. Server: kamailio (4.3.6 (x86_64/linux)). Content-Length: 0. . U 2016/10/20 13:44:33.727161 52.208.249.214:5060 -> 172.31.32.10:5060 SIP/2.0 500 I'm terribly sorry, server error occurred (1/SL). Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.2605d82fd11bd8f788cfad582884eb6d.0. Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.e425ad98e0436abc4661d1452a49670d.0. Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.2d047656fb9df0edfccd4a40d20237b7.0. Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.f54ab9c35b22814b15acbe16bcae387f.0. Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.db5960988332c2f4899a5e1c2b57e3c6.0. Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.62f1191bd5af8480698ce118984536c0.0. Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.5ad567803b13610279818912da2fcd81.0. Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.6a2baefed6cf4f6b6d34c7563d829e44.0. Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.6d74cec9e5f5e46dd9cd67338143127f.0. Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.a3a37d6bd0cd5d68fc786568f587f06e.0. Via: SIP/2.0/UDP 188.174.185.176:5060 ;received=188.174.185.176;rport=5060;branch=z9hG4bKPjAgF18hfvv20KOQ9H8G-fqfIPMjjcu9wq. From: <sip:Johanna@synth.ddns.net>;tag=vzlAJNNNES67ir6nsLMjvyFh5D0Thy6s. To: <sip:Johanna@synth.ddns.net>;tag=b27e1a1d33761e85846fc98f5f3a7e58.3300. Call-ID: 9Uj4TSeQAVwaMs3UHf8UKIhuoA4xBEZt. CSeq: 39586 REGISTER. Server: kamailio (4.3.6 (x86_64/linux)). Content-Length: 0. 2) My logfile /var/log/kamailio/kamailio.log is empty though /etc/rsyslog.d/kamailio.conf exists with the entry: local0.* -/var/log/kamailio/kamailio.log And in /etc/kamailio/kamailio.conf: log_facility=LOG_LOCAL0 Why is the logfile empty? Thanks Herman

2 1

Store and add custom variables for sip registrations
by Markus Bönke 01 Nov '16

01 Nov '16

Hallo all, we want to use kamailio as registrar server in front of freeswitch. When a user registers we need to lookup and store some user variables. To every INVITE that is received from a registered user we want to add the variables as x sip headers before we relay them to freeswitch. The question is what is the best way to implement this? We tried to store the variables in a htable on register and then lookup and attach the variables on every INVITE from there. Is this the correct approach, one issue we see is that htable is not writing new inserts directly to the database. I’ve also seen that there is a table location_atts in the kamailio db which looks like that it is intended to store custom key value pairs for registrations, but the seems to be no module which writes to that table, or is there one? Thanks for any help or suggestions Markus

2 1

TSILO module use failure
by Igor Olhovskiy 31 Oct '16

31 Oct '16

Hi! I’m trying to implement scheme much like Federico Cabiddu is described here https://www.youtube.com/watch?v=4XIrR9bwUkM Done a bit modification to his script, cause it’s not fully correct on 4.4 version So, parts of config looks like …. route[REGISTRAR] { … if (!save("location")) { sl_reply_error(); } else { route(PUSHJOIN); } exit; …. } route[LOCATION] { if (!lookup("location")) { …… if (is_method("INVITE")) { send_reply("100", "Trying"); route(SUSPEND); } } else { if (is_method("INVITE")) { if (t_newtran()) { ts_store("sip:$tU@$fd"); } $sht(vtp=>stored::$rU) = 1; } } route(RELAY); } ….. route[SUSPEND] { if(!t_suspend()) { send_reply("501", "Unknown destination"); exit; } $sht(vtp=>join::$rU) = "" + $T(id_index) + ":" + $T(id_label); } # append branches or resume the transaction route[PUSHJOIN] { … #if was suspended - recover, if not - ts_append ….. ts_append(«location", "sip:$tU@$fd"); return; … t_continue("$var(id_index)", "$var(id_label)", "INVRESUME"); } # lookup and relay after resuming transaction route[INVRESUME] { lookup("location"); t_relay(); ts_store("sip:$tU@$fd"); $sht(vtp=>stored::$rU) = 1; } Branches on ts_store are saved in format (sip:username@domain) and also in this format they are added on ts_append. All regarding to routes working ok, except on ts_append I got ERROR: tm [t_append_branches.c:168]: t_append_branches(): ERROR: t_append_branch: failure to add branches (-1) Also, it’s a mixed environment, like wss + udp + tcp. Any idea why? Thanks. -- Best regards, Igor

2 3

Re: [SR-Users] PCSCF cannot create via header for sipml5 ACK package
by Serhat Guler 30 Oct '16

30 Oct '16

Hi all, I am still stuck with the ACK message not being forwarded by the originating PCSCF. Any advice would be great. Thanks, Serhat On 24 October 2016 at 21:00, Serhat Guler <srtguler(a)gmail.com> wrote: > Hi Daniel, > > I am using only record_route() without any parameters. I do not have a > proper computer atm to draw the network diagram, but I can tell you shortly > about the network setup. > > I have only enabled websockets for the pcscf to allow ws and wss > connections. In that case there is a ws connection that uses UDP protocol. > This is the ACK to complete the session setup. > > the sipml5 client is configured as follows: > WebSocket Server URL: ws://192.168.0.11:880 > SIP outbound Proxy URL: udp://192.168.0.11:4060 > > Mercuro IMS client: uses UDP port as well: 4060 > > The call is made from sipml5 client. The Mercuro phone rings, and when I > reply the call, 200 OK is sent to sipml5 webrtc client, but the ACK from > sipml5 doesn't pass the PCSCF as I explained in the previous message. > > A part of PCSCF cfg file: > > # Check for Subsequent requests: > if (has_totag()) { > # sequential request withing a dialog should > # take the path determined by record-routing > if (loose_route()) { > if ($route_uri =~ "sip:mo@.*") { > setflag(FLT_MO); > } > if(!isdsturiset()) { > handle_ruri_alias(); > } > # RTP-Relay, if necessary > route(RTPPROXY); > t_relay(); > } else { > if ( is_method("ACK") ) { > if ( t_check_trans() ) { > # no loose-route, but stateful ACK; > # must be an ACK after a 487 > # or e.g. 404 from upstream server > t_relay(); > exit; > } else { > xlog("L_INFO", "ACK without matching transaction ... > ignore and discard!!!!!\n"); > # ACK without matching transaction ... ignore and > discard > exit; > } > } > sl_send_reply("404","Not here"); > } > exit; > } > > Cheers, > Serhat > > > > On 24 October 2016 at 20:18, Daniel-Constantin Mierla <miconda(a)gmail.com> > wrote: > >> Hello, >> >> I haven't noticed the log files, it's ok. >> >> From the Route header, I see that there is a proxy that uses WS: >> >> Route: <sip:mo@192.168.0.11:880;transport=ws;r2=on;lr=on;ftag=GxzKy >> 1nCMEI1mR0RztrB;did=e82.0c3> >> That is the address of the next hop and typically a proxy doesn't use >> websocket connection to another proxy. Can you show a diagram with the sip >> server nodes in your network and what protocols are used between them? >> >> Are you simply use record_route() function, or some other function or >> different parameters to it? >> >> Cheers, >> Daniel >> >> >> On 24/10/16 12:18, Serhat Guler wrote: >> >> Hi Daniel, >> >> Thanks for your reply. I actually attached a log file with debug level 3, >> consisting ACK related messages. If you would like to see more logs, I'll >> send a new log file in the evening. >> >> Cheers, >> Serhat >> >> On 24 October 2016 at 12:13, Daniel-Constantin Mierla <miconda(a)gmail.com> >> wrote: >> >>> Hello, >>> >>> can you get all the log messages for ACK but with debug=3 in the >>> kamailio.cfg? >>> >>> Cheers, >>> Daniel >>> >>> On 23/10/16 22:04, Serhat Guler wrote: >>> >>> Hello, >>> >>> I finally managed to place a call from sipml5 webrtc client to Mercuro >>> IMS client. The phone rings, and when I answer it sends 200 OK to the >>> sipml5 where as sipml5 send back an ACK message which never passes the >>> originating PCSCF. The PCSCF says: >>> >>> 8(3640) WARNING: <core> [msg_translator.c:2729]: via_builder(): TCP/TLS >>> connection (id: 0) for WebSocket could not be found >>> 8(3640) ERROR: <core> [msg_translator.c:1947]: >>> build_req_buf_from_sip_req(): could not create Via header >>> 8(3640) ERROR: <core> [forward.c:548]: forward_request(): building >>> failed >>> >>> I doubt that the WebSocket connection is closed, cause when I terminate >>> the call from Mercuro client a bye request is being sent to the sipml5. >>> >>> The ACK package: >>> >>> ACK sip:alice@192.168.0.10:49794;transport=udp SIP/2. >>> Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9 >>> hG4bKvuly7bmxnN4aqM4zZTIS;rport >>> From: "Bob"<sip:bob@net1.test>;tag=GxzKy1nCMEI1mR0RztrB >>> To: <sip:alice@net1.test>;tag=18823 >>> Contact: "Bob"<sip:bob@df7jal23ls0d.invalid;rtcweb-breaker=no;click2c >>> all=no;transport=ws>;+g.oma.sip-im;language="en,fr" >>> Call-ID: 5a500969-d0fa-14d1-7d0e-8605f4356ca6 >>> CSeq: 3887 ACK >>> Content-Length: >>> Route: <sip:192.168.0.11:4060;lr;sipml5-outbound;transport=udp> >>> Max-Forwards: 69 >>> Route: <sip:mo@192.168.0.11:880;transport=ws;r2=on;lr=on;ftag=GxzKy >>> 1nCMEI1mR0RztrB;did=e82.0c3> >>> Route: <sip:mo@192.168.0.11:4060;r2=on;lr=on;ftag=GxzKy1nCMEI1mR0Rz >>> trB;did=e82.0c3> >>> Route: <sip:mo@192.168.0.11:6060;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;di >>> d=e82.f062> >>> Route: <sip:mt@192.168.0.11:6060;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;di >>> d=e82.f062> >>> Route: <sip:mt@192.168.0.11:4060;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;di >>> d=e82.1c3> >>> User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04 >>> Organization: Doubango Telecom >>> >>> Have been thinking for quite a while, but couldn't really find a reason >>> why it wouldn't add the v,a header. A debug 3 level log file is also >>> attached. >>> >>> Thanks in advance, >>> Serhat >>> >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>> >>> -- >>> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >>> Kamailio Advanced Training, Berlin, Oct 24-26, 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/cg >>> i-bin/mailman/listinfo/sr-users >> >> -- >> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >> Kamailio Advanced Training, Berlin, Oct 24-26, 2016 - http://www.asipto.com >> >> >

2 3

digest authentication based on two password columns
by Jayesh Nambiar 28 Oct '16

28 Oct '16

Hello, I was wondering if it is possible to do a failover in digest authentication. For eg: I get two possible valid password values. Can I do a pv_proxy_authenticate in such a way where if authentication fails, I do a pv_proxy_autenticate again on the different value to check if that succeeds? Is it even allowed in digest authentication mechanism. Just getting opinions. Thanks for any help. - Jayesh

3 3

Re: [SR-Users] fetch contact
by Slava Bendersky 28 Oct '16

28 Oct '16

Hello Daniel, No that exactly the one, Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Friday, 28 October, 2016 07:37:31 Subject: Re: [SR-Users] fetch contact Hello, the contact address is one for attributes in $ulc(...), iirc, it is the =>addr -- see the wiki for pv cookbook for more details. Or are you looking for something else? Cheers, Daniel On 27/10/16 21:43, Slava Bendersky wrote: Hello Everyone, How to fetch contact from location table ? I tried look reg_fetch_contacts(), but I don't specific $ulc for contact. Thanks _______________________________________________ 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 -- 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

1 0

Errors when parsing Diversion header with multiple comma-separated diversions
by Björn Bylander 28 Oct '16

28 Oct '16

Hello! Given a Diversion header like the following: Diversion:"Foo Bar"<sip:+9876543211@127.0.0.101;user=phone>;privacy=off;answered-count=2;re ason=deflection;counter=1;answered,"_ somewhere"<sip:+9876543212@127.0.0.101;user=phone>;privacy=off;answered-coun t=1;reason=deflection;counter=1;answered Kamailio 4.4.3 (and probably earlier versions) emit error messages complaining about the comma character. According to RFC 6044 multiple comma-separated diversions in a single Diversion-header is allowed. Full INVITE as an example: INVITE sip:+1234567890@127.0.0.100:5060;user=phone;transport=tcp SIP/2.0 Via:SIP/2.0/TCP 127.0.0.101;branch=z9hG4bKBroadWorks.-1od4eg8-127.0.0.100V5060-0-82844591-13 54681939-1477634439005- From:<sip:+9876543210@127.0.0.101;user=phone>;tag=1354681939-1477634439005- To:<sip:+1234567890@127.0.0.100:5060;user=phone> Call-ID:BW0800390052810161562691307@127.0.0.101 CSeq:82844591 INVITE Contact:<sip:127.0.0.101:5060;transport=tcp> P-Asserted-Identity:<sip:+9876543210@127.0.0.101;user=phone> Privacy:none Diversion:"Foo Bar"<sip:+9876543211@127.0.0.101;user=phone>;privacy=off;answered-count=2;re ason=deflection;counter=1;answered,"_ somewhere"<sip:+9876543212@127.0.0.101;user=phone>;privacy=off;answered-coun t=1;reason=deflection;counter=1;answered Supported:100rel Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE X-BroadWorks-DNC:network-address="sip:+9876543210@127.0.0.101;user=phone" Accept:application/dtmf-relay,application/media_control+xml,application/sdp, application/x-broadworks-call-center+xml,multipart/mixed Max-Forwards:10 Content-Type:application/sdp Content-Disposition:session;handling=required Content-Length:267 v=0 o=BroadWorks 100442303 1 IN IP4 127.0.0.101 s=- c=IN IP4 127.0.0.102 t=0 0 m=audio 29778 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:20 Example log messages: Oct 28 08:00:39 kam1 /usr/sbin/kamailio[14935]: ERROR: <core> [parser/parse_addr_spec.c:488]: parse_to_param(): ERROR parse_to_param : invalid character ',' in status 21: <<;privacy=off;answered-count=2;reason=deflection;counter=1;answered>> Oct 28 08:00:39 kam1 /usr/sbin/kamailio[14935]: ERROR: <core> [parser/parse_addr_spec.c:254]: parse_to_param(): ERROR: parse_to_param :unexpected char ["] in status 21: <<;privacy=off;answered-count=2;reason=deflection;counter=1;answered,>> . Oct 28 08:00:39 kam1 /usr/sbin/kamailio[14935]: ERROR: <core> [parser/parse_addr_spec.c:906]: parse_addr_spec(): ERROR: parse_to: invalid To - unexpected end of header in state 21 Oct 28 08:00:39 kam1 /usr/sbin/kamailio[14935]: ERROR: <core> [parser/parse_diversion.c:69]: parse_diversion_header(): ERROR:parse_diversion_header: bad diversion header Oct 28 08:00:39 kam1 /usr/sbin/kamailio[14935]: ERROR: <core> [lvalue.c:345]: lval_pvar_assign(): non existing right pvar Oct 28 08:00:39 kam1 /usr/sbin/kamailio[14935]: ERROR: <core> [lvalue.c:405]: lval_assign(): assignment failed at pos: (562,21-562,35) The code on line 562 is $avp(div_user) = $(di{uri.user});. Can this be handled in any way by Kamailio without changing the parser? Best regards, Björn Bylander

2 1

fetch contact
by Slava Bendersky 28 Oct '16

28 Oct '16

Hello Everyone, How to fetch contact from location table ? I tried look reg_fetch_contacts(), but I don't specific $ulc for contact. Thanks

2 1

logfile is empty
by Hermann Norpois 27 Oct '16

27 Oct '16

my logfile /var/log/kamailio/kamailio.log is empty. 1) /etc/rsyslog.d/kamailio.cfg exists with the entry: log_facility=LOG_LOCAL0 2) /etc/rsyslog.con has the entry: # # don't log messages with LOG_LOCAL0 in /var/log/syslog anymore *.*;auth,authpriv.none,local0.none -/var/log/syslog # # log messages with LOG_LOCAL0 in /var/log/kamailio.log local0.* -/var/log/kamailio.log # local0.* -/var/log/kamailio/kamailio/kamailio.log Alternatively I edited /etc/rsyslog.d/kamailio.conf that should be used by rsylog.conf 3) /etc/rsyslog.d/kamailio.conf exists with the entry: local0.* -/var/log/kamailio/kamailio.log I am working with ubuntu on an amazon instance. Why is the logfile empty? Thanks Herman

3 2

How do I see the full SDP?
by Martin Taschen 27 Oct '16

27 Oct '16

Hello, kamailio seems to work fine. After registration I would like to see the complete SDP of an user. . How is this done? Thanks Martin Taschen

2 1

Re: [SR-Users] sr-users Digest, Vol 137, Issue 25
by surendra tiwari 27 Oct '16

27 Oct '16

i used the cnxcc module its successfully terminated the call. but if you want i can help you. On Thu, Oct 27, 2016 at 12:58 AM, <sr-users-request(a)lists.sip-router.org> wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > > You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. Re: cnxcc doesn't terminate calls (Igor Potjevlesch) > 2. logfile is empty (Hermann Norpois) > 3. Re: Out of memory in UB 210: OOM killed process 12261 > (kamailio) score 0 vm:1614768kB, rss:280200kB, swap:131408kB > (Jurijs Ivolga) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 26 Oct 2016 12:48:29 +0200 > From: "Igor Potjevlesch" <igor.potjevlesch(a)gmail.com> > To: "'Kamailio $SER$ - Users Mailing List'" > <sr-users(a)lists.sip-router.org> > Subject: Re: [SR-Users] cnxcc doesn't terminate calls > Message-ID: <009c01d22f76$7d0c9a30$7725ce90$(a)gmail.com> > Content-Type: text/plain; charset="utf-8" > > Hi, > > > > Yep, the dialog is created. I put the dialog into DB and see it. > > Nothing appears in the logs when the timer is reached. > > > > I succeed to do the same behaviour with dialog module only and > dlg_set_timeout. Looks to do why I want. > > > > I'm a bit frustrating about cnxcc because I really don't understand why > nothing happen after max_time, but, anyway, it works with dialog only. > > > > Regards, > > > > Igor. > > > > De : sr-users [mailto:sr-users-bounces@lists.sip-router.org] De la part > de Arsen > Envoyé : mercredi 26 octobre 2016 11:15 > À : Kamailio (SER) - Users Mailing List <sr-users(a)lists.sip-router.org> > Objet : Re: [SR-Users] cnxcc doesn't terminate calls > > > > Hi Igor, > > > > Make sure that you create a dialog for this call and $dlg_var(key) is > available (This pseudo-variable will be available only for subsequential > requests after doing loose_route().) > > > > Also check kamailio logs maybe you try to terminate call for wrong > customer id. > > > > On Wed, Oct 26, 2016 at 10:52 AM, Igor Potjevlesch < > igor.potjevlesch(a)gmail.com <mailto:igor.potjevlesch@gmail.com> > wrote: > > Hi, > > > > I'm testing cnxcc module in order to make a simple call duration > limitation. > > My understanding is to use the function like this : > > > > $dlg_var(customer_id) = "customer-" + $fU; > > $var(max_time) = 5; > > > > if (!cnxcc_set_max_time("$dlg_var(customer_id)", > "$var(max_time)")) > > > > Unfortunately, after 5 seconds, nothing happen. I don't even see an error > from dialog module. > > > > Anyone has a better understanding of the way the module runs? Thank you! > > > > Regards, > > > > Igor. > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > > > > > > -- > > Regards, > Arsen. > >

1 0

cnxcc doesn't terminate calls
by Igor Potjevlesch 26 Oct '16

26 Oct '16

Hi, I'm testing cnxcc module in order to make a simple call duration limitation. My understanding is to use the function like this : $dlg_var(customer_id) = "customer-" + $fU; $var(max_time) = 5; if (!cnxcc_set_max_time("$dlg_var(customer_id)", "$var(max_time)")) Unfortunately, after 5 seconds, nothing happen. I don't even see an error from dialog module. Anyone has a better understanding of the way the module runs? Thank you! Regards, Igor.

2 2

app_perl modifies Via header?
by Igor Potjevlesch 25 Oct '16

25 Oct '16

Hello, I try to use app_perl and cnxcc module. As soon as I load the module like this: # --------- app_perl params ---- modparam("app_perl", "filename", "/usr/local/etc/kamailio/AAA/aaa-script.pl") modparam("app_perl", "modpath", "/usr/local/lib64/kamailio/perl/") The Via header is modified for each calls: /usr/local/sbin/kamailio[21064]: INFO: <core> [parser/msg_parser.c:355]: parse_headers(): ERROR: bad header field [Via:ASIP/2.0/UDP 91.] /usr/local/sbin/kamailio[21064]: ERROR: <core> [parser/parse_via.c:2048]: parse_via(): ERROR: parse_via: bad char <A> on state 100 What could explain this behaviour? Regards, Igor.

2 4

Kamailio SIP TLS issue
by Sergey Basov 25 Oct '16

25 Oct '16

Hi All I have some strange behavior of kamailio with TLS. I have configured second IP addres on server, added it to tls listener, and tls.cfg file. But when I try to connect using openssl s_client -showcerts -connect 10.1.23.33:5061 -tls1 -state and openssl s_client -showcerts -connect 10.1.23.23:5061 -tls1 -state I see same certificates (sip2 my config samples are bellow) if I make changes in port number (for ip 10.1.23.33 set port 5091 in both config parts) - I see correct certificates. Does anyone have this problem? Thanks in advance. ----- listen section ---- listen=tls:10.1.23.23:5061 listen=tls:10.1.23.33:5061 ----- tls.cfg ------ [server:default] method = TLSv1+ verify_certificate = no require_certificate = no private_key = /etc/kamailio/keys/sip1.key certificate = /etc/kamailio/keys/sip1.crt [server:10.1.23.33:5061] method = TLSv1+ verify_certificate = no require_certificate = no private_key = /etc/kamailio/keys/sip1.key certificate = /etc/kamailio/keys/sip1.crt [server:10.1.23.23:5061] method = TLSv1+ verify_certificate = no require_certificate = no private_key = /etc/kamailio/keys/sip2.key certificate = /etc/kamailio/keys/sip2.crt -- Best regards, Sergey Basov e-mail: sergey.v.basov(a)gmail.com

2 3

Sending INVITE’s on REGISTER
by Igor Olhovskiy 25 Oct '16

25 Oct '16

Hi! I have this kind of scenario Receive an INVITE -> send it to user, but if possible, if I’ll receive REGISTER with same credentials I’m calling to - drop an INVITE there also, if call was not answered on first location. Means create other INVITE branch async on the fly, if possible. Where at least to look at? Thanks. -- Best regards, Igor

2 2

How compile Kamailio in freebsd : add carrierroute module
by AFRICALLSHOP 25 Oct '16

25 Oct '16

Hello, I try to compile kamailio in freebsd. I want to add carrierroute module. I do : 1) cd /usr/ports/net/kamailio 2) make install but carrierroute module is not compiled. Could you help me? Regards Abdoul.

3 2

How compile Kamailio in freebsd : add carrierroute module
by Abdoul Osséni 24 Oct '16

24 Oct '16

Hello, I try to compile kamailio in freebsd. I want to add carrierroute module. I did : 1) cd /usr/ports/net/kamailio/ 2) make install but carrierroute module is not compiled. Could you help me? Regards Abdoul.

1 0

Rewriting To on BYE
by Charles Phillips 24 Oct '16

24 Oct '16

I need a sanity check if you don’t mind… I am setting up a stateless proxy for several (separate, not load balanced) Asterisk servers behind Kamailio 4.2 (with domain and dispatcher). All works as expected except when called party hangs up, as Asterisk seems to address the BYE “To” (that is sent to the calling party) to the originating domain, and not the IP of the UA (as it does in other cases). When this happens, stateless Kamailio does not seem to know how to route them, which makes sense. I assume I would need to rewrite the “To” field on BYE message so that it contains the UA’s IP (to match the BYE’s that are routed correctly). The RURI header contains the IP/port of the UA, so I am attempting to use it and here is where I am stuck… Here is the subst from onreply_route: if (is_method("BYE")) { xlog("L_NOTICE", "BYE Detected \n"); subst('/^To:(.*)<sip:(.*)@(.*)>(.*)$/To: <sip:\2@$rd:$rp>\4/ig'); } Here is the “To”: To: "Test User 2" <sip:102@zxcv.asdf.qwerty.net>;tag=8AB53900B9646F1DF05330E7FBF846C8 Any obvious glaring mistakes here? Would I then need additional logic to forward the BYE on? My apologies if this is wordy, or nonsensical...

2 1

PCSCF cannot create via header for sipml5 ACK package
by Serhat Guler 24 Oct '16

24 Oct '16

Hello, I finally managed to place a call from sipml5 webrtc client to Mercuro IMS client. The phone rings, and when I answer it sends 200 OK to the sipml5 where as sipml5 send back an ACK message which never passes the originating PCSCF. The PCSCF says: 8(3640) WARNING: <core> [msg_translator.c:2729]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found 8(3640) ERROR: <core> [msg_translator.c:1947]: build_req_buf_from_sip_req(): could not create Via header 8(3640) ERROR: <core> [forward.c:548]: forward_request(): building failed I doubt that the WebSocket connection is closed, cause when I terminate the call from Mercuro client a bye request is being sent to the sipml5. The ACK package: ACK sip:alice@192.168.0.10:49794;transport=udp SIP/2. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKvuly7bmxnN4aqM4zZTIS;rport From: "Bob"<sip:bob@net1.test>;tag=GxzKy1nCMEI1mR0RztrB To: <sip:alice@net1.test>;tag=18823 Contact: "Bob"<sip:bob@df7jal23ls0d.invalid ;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr" Call-ID: 5a500969-d0fa-14d1-7d0e-8605f4356ca6 CSeq: 3887 ACK Content-Length: Route: <sip:192.168.0.11:4060;lr;sipml5-outbound;transport=udp> Max-Forwards: 69 Route: <sip:mo@192.168.0.11:880 ;transport=ws;r2=on;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;did=e82.0c3> Route: <sip:mo@192.168.0.11:4060 ;r2=on;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;did=e82.0c3> Route: <sip:mo@192.168.0.11:6060 ;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;did=e82.f062> Route: <sip:mt@192.168.0.11:6060 ;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;did=e82.f062> Route: <sip:mt@192.168.0.11:4060 ;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;did=e82.1c3> User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04 Organization: Doubango Telecom Have been thinking for quite a while, but couldn't really find a reason why it wouldn't add the v,a header. A debug 3 level log file is also attached. Thanks in advance, Serhat

2 3

configuration kamailio 4.4.2 with Asterisk 13.10.0
by Serge Gabriel Ahnoux 24 Oct '16

24 Oct '16

Hello I am trying to configure kamailio 4.4.2 with asterisk 13.10.0. My Os is Debian 7.4 I followed the tutorial : http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb… The two softwares are installed on my server. So they have the same IP adress but the port is different like they said in the tutorial. I use too Mysql to stock users informations I have 2 databases Kamailio and Asterisk. I use ODBC for communicate beetwen asterisk server and his database. However when my sip client (linephone) connects to kamalio, kamailio doesn't forward information to asterisk. I don't kow the reason. So i can send message but not call If someone knows how to configure, please help me. I join my kamailio configuration file. Regards. ahnouxs(a)gmail.com

2 1

Dropping non-provisional replies in onreply_route
by Alex Balashov 21 Oct '16

21 Oct '16

The core documentation says that in a named onreply_route[], only provisional replies can be drop()'d. To drop any reply, it is necessary to use a global onreply_route. Is there any workaround for this, i.e. so I can drop a 2xx reply from a specific TM transaction? The reason is, to know whether to drop it, I need access to either AVPs or, ideally, dialog variables. Since the global onreply_route is executed by the core, I presume I won't have access to anything that persists through TM there. Thanks! -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 5

Edge proxy / Registrar Script
by Infinicalls Infinicalls 21 Oct '16

21 Oct '16

Hi, I am trying to get a simple Edge Proxy - Registrar setup working behind NAT. Initially I tried various scripts - including the one mentioned in the Oubound page and Peter Dunkley' example - but got some errors related to PATH. Can somebody give me a working script of both proxy and registrar? Any help would be gladly appreciated. regards Ganesh Kumar -- --- http://www.infinicalls.com

2 2

Basic TM suspend question
by Alex Balashov 21 Oct '16

21 Oct '16

Hi, If a transaction is currently t_suspended() and has not been t_continued(), will t_check_trans() still find that transaction for retransmission-dampening purposes, e.g. if a retransmission of the same request as the original one is received? Thanks, -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 3

Postgres: Exclusive Transaction Locks vs. Vacuumer
by Stefan Kohlhauser 21 Oct '16

21 Oct '16

Hey everyone! We noticed that under high load of registrations and using the pua module we can reach a state where the PUA table bloats our Postgres to several GB. However, a SELECT on pua only shows a few hundred entries. The other half million or so entries are obsoleted but not cleaned up. The reason for this seems to be that the autovacuumer of the Postgres can't get to work because the PUA table is almost permanently locked. Our Postgres logs are full with errors about the autovacuumer being canceled because it couldn't get to work. Looking at modules/db_postgres/km_dbase.c::db_postgres_start_transaction() I noticed that there are two locking modes used: str lock_write_end_str = str_init(" IN EXCLUSIVE MODE"); str lock_full_end_str = str_init(" IN ACCESS EXCLUSIVE MODE"); According to https://www.postgresql.org/docs/9.2/static/explicit-locking.html these modes mean forbidding everything else except reads and forbidding everything else. However, according to this documentation ROW EXCLUSIVE is the standard lock for entry altering operations (INSERT, UPDATE, DELETE). And this would still allow the autovacuumer to do its job. Is there a specific reason why these more restricting locking modes are used instead of ROW EXCLUSIVE? Thanks. Best regards, Stefan

2 1

TCP FIN after 10 minutes
by gmele 20 Oct '16

20 Oct '16

Hello, we have deployed a Kamailio acting as SIP proxy on a RHEL 7.2 machine. Clients (mainly mobile phones) connects to the proxy using a TLS protected TCP connection. In the kamailio config, we've set : #!ifdef WITH_TLS enable_tls=yes tcp_async=yes tcp_connection_lifetime=3605 tcp_accept_no_cl=yes tcp_crlf_ping = yes #!endif Our problem is that, even if we set the tcp_connection_lifetime to > 1 hour, the tcp connection is closed after 10 minutes: on tcp dumps, we see clearly the TCP FIN sent by machine hosting the kamailio proxy... Setting parameters tcp_keepidle/keepintlv/keepcnt in the kamailio config didn't change the behavior. We also set TCP keepalived at system level, but without result... This TCP closure is causing us lot of problems when calls between 2 UAs last more than 10 minutes because the REINVITE or BYE messages are lost. Also, closing the TCP connection will wake up the mobile app and make it resend a REGISTER, thing we absolutely want to avoid. Is there a parameter we can use to avoid this closure? I had a look in previous posts, found people with the same problem as mine, but no real solution... Thx for your help. Giovanni -- View this message in context: http://sip-router.1086192.n5.nabble.com/TCP-FIN-after-10-minutes-tp152646.h… Sent from the Users mailing list archive at Nabble.com.

2 2

Re: [SR-Users] help with NOTIFY
by Slava Bendersky 20 Oct '16

20 Oct '16

Hello Everyone, This message as continue conversation from http://lists.sip-router.org/pipermail/sr-users/2015-March/087557.html. That my previous post about it. I never was be be able forward NOTIFY from asterisk to client through kamailio. Right now in use asterisk 14 pjsip. Any help thank you. volga629 From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 19 October, 2016 18:36:52 Subject: [SR-Users] help with NOTIFY Hello Everyone, I am trying forward NOTIFY to client. My setup is PBX Server ---- local lan ---kamailio ---- internet ---- client The problem with whole setup that kamailio is not forwarding NOTIFY to client. Here are log. U 2016/10/19 14:23:29.581609 10.18.130.50:5160 -> 10.18.130.46:5060 NOTIFY sip:10101@10.18.130.46:5060 SIP/2.0. Via: SIP/2.0/UDP 10.18.130.50:5160;rport;branch=z9hG4bKPjONocd4FvI5LkOmSR6ZCA5Z1f7SYDb3Ty. From: <sip:8101@networklab.ca>;tag=ziApds23FQsPL2zd6q8pCUlwW.jwa6mK. To: "Slava" <sip:10101@networklab.ca>;tag=734f813f. Contact: <sip:10.18.130.50:5160>. Call-ID: A0bJWg8v0nGmtUmiHzUixQ... CSeq: 18547 NOTIFY. Route: <sip:kamailio_local_ip:5060;lr;r2=on;ftag=734f813f>. Route: <sip:kamailio_public_ip:5081;transport=tls;lr;r2=on;ftag=734f813f>. Event: presence. Subscription-State: active;expires=413. Allow-Events: message-summary, presence, dialog, refer. Max-Forwards: 70. User-Agent: FPBX-13.0.188.9(14.0.2). Content-Type: application/pidf+xml. Content-Length: 579. . <?xml version="1.0" encoding="UTF-8"?> <presence entity="sip:8101@10.18.130.50:5160" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:pp="urn:ietf:params:xml:ns:pidf:person" xmlns:es="urn:ietf:params:xml:ns:pidf:rpid:status:rpid-status" xmlns:ep="urn:ietf:params:xml:ns:pidf:rpid:rpid-person"> <note>On the phone</note> <tuple id="8101"> <status> <basic>closed</basic> </status> <contact priority="1">"Slava" <sip:10101@networklab.ca></contact> </tuple> <pp:person> <status> <ep:activities>ep:busy</ep:activities> </status> </pp:person> </presence> U 2016/10/19 14:23:32.045388 10.18.130.46:5060 -> 10.18.130.50:5160 SIP/2.0 408 Request Timeout. Via: SIP/2.0/UDP 10.18.130.50:5160;rport=5160;branch=z9hG4bKPju05rMNlzhfukLvaq4ELmq.QI4GBqDN-m;received=10.18.130.50. From: <sip:8101@networklab.ca>;tag=Yzy2m6O2AFT6rGSDBmo6PjcGaCWT1i5H. To: "Slava" <sip:10101@networklab.ca>;tag=ad5a945e. Call-ID: 2pq5iebmtLdB_OsMUs3EOg... CSeq: 14623 NOTIFY. Server: kamailio (4.3.4 (x86_64/linux)). Content-Length: 0. Any help thank you. volga629 _______________________________________________ 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

1 0

sipml5 fail invite kamailio ims web sockets
by Serhat Guler 20 Oct '16

20 Oct '16

Dear all, I can register the sipml5 client to my kamailio IMS setup successfully, but when I try to make a call the invite fails with "478 Unresolvable destination". From the wireshark trace file I think the PCSCF on the terminating side cannot resolve the destination. After the invite request fails the caller re-registers to the networkj automatically. The wireshark file is attached. The registration process happens successfully but all of the nodes give some errors: 10(3773) INFO: <script>: [REGISTER] from [sip:alice@net1.test] to [sip:alice@net1.test] 10(3773) ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=425 a=25 n=xlog 10(3773) ERROR: <script>: REGISTER (sip:alice@net1.test (192.168.0.10:14115) to sip:alice@net1.test, 25c80750-d52d-30c8-d373-c0136401a927) 10(3773) ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=437 a=16 n=if 10(3773) ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=437 a=5 n=route 10(3773) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=627 a=16 n=if 10(3773) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=622 a=25 n=mf_process_maxfwd_header 10(3773) DEBUG: <core> [parser/msg_parser.c:153]: get_hdr_field(): get_hdr_field: cseq <CSeq>: <59840> <REGISTER> 10(3773) DEBUG: <core> [parser/msg_parser.c:187]: get_hdr_field(): DEBUG: get_hdr_body : content_length=0 10(3773) DEBUG: maxfwd [mf_funcs.c:85]: is_maxfwd_present(): value = 70 10(3773) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=633 a=16 n=if 10(3773) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=627 a=26 n=sanity_check 10(3773) DEBUG: <core> [parser/msg_parser.c:89]: get_hdr_field(): found end of header 10(3773) DEBUG: sanity [mod_sanity.c:248]: w_sanity_check(): sanity checks result: 1 10(3773) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=639 a=16 n=if 10(3773) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=633 a=24 n=has_totag 10(3773) DEBUG: siputils [checks.c:97]: has_totag(): no totag 10(3773) DEBUG: <core> [select.c:412]: run_select(): Calling SELECT 0xb6e5fe40 10(3773) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=645 a=16 n=if 10(3773) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=639 a=25 n=is_method 10(3773) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=649 a=16 n=if 10(3773) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=645 a=24 n=t_lookup_request 10(3773) DEBUG: tm [t_lookup.c:1011]: t_check_msg(): DEBUG: t_check_msg: msg id=2 global id=1 T start=0xffffffff 10(3773) DEBUG: tm [t_lookup.c:466]: t_lookup_request(): t_lookup_request: start searching: hash=21728, isACK=0 10(3773) DEBUG: tm [t_lookup.c:424]: matching_3261(): DEBUG: RFC3261 transaction matching failed 10(3773) DEBUG: tm [t_lookup.c:648]: t_lookup_request(): DEBUG: t_lookup_request: no transaction found 10(3773) DEBUG: tm [t_lookup.c:1080]: t_check_msg(): DEBUG: t_check_msg: msg id=2 global id=2 T end=(nil) 10(3773) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=652 a=16 n=if 10(3773) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=649 a=25 n=is_method 10(3773) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=650 a=26 n=send_reply 10(3773) DEBUG: sl [sl.c:280]: send_reply(): reply in stateless mode (sl) 10(3773) DEBUG: websocket [ws_conn.c:429]: wsconn_get(): wsconn_get for id [2] 10(3773) DEBUG: websocket [ws_conn.c:437]: wsconn_get(): wsconn_get returns wsc [0xb2329228] refcnt [2] 10(3773) DEBUG: websocket [ws_frame.c:751]: ws_frame_transmit(): Tx message: The SCSCF says (while inviting): 0(3582) ERROR: dialog_ng [dlg_handlers.c:196]: populate_leg_info(): bad sip message or missing Contact hdr 0(3582) ERROR: dialog_ng [dlg_handlers.c:453]: dlg_onreply(): could not add further info to the dlg out Making calls between 2 IMS clients(boghe) is possible. Would appreciate any help (I can share the cfg files if needed). Cheers, Serhat

2 3

Kamailio script config - Generate Documentation
by José Seabra 20 Oct '16

20 Oct '16

Hello there, I'm sending this email in order to know if anyone had work or has any idea how we can generate Documentation from kamailio script using Comments? This would be great because we could write comments explaining the Route method during script development and use it to generate documentation about our kamailio script, saving time and leave our deployment documented. Thank you -- Best Regards José Seabra

3 5

Regarding pua_reginfo, server_address
by Ram Anji 20 Oct '16

20 Oct '16

Hi All, I am configured kamailio to handle reg info event subscriptions, My server is running if i configure the following parameter. #kamailio local ip is 192.168.20.20 *modparam("pua_reginfo", "server_address", "sip:reginfo@192.168.20.20 <sip%3Areginfo(a)192.168.20.20>")* If i am not configured the above server address, i am getting bad configuration Oct 18 11:31:36 arm /usr/sbin/kamailio[2787]: INFO: auth [auth_mod.c:333]: mod_init(): auth: qop set, but nonce-count (nc_enabled) support disabled Oct 18 11:31:36 arm /usr/sbin/kamailio[2787]: INFO: permissions [parse_config.c:251]: parse_config_file(): file not found: /etc/kamailio/permissions.allow Oct 18 11:31:36 arm /usr/sbin/kamailio[2787]: INFO: permissions [permissions.c:607]: mod_init(): default allow file (/etc/kamailio/permissions.al... rule set Oct 18 11:31:36 arm /usr/sbin/kamailio[2787]: INFO: permissions [parse_config.c:251]: parse_config_file(): file not found: /etc/kamailio/permissions.deny Oct 18 11:31:36 arm /usr/sbin/kamailio[2787]: INFO: permissions [permissions.c:616]: mod_init(): default deny file (/etc/kamailio/permissions.den... rule set *Oct 18 11:31:36 arm /usr/sbin/kamailio[2787]: ERROR: pua_reginfo [pua_reginfo.c:105]: mod_init(): server_address parameter not set* *Oct 18 11:31:36 arm /usr/sbin/kamailio[2787]: ERROR: <core> [sr_module.c:944]: init_mod(): Error while initializing module pua_reginfo (/usr/lib/...ginfo.so)* Oct 18 11:31:36 arm /usr/sbin/kamailio[2787]: INFO: <core> [sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Oct 18 11:31:36 arm kamailio[2771]: already running ... failed! Oct 18 11:31:36 arm kamailio[2771]: . Oct 18 11:31:36 arm systemd[1]: Started LSB: Start the Kamailio SIP proxy server. Hint: Some lines were ellipsized, use -l to show in full. I want to know what is the significance of this parameter and find no where this value is used. Thanks in advance. Regards, Raman

2 1

RTPProxy
by Rodrigo Moreira 20 Oct '16

20 Oct '16

Hi, What is difference between modules rtpproxy and rtpengine? Best regards. -- Rodrigo M. (37) 9132-4539 (34) 9889-3069 rodrigo.moreira2007

5 12

memory is not freed in system, while kamailio's statistic shows OK
by Vasiliy Ganchev 20 Oct '16

20 Oct '16

Hi folk! Have a production server - used as webrtc to sip proxy - it is built on kamailio v 4.4.2 - Server handles connections from ~4000 incoming SIP trunks (transport=TLS). Kamailio provides auth for this trunks. - Kamailio receives also connections from Webrtc app. Calls from Webrtc routed to one of the 4000 sip trunks. Server is running on Debian GNU/Linux 7 The number of calls on the system is small. Most of the load is generated by registration/re-registrations of SIP trunks The problem is following: - output of kamcmd core.shmmem and kamcmd pkg.stats shows constant using of memory - output of htop shows constantly increasing memory usage, e.g. after one week working: PID USER PRI NI VIRT RES SHR S CPU% MEM% TIME+ Command ..... 6013 kamailio 20 0 744M 134M 130M S 0.0 3.6 2:28.80 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 512 -M 32 -u kamailio -g kamailio 6002 kamailio 20 0 744M 134M 130M S 0.0 3.6 2:30.93 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 512 -M 32 -u kamailio -g kamailio 6012 kamailio 20 0 744M 134M 130M S 0.0 3.5 2:35.00 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 512 -M 32 -u kamailio -g kamailio 5991 kamailio 20 0 744M 133M 129M S 0.0 3.5 2:05.88 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 512 -M 32 -u kamailio -g kamailio 6018 kamailio 20 0 744M 133M 129M S 0.0 3.5 2:31.90 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 512 -M 32 -u kamailio -g kamailio ..... Despite running with: PKG_MEMORY=32, Amount of used memory per process is higher. Finishes all with monit killing kamailio, because of memory usage more than allowed studied this tutorial: https://www.kamailio.org/wiki/tutorials/troubleshooting/memory but did not find a way how to dig up the problem's root. Tried also to reproduce the issue in lab environment (having clone of production server). Generated load with incoming registration using PJSIP, but with no success - memory usage is constant. kamailio -v version: kamailio 4.4.2 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled with gcc 4.7.2 Any suggestion/idea how to troubleshoot deeper would be highly apprecieted Cheers! -- View this message in context: http://sip-router.1086192.n5.nabble.com/memory-is-not-freed-in-system-while… Sent from the Users mailing list archive at Nabble.com.

2 3

help with NOTIFY
by Slava Bendersky 19 Oct '16

19 Oct '16

Hello Everyone, I am trying forward NOTIFY to client. My setup is PBX Server ---- local lan ---kamailio ---- internet ---- client The problem with whole setup that kamailio is not forwarding NOTIFY to client. Here are log. U 2016/10/19 14:23:29.581609 10.18.130.50:5160 -> 10.18.130.46:5060 NOTIFY sip:10101@10.18.130.46:5060 SIP/2.0. Via: SIP/2.0/UDP 10.18.130.50:5160;rport;branch=z9hG4bKPjONocd4FvI5LkOmSR6ZCA5Z1f7SYDb3Ty. From: <sip:8101@networklab.ca>;tag=ziApds23FQsPL2zd6q8pCUlwW.jwa6mK. To: "Slava" <sip:10101@networklab.ca>;tag=734f813f. Contact: <sip:10.18.130.50:5160>. Call-ID: A0bJWg8v0nGmtUmiHzUixQ... CSeq: 18547 NOTIFY. Route: <sip:kamailio_local_ip:5060;lr;r2=on;ftag=734f813f>. Route: <sip:kamailio_public_ip:5081;transport=tls;lr;r2=on;ftag=734f813f>. Event: presence. Subscription-State: active;expires=413. Allow-Events: message-summary, presence, dialog, refer. Max-Forwards: 70. User-Agent: FPBX-13.0.188.9(14.0.2). Content-Type: application/pidf+xml. Content-Length: 579. . <?xml version="1.0" encoding="UTF-8"?> <presence entity="sip:8101@10.18.130.50:5160" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:pp="urn:ietf:params:xml:ns:pidf:person" xmlns:es="urn:ietf:params:xml:ns:pidf:rpid:status:rpid-status" xmlns:ep="urn:ietf:params:xml:ns:pidf:rpid:rpid-person"> <note>On the phone</note> <tuple id="8101"> <status> <basic>closed</basic> </status> <contact priority="1">"Slava" <sip:10101@networklab.ca></contact> </tuple> <pp:person> <status> <ep:activities>ep:busy</ep:activities> </status> </pp:person> </presence> U 2016/10/19 14:23:32.045388 10.18.130.46:5060 -> 10.18.130.50:5160 SIP/2.0 408 Request Timeout. Via: SIP/2.0/UDP 10.18.130.50:5160;rport=5160;branch=z9hG4bKPju05rMNlzhfukLvaq4ELmq.QI4GBqDN-m;received=10.18.130.50. From: <sip:8101@networklab.ca>;tag=Yzy2m6O2AFT6rGSDBmo6PjcGaCWT1i5H. To: "Slava" <sip:10101@networklab.ca>;tag=ad5a945e. Call-ID: 2pq5iebmtLdB_OsMUs3EOg... CSeq: 14623 NOTIFY. Server: kamailio (4.3.4 (x86_64/linux)). Content-Length: 0. Any help thank you. volga629

1 0

Active calls limit
by Ivan Dudko 19 Oct '16

19 Oct '16

Hello! I am trying to implement limit of active calls for subscribers. And for each subscriber this limit must be personal. I find example route in presentation of Daniel-Constantin Mierla. And i need to set $xavp(caller=>active_calls) = 1; for each dialog. But i can't understand how to load this number of active calls from some-thing similar of user profile or subscriber table? route[DIALOG] { if (is_method("CANCEL") || (has_totag() && is_method("INVITE|BYE|ACK"))) { dlg_manage(); return; } if (is_method("INVITE") && !has_totag() && !isflagset(WITH_ACTIVE_CALLS_LIMIT)) { if( $xavp(caller[0]=>active_calls) != $null && $xavp(caller[0]=>active_calls) > 0 ) { if(!get_profile_size("caller", "$fU@$fd", "$var(acsize)")) { send_reply("500", "No more active calls"); exit; } if($var(acsize)>=$xavp(caller[0]=>active_calls)) { send_reply("403", "No more active calls"); exit; } set_dlg_profile("caller", "$fU@$fd"); } setflag(WITH_ACTIVE_CALLS_LIMIT); dlg_manage(); } }

2 5

Kamailio Bench
by Rodrigo Moreira 19 Oct '16

19 Oct '16

Hello, i need your help, I'm having problems with the reverse response of an INVITE passing through a rtpproxy (NAT). Does anyone have any idea what might be going on? UE is informed of the result INVITE, but the UAS does not receive the test bench detail. Greetings. -- Rodrigo M. (37) 9132-4539 (34) 9889-3069 rodrigo.moreira2007

2 1

Problems with CDRs
by Diogenes Marcano 19 Oct '16

19 Oct '16

I am using kamailio with ubuntu, but I can't obtain CDRs. In my conf. file related to CDR I have: modparam("acc", "cdr_enable", 1) modparam("acc", "cdr_start_on_confirmed", 1) modparam("acc", "cdrs_table", "acc_cdrs") modparam("acc", "cdr_start_id", "start_time") modparam("acc", "cdr_end_id", "end_time") modparam("acc", "cdr_duration_id", "duration") But I don't know where is the problem. Regards Diogenes Marcano +58 414 117 2011

2 1

BYE issue
by Nelson Migliaro 18 Oct '16

18 Oct '16

Hello everybody, I am having issues with one SIP vendor. I have a Kamailio in bridge mode (private IP / Public IP) and some Asterisk and Media Gateways. Calls get established and I have two way audio but when the remote party hangs up the call, the BYE arrives to the Kamailio and does not move forward. I think the problem is SIP vendor rewrite the BYE header and change the asterisk IP with the public IP of the kamailio. The IP that appears in the header of the BYE have to be the same that appears in the contact (UAC that send the call, in my case the Asterisk). Vendor should not change that IP. ¿Am I correct? Thank you ----------------------------------------------------------------------------------------------------- INVITE ---------------------------------------------------------------------------------------------------- 2016/10/17 18:50:49.110967 PUBLIC-KAMAILIO-IP:5060 -> VENDOR-IP:6060 INVITE sip:DESTINATION-NUMBER@VENDOR-IP:6060 SIP/2.0 Record-Route: <sip:PUBLIC-KAMAILIO-IP;r2=on;lr=on;ftag=as5e87b96c;vsf=AAAAAAAAAAAAAAAAAABQUk9fRVYAU0UuODY-;vst=AAAAAAQEAw8MDgsAAHYAcVddXkZWRVVDVl1MMDIudm9pY2U G9jYWw-;did=09b.9572;nat=yes> Record-Route: <sip:PRIVATE-KAMAILIO-IP;r2=on;lr=on;ftag=as5e87b96c;vsf=AAAAAAAAAAAAAAAAAABQUk9fRVYAU0UuODY-;vst=AAAAAAQEAw8MDgsAAHYAcVddXkZWRVVDVl1MMDIudm9pY2U G9jYWw-;did=09b.9572;nat=yes> Via: SIP/2.0/UDP PUBLIC-KAMAILIO-IP;branch=z9hG4bK06a.07540d0e2f32a811ecf9c0a5235dc77a.1 Via: SIP/2.0/UDP PRIVATE-ASTERISK-IP:5060;received=PRIVATE-ASTERISK-IP;branch=z9hG4bK6bb5a7b3;rport=5060 Max-Forwards: 69 From: "SOURCE-NUMBER" <sip:SOURCE-NUMBER@MY-COMPANY>;tag=as5e87b96c To: <sip:DESTINATION-NUMBER@VENDOR-IP> Contact: <sip:SOURCE-NUMBER@PRIVATE-ASTERISK-IP:5060> Call-ID: 025cc3717ba59faa000cf4db6f8be588@PRIVATE-ASTERISK-IP:5060 CSeq: 102 INVITE User-Agent: UAC Date: Mon, 17 Oct 2016 16:53:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 269 v=0 o=root 292850421 292850421 IN IP4 PUBLIC-KAMAILIO-IP s=Asterisk PBX c=IN IP4 PUBLIC-KAMAILIO-IP t=0 0 m=audio 23456 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv a=nortpproxy:yes ----------------------------------------------------------------------------------------------------- BYE ----------------------------------------------------------------------------------------------------- 2016/10/17 18:50:58.241666 VENDOR-IP:6060 -> PUBLIC-KAMAILIO-IP:5060 BYE sip:SOURCE-NUMBER@PUBLIC-KAMAILIO-IP:5060 SIP/2.0 Via: SIP/2.0/UDP VENDOR-IP:6060;branch=z9hG4bKeff4.48943e76.0 Via: SIP/2.0/UDP VENDOR-IP:5060;branch=z9hG4bK1d4e605e4ll19f74fBYE421ce8658050206 Max-Forwards: 34 Route: <sip:PUBLIC-KAMAILIO-IP;lr;r2=on;ftag=as5e87b96c;vsf=AAAAAAAAAAAAAAAAAABQUk9fRVYAU0UuODY-;vst=AAAAAAQEAw8MDgsAAHYAcVddXkZWRVVDVl1> Route: <sip:PRIVATE-KAMAILIO-IP;lr;r2=on;ftag=as5e87b96c;vsf=AAAAAAAAAAAAAAAAAABQUk9fRVYAU0UuODY-;vst=AAAAAAQEAw8MDgsAAHYAcVddXkZWRVVDVl1> To: "SOURCE-NUMBER"<sip:SOURCE-NUMBER@YO>;tag=as5e87b96c From: <sip:DESTINATION-NUMBER@PUBLIC-KAMAILIO-IP>;tag=421ce86-co1547-INS001 Call-ID: 025cc3717ba59faa000cf4db6f8be588@PRIVATE-ASTERISK-IP:5060 CSeq: 154701 BYE User-Agent: VENDOR Content-Length: 0 -----------------------------------------------------------------------------------------------------

3 5

420 Bad Extension error
by Infinicalls Infinicalls 18 Oct '16

18 Oct '16

Hi all, I am testing Edge Proxy setup with the example given here : http://kamailio.org/docs/modules/4.2.x/modules/outbound.html and I get this error : U 2016/10/06 09:18:48.331580 10.0.0.4:5060 -> 52.233.25.164:5060 SIP/2.0 420 Bad Extension. Via: SIP/2.0/UDP 10.1.0.4;branch=z9hG4bK162a.e9f73fe5e52ded1f30801aa00ee2a771.0; received=52.233.25.164. Via: SIP/2.0/UDP 10.114.70.81:1228;received=113.193.149.49;rport=1228;branch=z9h G4bKPj3ec7c87e09d6452b9e68e60052cb2651. From: "User1" <sip:user1@infinicalls-proxy1.myserver.com>; tag=ac1e36aec8bf445bbec8e43cf28e9f48. To: "User1" <sip:user1@infinicalls-proxy1.myserver.com>;ta g=b27e1a1d33761e85846fc98f5f3a7e58.b172. Call-ID: a270d285de6145c6b8670b909b104e74. CSeq: 20149 REGISTER. Contact: <sip:user1@10.114.70.81:1228;ob>;expires=300. Unsupported: path. Path: <sip:10.1.0.4;lr>. Supported: outbound. P-Registrar-Error: No support for found Path indicated. Server: kamailio (4.4.3 (x86_64/linux)). Content-Length: 0. Both my proxy and registrar are behind NAT. rtpproxy is installed and started. I already have modparam("registrar", "use_path", 1) in my registrar. Any idea what is missing? Thanks. regards Ganesh Kumar

2 17

Sip route can be as IMS PCSCF node
by Surender Singh 17 Oct '16

17 Oct '16

Hi, Can we use sip router as IMS PCSCF to connectivity with IMS. Does sip router supports the diameter RX interface to connectivity with PCRF . Regards Surender Singh ::DISCLAIMER:: ---------------------------------------------------------------------------------------------------------------------------------------------------- The contents of this e-mail and any attachment(s) are confidential and intended for the named recipient(s) only. E-mail transmission is not guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or may contain viruses in transmission. The e mail and its contents (with or without referred errors) shall therefore not attach any liability on the originator or HCL or its affiliates. Views or opinions, if any, presented in this email are solely those of the author and may not necessarily reflect the views or opinions of HCL or its affiliates. Any form of reproduction, dissemination, copying, disclosure, modification, distribution and / or publication of this message without the prior written consent of authorized representative of HCL is strictly prohibited. If you have received this email in error please delete it and notify the sender immediately. Before opening any email and/or attachments, please check them for viruses and other defects. ----------------------------------------------------------------------------------------------------------------------------------------------------

2 1

sipml5 register to kamailio IMS (Unauthorized - Challenging the UE)
by Serhat Guler 17 Oct '16

17 Oct '16

Hi, I am trying to register sipml5 webrtc client to my kamailio IMS setup. I have tried to register the client both with ws and wss, but it seems to be that the sipml5 doesn't calculate the authentication digest right. The authentication mechanism i set to AKAv1-MD5 as default in the hss. A simple wireshark file is attached. .10 being the host, .11 being the kamailio server. >From the output of scscf we can see that the digests do not match. [REGISTER] from [sip:bob@net1.test] to [sip:bob@net1.test] ims_auth [authorize.c:824]: authenticate(): uri=sip:net1.test nonce=rB2iDuerHwoy+LUStSOsYojAESfWmAAApFZ3XNB8FdA= response=61cbdbeb47c9880ededfca51c3801800 qop=auth-int nc=00000001 cnonce=2d4545cf4c935c8094c8b1da3d4a2976 hbody=d41d8cd98f00b204e9800998ecf8427e ims_auth [authorize.c:872]: authenticate(): UE said: 61cbdbeb47c9880ededfca51c3801800 and we expect 7e27ef414cf37d96cd1c849bb7e59415 ha1 a53988ba0b257941bc747b1026225c77 (REGISTER) tm [tm.c:1265]: w_t_reply(): ERROR: t_reply: cannot send a t_reply to a message for which no T-state has been established Thanks a lot, Serhat

2 2

How to use sql transformations?
by Daniel Tryba 17 Oct '16

17 Oct '16

How to use the sql transformation? https://www.kamailio.org/wiki/cookbooks/4.4.x/transformations#sql_transform… has the following example: xlog("$$rm = $rm = $(rm{s.sql})"); But adding this to the request_route and starting kamailio will fail: ERROR: pv [pv_trans.c:2351]: tr_parse_string(): unknown transformation: sql}/sql/3! ERROR: <core> [pvapi.c:1629]: tr_lookup(): error parsing [{s.sql}] ERROR: <core> [pvapi.c:1010]: pv_parse_spec2(): bad tr in pvar name "rm" ERROR: <core> [pvapi.c:1036]: pv_parse_spec2(): invalid parsing in [$(rm{s.sql})] at (4) ERROR: xlog [xlog.c:512]: xdbg_fixup_helper(): wrong format[$$rm = $rm = $(rm{s.sql})] ERROR: <core> [route.c:1154]: fix_actions(): fixing failed (code=-1) at cfg:/etc/kamailio/kamailio.cfg:372 line 372 is the above xlog and sqlops.so is loaded (and works). Anybody got a working example of this? Or an other hint to prevent sql injections when using user supplied variables in sql queries?

1 0

503 Service unavailable to 500
by Dhruvin Desai 17 Oct '16

17 Oct '16

Hi 503 Service unavailable received at kamailio SIP server is forwarded as 500 server internal error to the UAC. Which seems fine based on the fact that there is no retry-after header. Would like to know if there is any way to to control the SIP Headers like adding Proprietary SIP Headers to the 500 "Server Interrnal Errror" created by kamalio stack for 503 service unavailable (without retry-after Header) It seems that Kamalio does not retain the proprietary sip headers received in 503 to 500 response forwarded. Any insight into the issue would be helpful -- Warm Regards Dhruvin Desai.

2 1

RTPEngine & Homer
by Grant Bagdasarian 17 Oct '16

17 Oct '16

Hello, I'm trying to get rtpengine to send its stats to Homer, but I'm unsure to which IP I should configure RTPengine to send the stats? We have a Kamailio server configured with the sipcapture module, which works perfectly for SIP traffic, but RTP stats are not visible in Homer. Do I need to point RTPengine to the sip capture server or directly to Homer? Running RTPEngine 4.5.2 and Kamailio 4.4.1. Regards, Grant

1 0

HOMER / HEP support in SEMS?
by Daniel Pocock 14 Oct '16

14 Oct '16

Does anybody know if SEMS has HOMER / HEP support like Kamailio, or if it is under development? It would be useful to extract RTCP stats from SRTP streams, the standalone HOMER captagent can't see inside them when they are encrypted (SRTCP) Regards, Daniel

2 1

Accounting CDR generation
by Ivan Dudko 14 Oct '16

14 Oct '16

Hello! I am setting this in kamailio.cfg #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) modparam("acc", "cdr_enable", 1) modparam("acc", "cdr_start_on_confirmed", 1) modparam("acc", "cdrs_table", "acc_cdrs") modparam("acc", "cdr_extra","src_user=$fU;src_domain=$fd;src_ip=$si;dst_user=$rU;dst_ouser=$tU;dst_domain=$rd;sip_code=$rs;sip_reason=$rr;callid=$dlg(callid);dst_ip=$Ri") modparam("acc", "cdr_start_id", "start_time") modparam("acc", "cdr_end_id", "end_time") modparam("acc", "cdr_duration_id", "duration") #!endif But in CDRs table ('acc_cdrs') field dst_user=$rU is always empty if call is not completed with remote side answer. But in accounting table ('acc') this field isn't empty. Is this behavior is correct? I think it is not correct.

2 7

Websocket with kamailio IMS
by Serhat Guler 13 Oct '16

13 Oct '16

Dear all, I am trying to configure websockets for kamailio IMS. My basic kamailio(without IMS) setup works properly with websockets and without websockets. The IMS functionality also works, but only with IMS clients and now I want to use webrtc clients and make calls. At the moment I am trying to add the websockets support to the pcscf. I have did some changes to the pcscf cfg file. My problem is not really related to the cfg of pcscf as the component is listening for tcp/tls connections, but I cannot connect through the assigned ports. The output of the pcscf component and the the wireshark pcap file for the Websocket are attached. (I basically tried to browse http://server.ip.address:tcp.port or https://server.ip:tls.port without using a webrtc client yet). 192.168.0.10 -> host 192.168.0.11 -> virtual machine running kamailio Thanks in advance, Serhat

2 1

Reload PIPILIMIT table from DB
by Grant Bagdasarian 12 Oct '16

12 Oct '16

Hello, Is it possible to live reload the pipelimit module so it gets new values from the DB without a restart of Kamailio? I couldn't find a command to do this. Regards, Grant Bagdasarian CM

2 3

prefix_route and db_cluster not working
by Martin Nyström 12 Oct '16

12 Oct '16

Hi, I have an issue when I am unable to get prefix_route to work together with db_cluster. If I set the db_url manually it works great, pointing it to a specific database. However we need the redundancy for this module. Dispatcher uses the same cluster without any issues. The following are my configuration sets: # DB CLUSTER (db_cluster.inc) loadmodule "db_cluster.so" modparam("db_cluster", "connection", "KAMDB01=>mysql://kamailio:password123@192.168.55.10/kamailio") modparam("db_cluster", "connection", "KAMDB02=>mysql://kamailio:password123@192.168.55.11/kamailio") modparam("db_cluster", "cluster", "AccCluster=>KAMDB02=9s9s;KAMDB01=4s4s") modparam("db_cluster", "cluster", "Settings=>KAMDB02=9s9s;KAMDB01=4s4s") # KAMAILIO.CFG (kamailio.cfg) # PREFIX ROUTE MODPARAM loadmodule "prefix_route.so" include_file "db_cluster.inc" modparam("prefix_route", "db_url", "cluster://Settings") modparam("prefix_route", "db_table", "prefix_route") modparam("prefix_route", "exit", 1) Syslog output: Oct 10 09:53:03 kamailio01 /usr/local/sbin/kamailio[5225]: ERROR: <core> [db_cmd.c:122]: db_cmd(): DB driver cluster does not implement mandatory db_cmd function Oct 10 09:53:03 kamailio01 /usr/local/sbin/kamailio[5225]: ERROR: <core> [db_cmd.c:168]: db_cmd(): db_cmd: Cannot create db_cmd structure Oct 10 09:53:03 kamailio01 /usr/local/sbin/kamailio[5225]: ERROR: prefix_route [prefix_route.c:126]: pr_db_load(): prefix_route: db_load(): db_cmd() failed Oct 10 09:53:03 kamailio01 /usr/local/sbin/kamailio[5225]: : prefix_route [prefix_route.c:214]: mod_init(): prefix_route: db_load() failed Oct 10 09:53:03 kamailio01 /usr/local/sbin/kamailio[5225]: ERROR: <core> [sr_module.c:974]: init_mod(): Error while initializing module prefix_route (usr/local/lib64/kamailio/modules/prefix_route.so) Oct 10 09:53:03 kamailio01 /usr/local/sbin/kamailio[5225]: INFO: <core> [sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Ubuntu 14.04LTS version: kamailio 4.3.6 (x86_64/linux) 105a17 flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 105a17 compiled on 10:06:12 Oct 4 2016 with gcc 4.8.4 Thank you.

2 3

Enabling accounting module
by Diogenes Marcano 12 Oct '16

12 Oct '16

Can I enable accounting modules acc and acc_cdrs with out diameter? Best regard

2 1

Permissions module question
by Ivan Dudko 11 Oct '16

11 Oct '16

Hello! I am new to kamailio and trying to use it vanilla config. Now main question is how to use ip based auth. I found recent post of Daniel-Constantin Mierla: http://lists.sip-router.org/pipermail/sr-users/2011-December/071147.html Here he recommends to use 'address' table from permissions module, I try yo use advice and add this lines at config begin: #!define WITH_MYSQL #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!define WITH_AUTH #!define WITH_IPAUTH #!define WITH_USRLOCDB Database is created and kamailio can access it. I am add user 1000 to kamailio via kamctl and successful register it with soft-phone. I create trunk without registration to kamailio on asterisk server. And trying to call from asterisk to user 1000. Call is successful. I try to create file /etc/kamailio/permissions.deny with content 'ALL : ALL'. And retry previous call. It still sucessful. I try to add record with asterisk address to 'address' table with group 1. And retry previous call. It still sucessful. I am confused. I do not now how to disable any address for ip_auth except if it in the 'address' table. And allow any address with if it request kamailio with registration. -- Best Regards, Ivan Dudko

3 4

Kamailio - DMQ dns srv Question
by José Seabra 11 Oct '16

11 Oct '16

Hello, I have a doubt related with DMQ dns behavior, I noticed that when kamailio starts, it tries to resolve DMQ name configured on parameter notification_address as the following sequence: 1. SRV 2. A 3. AAAA Isn't supposed kamailio try first resolve the NAPTR DMQ name, and then SRV? I'm asking this because kamailio is trying resolve the SRV record without any transport protocol specified on query, as my dns server only accepts queries on the format "_Service._Proto.Name" the SRV will never be resolved. Thank you. BR -- José Seabra

4 13

transaction is not suspended ERROR
by david 11 Oct '16

11 Oct '16

Hello all i'm having some errors sometimes and i'm not able to know why they happen i made a lua configuration script where i create some ESL sockets with several freeswitches (10) (around 500-600 sockets in total, depending on the number os children (8) and listen ips we have (5)), and i use the lua script to send some API to the FSW by a function. No matter what the FSW replies. i dont know why, but i think that having that conn:bgapi($command) in the lua script, makes the issue to happen. theorically, the lua script waits to receive the response from the fsw through the tcp socket, and after that the script continues (with an async_sleep command), but after the API sent, i see (only sometimes, few times) an async_sleep seem to fail when doing the t_continue with the following error WARNING: tm [t_suspend.c:186]: t_continue(): transaction is not suspended [20748:225229907] could it be posible that having manye ESL sockets opened in the kamailio server makes this to happen? how many async workers are needed? i have 8 right now is there a way where i can know or correlate the log to a $ci or something known? best regards david

2 3

Kamailio on Redhat 7.2
by kamailio noob 11 Oct '16

11 Oct '16

Hi All, Can anyone please provide me the details that how to install kamailio on Redhat 7.2. I am following the below url guide for installation of Kamailio. https://www.kamailio.org/wiki/install/4.4.x/git when i compile kamailio code I am not able to find certain scripts in /usr/local/sbin Can anyone help regarding this ? Regards, Atif Ramzan.

2 1

Busy return code when a user has an ongoing transaction
by Gholamreza Sabery 10 Oct '16

10 Oct '16

I want to be able to return a Busy code foe a user that is talking to someone else and receives an INVITE for a new call. How can I implement this in Kamailio? Regards

2 2

Order of record-route
by Dirk Teurlings - Signet B.V. 10 Oct '16

10 Oct '16

Hi, Looking at an issue with SIP-Cisco and a Barracuda firewall, I've come across something I need to know more about. The setup is as follows: CLIENT -> KAMAILIO -> ASTERISK CLIENT <- KAMAILIO <- ASTERISK Client sends INVITE, Kamailio forwards to Asterisk, SIP 100 and SIP 183 are being sent and finally a 200 OK. We now expect the Client to send back an ACK packet, but that never reaches us. In this case the SIP 183 is the first packet that contains the full record route of the path, the 200 OK contain the same record-routes being: Record-Route: <sip:192.168.0.200;r2=on;lr;ftag=12D1120C-1C4;did=4c8.9203;nat=yes> Record-Route: <sip:1.2.3.4;r2=on;lr;ftag=12D1120C-1C4;did=4c8.9203;nat=yes> Record-Route: <sip:4.3.2.1;r2=on;lr;did=4c8.3f471377> Record-Route: <sip:10.0.0.101;r2=on;lr;did=4c8.3f471377> 192.168.0.200 is the private address of Kamailio 1.2.3.4 is the public address of Kamailio 4.3.2.1 is the public address of the Client 10.0.0.101 is some private addres of the Client What we're seeing is that the Client is sending his ACK message to our 192.169.200 private address, which he will never reach. So me question in this case is; is the order of the Record-route headers relevant and do I have any say in which order Kamailio outputs these headers? To be clear, the Kamailio config only contains one reference to record_route which is caught for INVITE messages, which only fires when all things like NAT and transaction checks haven't already picked it up. Kind regards, Dirk

3 5

kamctl - Error opening FIFO
by Hermann Norpois 10 Oct '16

10 Oct '16

Hello, I have problems with the behaviour of kamctl ubuntu@ip-172-31-32-10:/etc/kamailio$ kamctl ul show ERROR: Error opening Kamailio's FIFO /var/run/kamailio/kamailio_fifo ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name", "/var/run/kamailio/kamailio_fifo")' in your config ERROR: and also have loaded the mi_fifo module. #my FIFO file ubuntu@ip-172-31-32-10:/etc/kamailio$ ls -l /var/run/kamailio/kamailio_fifo prw-rw---- 1 kamailio kamailio 0 Oct 7 09:01 /var/run/kamailio/kamailio_fifo #FIFO in config ubuntu@ip-172-31-32-10:/etc/kamailio$ grep fifo kamailio.cfg loadmodule "mi_fifo.so" # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/var/run/kamailio/kamailio_fifo") #With sudo it works ubuntu@ip-172-31-32-10:/etc/kamailio$ sudo kamctl ul show Domain:: location table=1024 records=0 max_slot=0 Further ... ubuntu@ip-172-31-32-10:/etc/kamailio$ kamctl restart INFO: Stopping Kamailio : ERROR: No PID file found (/var/run/kamailio.pid)! Kamailio probably not running INFO: check with 'ps axw | /bin/egrep kamailio' #But kamailio is running ubuntu@ip-172-31-32-10:/etc/kamailio$ ps axw|egrep kamailio 4122 ? S 0:00 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4123 ? S 0:00 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4124 ? S 0:00 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4125 ? S 0:00 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4126 ? S 0:00 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4127 ? S 0:46 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4128 ? S 0:44 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4129 ? S 0:44 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4130 ? S 0:44 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4131 ? S 0:03 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4132 ? S 909:27 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4133 ? S 0:00 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4134 ? S 5:08 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4135 ? S 25:51 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4136 ? S 26:02 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4137 ? S 26:08 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4138 ? S 26:18 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4139 ? S 10:17 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 27344 pts/1 S+ 0:00 egrep --color=auto kamailio Whats wrong with my usage of kamctl? Thanks Hermann

3 2

Q: (markup) language for defining structures and function prototypes
by Daniel-Constantin Mierla 10 Oct '16

10 Oct '16

Hello, writing to see if anyone is aware of any (well known) markup language for defining structures and function prototypes. The goal is to use it for specifying the functions exported by KEMI (https://www.kamailio.org/wiki/devel/config-engines) so we can generate C code for the exported structure, but more important, to get at least the index with the functions available over the KEMI. I feel it's going to be a big burden to write C code and docs in two places, even worse, to keep both of them sync'ed over the time. At this moment, there are many functions exported via KEMI, but not documented anywhere (apart of the C code). It should allow defining the name of the function, the parameters with their associated types, return code and eventually the description for each of them. At the end we could invent one based on xml format (or something similar), but I would better use something that already exists there. I searched quickly, but couldn't find anything straightaway. A hint could be a format that is used to document the RPC commands. Once the format is decided, we may have to write a tool to generate C code to be compiled with the module and the xml docbook file to be included in the documentation. Any idea to solve this welcome! Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Oct 24-26, 2016 - http://www.asipto.com

1 0

Kamailio stop processing SIP
by Igor Potjevlesch 07 Oct '16

07 Oct '16

Hello, Someone has an idea? Again, Kamailio stop processing SIP during more than 30s. Nothing print in the logs. The processing back on when a script detect no answer and restart the service. Regards, Igor. De : Igor Potjevlesch [mailto:igor.potjevlesch@gmail.com] Envoyé : vendredi 23 septembre 2016 11:05 À : sr-users(a)lists.sip-router.org Objet : Dimension number of child Hello, I'm wondering if there's a best practise to dimension the number of child to setup? I see, very often, Kamailio that stops to reply to SIP traffic for couple a seconds. This cause some calls drop if RE-INVITE are not processed for example. So, I'm wondering if the issue is not due to a lack of child for processing the SIP traffic. Thank you for your inputs. Regards, Igor.

3 6

Migrating from OpenSER config file question.
by Brian McCrary 07 Oct '16

07 Oct '16

Hello group, I know this may be hard to believe but I'm in the process of upgrading an old, but stable, OpenSER 1.0 group of servers to Kamilio. I'm basically going to sort of start from scratch with the database and export my old MySQL database and write some scripts to reimport the data using kamctl scripts since the database structure has changed substantially since then it looks like. I've mostly migrated the config file over fairly easily, except for AVPops. I've got a section of config file that writes into the usr_preferences database for call forwarding, but I seem to get the same error anywhere I try to write to the database using AVPops. Here's the output I get when checking the config: 0(1378) DEBUG: <core> [pvapi.c:268]: pv_cache_add(): PV cache not initialized, doing it now 0(1378) ERROR: <core> [pvapi.c:828]: pv_parse_spec2(): error searching pvar "avp" 0(1378) ERROR: <core> [pvapi.c:1032]: pv_parse_spec2(): wrong char [s/115] in [$avp(s:callfwd)] at [5 (5)] 0(1378) : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 145, column 4-18: Can't get from cache: $avp(s:callfwd) ERROR: bad config file (1 errors) Here's what I think are the revelant portions of the config file: modparam("avpops", "db_url", "mysql://user:pass@localhost/kamailio") modparam("avpops", "avp_table", "usr_preferences") modparam("avpops", "uuid_column", "uuid") modparam("avpops", "username_column", "username") modparam("avpops", "domain_column", "domain") modparam("avpops", "attribute_column", "attribute") modparam("avpops", "value_column", "value") modparam("avpops", "type_column", "type") <snip> if (avp_db_load("$from/username", "$avp(s:callfwd)")) { if(!avp_db_delete("$from/username", "$avp(s:callfwd)/usr_preferences")) { sl_send_reply("500", "Server DB error"); } } # avp_write("$ruri","$avp(s:callfwd)"); # Old OpenSER 1.0 syntax... $avp(s:callfwd) = $ruri; # New Kamailio syntax ?? avp_db_store("$from/username","$avp(s:callfwd)/usr_preferences"); t_relay(); return; Basically I just want to write the ruri into the usr_preferences database with an attribute of callfwd but not sure what I'm missing. I see the error "Can't get from cache" but not sure how that pertains to what I'm trying to do. Anyone have any ideas? Thanks, Brian

2 5

Size of passwords fields in db
by Daniel-Constantin Mierla 06 Oct '16

06 Oct '16

Hello, writing here to decide on a topic opened by pull request 779: - https://github.com/kamailio/kamailio/pull/779 what would be a fair size for db column storing a password that one would like to have for proper security? I would like to make it consistent over all tables that have a password column by defining a xml entity for the size of these columns. The pull request suggests 64 chars, has anyone other opinions on making it larger or smaller? If they are defined varchar, then should not be a problem of allocated size, so we can go with 128 if someone feels it worth doing larger now so we don't have to change it again in the near future. This change is about db schema, the modules I expect to work with allocated strings (or have length checks) in this case and should not be affected. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Oct 24-26, 2016 - http://www.asipto.com

2 4

Fwd: TOPOH Module in Kamialio
by Ginti Saurabh 06 Oct '16

06 Oct '16

Hi, Can you let me know what is TOPOH module do in kamailio? what does it do? and if commented out this line then will it effect the system?

2 6

Load balancing with Kamailio
by Nitesh Bansal 06 Oct '16

06 Oct '16

Hello, I'm planning on doing some smart load balancing with Kamailio. We have a distributed network, with multiple Kamailio boxes in different locations serving as Ingress SBC, these Kamailio boxes are the entry point for a SIP call and then they route the call to a pre-configured Asterisk boxes. I want to move away from this, I would like these Kamailios to be able to distribute the traffic to Asterisk boxes based on the actual load on these boxes, the goal is to be more dynamic? Is there any Kamailio module which could do that? Do I need to integrate some other tool(Homer etc) with Kamailio to achieve this? Any suggestions are welcome. -- *Nitesh Bansal* Business Development Engineer http://www.voxbone.com/

2 1

Serial Forking and t_suspend
by Kelvin Chua 06 Oct '16

06 Oct '16

we have serial forking already setup properly but need to t_suspend(). found out that it completely breaks the destination set. other than manually rebuilding the destination set after t_continue(), are there any other solution out there? Kelvin Chua

2 1

Re: [SR-Users] Identical Kamailio setup on 2 hosts
by Daniel Tryba 05 Oct '16

05 Oct '16

On Fri, Sep 30, 2016 at 10:29:16PM +0530, Infinicalls Infinicalls wrote: > Hi Daniel, > This seems to be ideal setup for my requirement. Would you mind > sharing the kamailio.cfg file with me (with all the important data > masked) ? Not much important stuff to censor. LBIPADDR is the ipadress of the machine itself. REGISTRARNET/30 is the net of the backend machines. The dispatcher.list contains the backend registrars The backend is essentially the default config (with an always on rtpengine) with the following additions: You need to enable path.so and set modparam("path", "use_received", 1) Tell the registrar module to use path (path_mode 1 fixes you 420 problem): modparam("registrar", "use_path", 1) modparam("registrar", "path_mode", 1) modparam("registrar", "path_use_received", 1) If you want to send keepalives from the registrar to the enduser: modparam("nathelper", "ping_nated_only", 0) modparam("nathelper", "udpping_from_path", 1) the loadbalancer config has a little blurb about relaying the sipping from the backend manually using the path header. But I'd advice to ping from the an external process to have better control (eg sipsak)

2 3

Addressing a distinct PSI result is: 555 AS Contacting Failed - iFC terminated dialog
by Valentin Christoph 05 Oct '16

05 Oct '16

Hi all, I'm trying to send a call to an AS via a kamailio S-CSCF (release 4.4.3). I'm addressing the AS by a distinct PSI: sip:04325432101@domain.net;user=gsmr. At the first try the S-CSCF sends a SAR to HSS and receives a successful SAA (tshark traces attached), but the S-CSCF rejects the call with 555 AS Contacting Failed - iFC terminated dialog. At the second try (a few seconds later), the S-CSCF doesn't exchange SAR/SAA with the HSS, and now the call is successfully forwarded to the AS. This problem is permanent, it is not the result of a temporary loss of connectivity between S-CSCF and AS. I checked the iFCs that I had configured at the HSS and I think, they are OK. I attached - tshark traces from the node, where the S-CSCF resides: 2016_09_19_error_555_17_scscf_filtered.pcap - WITH_DEBUG logs from kamailio: 2016_09_19_error_555_17_kamailio_1st_try.log - A tarball of /usr/local/etc/kamailio: 2016_09_19_error_555_17_kamailio.tar.gz Can anybody help? Is it a bug in kamailio or is the problem on my side? Thanks, Christoph The information contained in this e-mail message is privileged and confidential and is for the exclusive use of the addressee. The person who receives this message and who is not the addressee, one of his employees or an agent entitled to hand it over to the addressee, is informed that he may not use, disclose or reproduce the contents thereof, and is kindly asked to notify the sender and delete the e-mail immediately.

2 4

484 Address Incomplete error
by Infinicalls Infinicalls 05 Oct '16

05 Oct '16

Hi all, I am trying a simple proxy (with dispatcher) and a registrar. However, I have this issue U 2016/10/05 08:22:44.057195 10.0.0.4:5060 -> 52.233.25.164:5060 SIP/2.0 484 Address Incomplete. Via: SIP/2.0/UDP 10.1.0.4;branch=z9hG4bK8653.a9c65320000000000000000000000000.0;received=52.233.25.164. To: <sip:51.140.187.167:5060>;tag=b27e1a1d33761e85846fc98f5f3a7e58.b79c. From: <sip:calls@infinicalls.com>;tag=533cb9e91f4b999cf76861cbb9ed54ed-d3d8. CSeq: 10 OPTIONS. Call-ID: 28fcdbee608363cf-5442(a)127.0.0.1. Server: kamailio (4.4.3 (x86_64/linux)). Content-Length: 0. I have set modparam("dispatcher", "flags", 2) Any idea? Thanks. regards Ganesh Kumar

2 1

NatHelper - Received adding on Contact header
by Yasin CANER 04 Oct '16

04 Oct '16

3 4

Sip Proxy server_kamailio Features
by Surender Singh 03 Oct '16

03 Oct '16

Hello, Thanks to all to develop the such type Sip Proxy server. Please suggest whether SIP proxy server is open source ,If yes than it can fulfil the my below requirement . 1) I having some sip clients which are registered on IMS .Can I use the kamailio Sip proxy sever between sip client and IMS. Sip client ------------------Kamailio Sip Proxy ---------------IMS 2) Second things, can I use Kamailio Sip Proxy to interconnect two IMS nodes . Can I create the SIP trunk without registration mode (username and password) between IMS nodes using Kamailio. Because the my IMS node doesn't support sip trunk in registration mode. Its support one normal trunk (SIP-T) My current IMS set up like below. IMS-1 (IBCF)---------------IMS2(IBCF). Regards Surender Singh ::DISCLAIMER:: ---------------------------------------------------------------------------------------------------------------------------------------------------- The contents of this e-mail and any attachment(s) are confidential and intended for the named recipient(s) only. E-mail transmission is not guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or may contain viruses in transmission. The e mail and its contents (with or without referred errors) shall therefore not attach any liability on the originator or HCL or its affiliates. Views or opinions, if any, presented in this email are solely those of the author and may not necessarily reflect the views or opinions of HCL or its affiliates. Any form of reproduction, dissemination, copying, disclosure, modification, distribution and / or publication of this message without the prior written consent of authorized representative of HCL is strictly prohibited. If you have received this email in error please delete it and notify the sender immediately. Before opening any email and/or attachments, please check them for viruses and other defects. ----------------------------------------------------------------------------------------------------------------------------------------------------

2 1

Kamailio number of concurrent calls
by Gholamreza Sabery 03 Oct '16

03 Oct '16

I used SIPp to stress test different scenarios in Kamailio, but how can I simulate a real call with media? SIPp has media sending capabilities but in not enough for example to simulate 1000 calls persecond! How can I test this?

5 7

How to check in Kamailio if a current sip message 200 OK is a response to the BYE.
by AFRICALLSHOP 01 Oct '16

01 Oct '16

Hello, I try to check in Kamailio if a current sip message "200 OK" is a response to a "BYE" request. Could you confirm the following code please? IF (IS_METHOD("BYE") && STATUS == 200) { } Regards Abdoul.

2 3

Proxy unable to connect - 420 Bad Extension error
by Infinicalls Infinicalls 01 Oct '16

01 Oct '16

Hi, I have setup a proxy and able to connect to the registrar. However, after a few attempts, it throws the "420:Bad Extension" error, saying "Unsupported: path. Path: <sip:10.1.0.4;lr>. Supported: outbound. P-Registrar-Error: No support for found Path indicated." Any idea, how to overcome this ? 10.1.0.4 is not my IP anywhere. I don;t know where it takes that value from. Thanks. regards Ganesh Kumar

2 2

Call forking priority
by Igor Potjevlesch 30 Sep '16

30 Sep '16

Hello list, I have a proxy that manage fix calls and a proxy that manage mobile calls. I want to implement call forking so when I call to a fix phone the call goes to the fix proxy and it forks to the mobile proxy who manage the call and viceversa. I want the fix to ring 2 or 3 times before the mobile. In order to do so I relay first to the fixed proxy, which will bounce back to mobile proxy the branch for mobile device. But now if I want to be able to choose which phone (mobile or fixe) rings before I have tried by doing the "async_route("RELAY", "7");" in the fixe proxy like in the mobile proxy but it doesn't work. My code is: Fix proxy: if($rU==123456789){ add_diversion("forking"); $var(priority)=1; append_branch("sip:987654321@proxy_mobile"); } route[INVITE]{ if($var(priority) == "1"){ async_route("RELAY", "7"); }else{ route(RELAY); exit; } Mobile proxy: if($tU==987654321){ $rU=123456789; } route[INVITE]{ if($dir == "forking"){ async_route("RELAY", "7"); }else{ route(RELAY); exit; } } Thanks.

2 2

ims_charging state machine issue
by 4fba81+8i7felu07050＠guerrillamail.com 30 Sep '16

30 Sep '16

Wanted to ask about changing rates while a call is not yet connected. I did this, $var(cc_ret1) = Ro_CCR_Stop("term", "487", "Request Terminated"); $var(cc_ret2) = Ro_CCR("RO_REPLY", "term", 30, "1", "1"); however, upon sending the new CCR, i get this ERROR: cdp [acctstatemachine.c:185]: cc_acc_client_stateful_sm_process(): Received unknown event [1] in state [1] and CRITICAL: ims_charging [ro_timer.c:122]: insert_ro_timer(): Trying to insert a bogus ro tl=0x7f484ab48728 tl->next=0x7f484ab42858 tl->prev=0x7f484aaa1090 looks like there is some statemachine issue. question is, how do we change rate? ---- Sent using GuerrillaMail.com Block or report abuse: https://www.guerrillamail.com/abuse/?a=UlM5EB4FUac1hAen5G1RYR3XANifx8NV1epN…

1 0

Identical Kamailio setup on 2 hosts
by Infinicalls Infinicalls 30 Sep '16

30 Sep '16

Hi, I have two identical hosts running Kamailio with the same set of users. They are located in different locations. Both are running behind NAT and I've enabled rtpproxy for that and also advertised the public IP. Both the MySQL DBs are similar and have the same data. Both the hosts have different sub-domains in alias and I have set auto_alias=no Now, the problem is my users are not able to communicate from one server to another. user1(a)mydomain.com from host1 is not able to communicate to user1(a)mydomain.com at host2, though inter-server communication is happening nicely. Any idea how to solve this? Thanks. regards Ganesh Kumar -- --- http://www.infinicalls.com

3 8

Load balancing traffic with Kamailio
by Nitesh Bansal 29 Sep '16

29 Sep '16

Hello, I'm planning on doing some smart load balancing with Kamailio. We have a distributed network, with multiple Kamailio boxes in different locations serving as Ingress SBC, these Kamailio boxes are the entry point for a SIP call and then they route the call to a pre-configured Asterisk boxes. I want to move away from this, I would like these Kamailios to be able to distribute the traffic to Asterisk boxes based on the performance of the Asterisk ,it would be great if we could take a look at SIP response time from Asterisk and use that as metric for load balancing the traffic) Is there any Kamailio module which could do that? Do I need to integrate some other tool with Kamailio to achieve this? Any suggestions are welcome. -- *Nitesh Bansal* Business Development Engineer http://www.voxbone.com/

3 3

Re: [SR-Users] Distributed Message Queue Module
by Charles Chance 29 Sep '16

29 Sep '16

Hello, That depends entirely on what you need to replicate between servers. DMQ is just a communication channel. How you use it is up to you. For example, if you enable dmq_usrloc, there's no need to share a location database. Htable has an option for replicating over dmq too. The number of direct module integrations like dmq_usrloc/htable is fairly limited, but you can also share arbitrary data between nodes yourself from within your config. I think you'll need to provide a little more information about what you're trying to achieve, however, in order for us to offer any further advice. Cheers, Charles On 29 Sep 2016 08:08, "Moacir Ferreira" <moacirferreira(a)hotmail.com> wrote: Hi, When using the DMQueue module what is the best approach, using one SQL server/database for each member node or having a single SQL server for all nodes? Thanks, Moacir _______________________________________________ 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 -- Sipcentric Ltd. Company registered in England & Wales no. 7365592. Registered office: Faraday Wharf, Innovation Birmingham Campus, Holt Street, Birmingham Science Park, Birmingham B7 4BB.

1 0

Distributed Message Queue Module
by Moacir Ferreira 28 Sep '16

28 Sep '16

Hi, When using the DMQueue module what is the best approach, using one SQL server/database for each member node or having a single SQL server for all nodes? Thanks, Moacir

1 0

memory problem
by beer Ll 28 Sep '16

28 Sep '16

hi all I have kamailio 4.2.0 on Debian Jessie the parameters on startup of kamailio are : SHM_MEMORY=512 PKG_MEMORY=32 the server is dedicated to the kamailio SIP service with authentication and accounting on a separated db server the server process about 30/50 call setups per second everythings works good but after 5/6 days the processes lock and I received a lot of error log like these: <core> [rvalue.c:1710]: rval_str_add2(): rv eval out of memory <core> [msg_translator.c:362]: received_builder(): out of memory <core> [msg_translator.c:2717]: via_builder(): out of memory ... <core> [msg_translator.c:2423]: build_res_buf_from_sip_req(): out of memory; needs 514 ... <core> [data_lump.c:172]: insert_new_lump_before(): out of memory acc [acc_extra.c:267]: extra2strar(): extra2strar: out of memory. after a kamailio reboot evrethings works again for 5/6 days. what can I do to resolve?

3 3

RPC tcp limit buffer
by pars3c 28 Sep '16

28 Sep '16

Hello, i'm trying to list the dialog (dlg_profile list) in the proxy using an RPC call to jsonrpc-s (dlg_profile list). But when there are many calls , kamailio send an error because the tcp buffer is not big enough. The only solution is to increase the parameter tcp_conn_wq_max every time i've an error ? Thanks

2 2

Accessing cached database table
by Daniel W. Graham 28 Sep '16

28 Sep '16

Is there a way to access a cached database table from the config file, without using a function from a module? Example - Lookup ip_addr from 'address' table or src_ip from 'trusted' table based on the 'tag' column. No function exists in permissions module to do this, but running 'kamctl address show ' or 'kamctl trusted show ', show that the tables along with tag column are cached. Thanks! -Dan

2 2

invalid sendtoparameters
by Linux Vince 27 Sep '16

27 Sep '16

I am getting this error on every outgoing request. Below are log entries i found Sep 27 22:09:06 centos1 /usr/sbin/kamailio[22421]: ERROR: <core> [udp_server.c:550]: udp_send(): sendto(sock,0x7fb76f5da5e0,1568,0, 10.3.4.2:5060,16): Invalid argument(22) Sep 27 22:09:06 centos1 /usr/sbin/kamailio[22421]: CRITICAL: <core> [udp_server.c:555]: udp_send(): invalid sendtoparameters#012one possible reason is the server is bound to localhost and#012attempts to send to the net Sep 27 22:09:06 centos1 /usr/sbin/kamailio[22421]: ERROR: <core> [forward.h:201]: msg_send_buffer(): udp_send failed SNDTO values are SNDTO ,10.3.4.2 PORT 5060 I have confirmed the PID and it is bound to my public IP 22417 main process - attendant 22418 udp receiver child=0 sock=10.3.4.1:5060 22419 udp receiver child=1 sock=10.3.4.1:5060 22420 udp receiver child=2 sock=10.3.4.1:5060 22421 udp receiver child=3 sock=10.3.4.1:5060 22422 udp receiver child=4 sock=10.3.4.1:5060 22423 udp receiver child=5 sock=10.3.4.1:5060 22424 udp receiver child=6 sock=10.3.4.1:5060 22425 udp receiver child=7 sock=10.3.4.1:5060 22426 udp receiver child=0 sock=127.0.0.1:5060 22427 udp receiver child=1 sock=127.0.0.1:5060 22428 udp receiver child=2 sock=127.0.0.1:5060 22429 udp receiver child=3 sock=127.0.0.1:5060 22430 udp receiver child=4 sock=127.0.0.1:5060 22431 udp receiver child=5 sock=127.0.0.1:5060 22432 udp receiver child=6 sock=127.0.0.1:5060 22433 udp receiver child=7 sock=127.0.0.1:5060 22434 slow timer 22435 timer 22436 secondary timer 22437 ctl handler 22438 MI FIFO 22439 WEBSOCKET KEEPALIVE 22440 tcp receiver (generic) child=0 22441 tcp receiver (generic) child=1 22442 tcp receiver (generic) child=2 22443 tcp receiver (generic) child=3 22444 tcp receiver (generic) child=4 22445 tcp receiver (generic) child=5 22446 tcp receiver (generic) child=6 22447 tcp receiver (generic) child=7 22448 tcp main process How do i identify the issue ?

1 1

SIREMIS MANUAL
by Carlyle Edwards 27 Sep '16

27 Sep '16

Hi, is there a manual for SIREMIS?

2 1

Kamailio IMS NAT issue: PRACK/ACK RURI changing in P-CSCF
by Owais 26 Sep '16

26 Sep '16

Hi all, My scenario is as follows: I have All the CSCFs on the same IP using different ports. P-CSCF advertises a public IP. Rest of the nodes including I-CSCF and S-CSCF listen local only. Behavior: When an ACK or PRACK goes from P-CSCF to S-CSCF, the RURI changes such that alias parameter is removed. Hence, P-CSCF ends up sending PRACK and ACK on private IP of UE. Here's what the config looks like: Any hints? -- View this message in context: http://sip-router.1086192.n5.nabble.com/Kamailio-IMS-NAT-issue-PRACK-ACK-RU… Sent from the Users mailing list archive at Nabble.com.

2 2

REGISTER between Kamailio 4.4 and Asterisk 13
by Grant Bagdasarian 26 Sep '16

26 Sep '16

Hello, We're trying to get our Asterisk test server to REGISTER with Kamailio. The first attempts are ok, but once the nonce timer is exceeded Kamailio starts rejecting the REGISTER requests from Asterisk, due to the nonce being expired. The auth_check function is the one returning error -4. Is this an issue in Asterisk or Kamailio? Which component is responsible for generating a new nonce value? I'm a bit confused regarding the flow of SIP REGISTER. I hope someone could help me out on this. Thanks! Regards, Grant Bagdasarian

2 2

Config changes needed from 4.4.2 to 4.4.3 (and a bug)
by Sebastian Damm 26 Sep '16

26 Sep '16

Hi, Kamailio release 4.4.3 was tagged as a "bug fix release", no config changes should be needed. However, we had to change two things in our config file to get 4.4.3 running. I thought, I might share those on the list. 1) Defines must not have '-' in the names. In 4.4.2 we had a Define like this: #!define NO-FRAUDCHECK This threw an error when trying to restart with 4.4.3. Replacing it with #!define NO_FRAUDCHECK fixed the problem. 2) Module Order problems. We introduced the usage of http_async_client module a few weeks ago, and this worked perfectly. However, 4.4.3 complained that the pv module was not available before loading http_async_client. After moving up the pv module above http_async_client, everything worked again. 3) There's a bug in uac_auth function. I've opened an issue for that: https://github.com/kamailio/kamailio/issues/797 Best Regards, Sebastian

2 1

Textops replace question
by Igor Olhovskiy 26 Sep '16

26 Sep '16

Hi! I’m trying to use replace function, but got strange results. Idea - modify Contact header What I got as start packet. (202 Accepted on onreply_route) … Contact: <sip:*97@10.0.20.81:5060> … I give a command replace("Contact: <.*>», "Contact: <" + $fu + ">"); $fu = sip:11@master.rufan.at And, as a result I got Contact: <sip:11@master.rufan.at>sip:*97@10.0.20.81:5060 Why it’s adding old Contact part at the end? What can cause this? On onreply_route I have only this command. Thanks! -- Best regards, Igor

3 4

Dialog sharing between Kamailio servers
by Michael Jepson 26 Sep '16

26 Sep '16

Hi all, I am trying to limit the concurrent calls of a customer, but we have multiple Kamailio servers in our platform. I have the dialog module up and running and it is storing its data in a database, but when I use get_profile_size, it will only return the number of dialogs in that profile for the current machine, it does not add the others. How can I get Kamailio to add the others, i.e. how can I get Kamailio to consult the database to get the correct size. Now, Kamailio seems to only write to the database, but not read from it. Best regards, MICHAEL JEPSON

2 1

Load balancing with Kamailio
by NITESH BANSAL 26 Sep '16

26 Sep '16

Hello, I'm planning on doing some smart load balancing with Kamailio. We have a distributed network, with multiple Kamailio boxes in different locations serving as Ingress SBC, these Kamailio boxes are the entry point for a SIP call and then they route the call to a pre-configured Asterisk boxes. I want to move away from this, I would like these Kamailios to be able to distribute the traffic to Asterisk boxes based on the actual load on these boxes, the goal is to be more dynamic? Is there any Kamailio module which could do that? Do I need to integrate some other tool(Homer etc) with Kamailio to achieve this? Any suggestions are welcome.

3 3

LCR routing, and finaly route to media gateway if not found
by Loic Chabert 26 Sep '16

26 Sep '16

Hello List, I'm finishing my implementation of LCR with kamailio for outgoing calls but one use case does not work. Let me explain: - I have some sbc for one destination. If one destination fail, kamailio try to second one etc.. - If no SBC is available for this destination (for exemple, all are down), i want to send my call to an error server (asterisk), play an error sound. To accomplish this, i re-write the RURI: "sip:error1@errorserver.local", and launch RELAY route. But kamailio dont make a DNS query to find IP of "errorserver.local", only change RURI and try to send to the last LCR gateway. I have also try to add a new branch, but same result... Why kamailio do not make a new DNS query ? *route[RELAY] { #if (is_method("INVITE")) { # if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); #} if (!t_relay()) { sl_reply_error(); } exit;}route[NOTFOUND]{ $ru = "sip:error1@errorserver.local"; xlog("L_WARN", "[$ci]|$si|Routing this call to $ru (error server)\n"); append_branch($ru); route(RELAY);}route[LCR] { xlog("L_INFO", "[$ci]|$si|Loading gateways...\n"); if (!load_gws(1)) { xlog("L_ERR", "[$ci]|$si|Can't load LCR gateways\n"); sl_send_reply("503", "Unable to load gateways"); exit; } else { $var(i) = 0; while(is_avp_set("$(avp(i:709)[$var(i)])")) { xlog("L_INFO", "[$ci]|$si|This gateway was found: gw_uri_avp[$var(i)]=$(avp(i:709)[$var(i)]) \n"); $var(i) = $var(i) + 1; }; if(is_avp_set("$avp(i:709)")) { xlog("L_INFO", "[$ci]|$si|Trying gateway ‘$avp(i:709)’\n"); } else { xlog("L_INFO", "[$ci]|$si|No more gateways. Goto route NOTFOUND...\n"); route(NOTFOUND); }; # try the first matched gateway if (next_gw()) { xlog("L_INFO", "[$ci]|$si|New request URI built: $ru . Relaying it (and arming backup destination if needed)."); # Route to failure for failover t_on_failure("lcrfailure"); route(RELAY); } else { xlog("L_INFO", "[$ci]|$si|No more gateways (after next_gw()). Goto route NOTFOUND..."); route(NOTFOUND); }; }; exit;}failure_route[lcrfailure] { # the previous gateway is no good if (t_check_status("408|50[34]")) { xlog("L_WARN", "[$ci]|$si|Entering in entering failure_route[lcrfailure]. Error code: $T_reply_code \n"); if(is_avp_set("$avp(i:709)")) { xlog("L_INFO", "[$ci]|$si|Trying gateway ‘$avp(i:709)’\n"); } else { xlog("L_ERR", "[$ci]|$si|AVP is not set\n"); }; if (next_gw()) { xlog("L_INFO", "[$ci]|$si|New request URI built: $ru . Relaying it (and arming backup destination if needed)."); # Route to failure for failover t_on_failure("lcrfailure"); route(RELAY); } else { xlog("L_INFO", "[$ci]|$si|No more gateways (after next_gw()). Goto route NOTFOUND..."); route(NOTFOUND); }; exit; };}* Many thanks, Regards.

3 2

how to config gateway in kamailio
by Gaurav Bmotra 26 Sep '16

26 Sep '16

Hi i want to config gateway in kamailio i want to know how can i config IP Authenticate gateway and user/ pass gateway -- *Regards:* Gaurav Bmotra

2 1

Scripting Issue
by aman jeff 26 Sep '16

26 Sep '16

Hai Team, I can't able to set maxptime in SDP Please help me

1 1

Location Table In db_mode 3
by Gholamreza Sabery 25 Sep '16

25 Sep '16

I have multiple Kamailio servers in an Active/Active scenario (the database is a MySQL server) using db_mode 3. But when the number of clients goes up writes on the location table become too much and it causes database problems. I even put my location table on a separate SSD disk on the server to separate it's IO but still at times I have problems. How can I move location table into memory? Is this a good solution? What other options do I have?

4 12

how to config gateway in kamailio
by Bmotra Infotech 24 Sep '16

24 Sep '16

i want to config gateway in kamailio i want to know how can i config IP Authenticate gateway and user/ pass gateway Bmotra Infotech

2 1

Re: [SR-Users] LCR routing, and finaly route to media gateway if not found
by Loic Chabert 23 Sep '16

23 Sep '16

In fact, kamailio is out from lcr at this point, no more gateway to try (all gateways are down). I just want to send call to a static server for playing sound. To be more specific, if no gateways are not set on database, a sound must be played. Should i change $du and $ru to make it work ? Le 23 sept. 2016 4:36 PM, "Juha Heinanen" <jh(a)tutpro.com> a écrit : Loic Chabert writes: > To accomplish this, i re-write the RURI: "sip:error1@errorserver.local", > and launch RELAY route. But kamailio dont make a DNS query to find IP of > "errorserver.local", only change RURI and try to send to the last LCR > gateway. > > I have also try to add a new branch, but same result... Why kamailio do not > make a new DNS query ? check if both hostname and ip address are given for the gateway. if so, next_gw() also sets destination uri. -- juha _______________________________________________ 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 ims nat issue: PRACK/ACK ruri changing in p-cscf
by Owais 23 Sep '16

23 Sep '16

Hi all, My scenario is as follows: I have All the CSCFs on the same IP using different ports. P-CSCF advertises a public IP. Rest of the nodes including I-CSCF and S-CSCF listen local only. Behavior: When an ACK or PRACK goes from P-CSCF to S-CSCF, the RURI changes such that alias parameter is removed. Hence, P-CSCF ends up sending PRACK and ACK on private IP of UE. Here's what the config looks like: #!define WITH_NAT Any hints? -- View this message in context: http://sip-router.1086192.n5.nabble.com/Kamailio-ims-nat-issue-PRACK-ACK-ru… Sent from the Users mailing list archive at Nabble.com.

1 0

604 Does not exist anywhere - HSS User Unknown
by Rodrigo Moreira 23 Sep '16

23 Sep '16

Hi, After I am registred on Kamailio a try to make a call (INVITE), and S-CSCF returns the messege: 604 Does not exist anywhere - HSS User Unknown What I am doing wrong? Is the contact entry (Alice) wrong? How should the contact entry on the list? sip:alice@mydomain.com Best Regards., -- Rodrigo M. (37) 9132-4539 (34) 9889-3069 rodrigo.moreira2007

3 3

Kamailio Not Starting
by Linux Vince 23 Sep '16

23 Sep '16

I am getting below error when i start kamailio Not starting : invalid configuration file! 0(2360) ERROR: <core> [mem/f_malloc.c:415]: fm_search_defrag(): fm_search_defrag(0x7f732c5ff010, 232); Free fragment not found! 0(2360) ERROR: <core> [mem/f_malloc.c:498]: fm_malloc(): fm_malloc(0x7f732c5ff010, 232) called from core: route_struct.c: mk_action(118), module: core; Free fragment not found! 0(2360) CRITICAL: <core> [route_struct.c:137]: mk_action(): memory allocation failure 0(2360) ERROR: <core> [mem/f_malloc.c:415]: fm_search_defrag(): fm_search_defrag(0x7f732c5ff010, 40); Free fragment not found! 0(2360) ERROR: <core> [mem/f_malloc.c:498]: fm_malloc(): fm_malloc(0x7f732c5ff010, 40) called from core: cfg.y: yyparse(2636), module: core; Free fragment not found! 0(2360) : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /etc/kamailio/proxy.cfg, line 5372, column 3-28: Not enough memory ERROR: bad config file (1 errors) I tried increasing SHM_MEMORY and PKG_MEMORY under /etc/default/kamailio I doubled memory but still getting same error. Is there any other place i should look at ?

2 3

Kamailio ims nat issue: PRACK/ACK ruri changing in p-cscf
by Owais Ahmad 23 Sep '16

23 Sep '16

Hi all, My scenario is as follows: I have All the CSCFs on the same IP using different ports. P-CSCF advertises a public IP. Rest of the nodes including I-CSCF and S-CSCF listen local only. Behavior: When an ACK or PRACK goes from P-CSCF to S-CSCF, the RURI changes such that alias parameter is removed. Hence, P-CSCF ends up sending PRACK and ACK on private IP of UE. Here's what the config looks like: listen=udp:lo.ca.l.ip:6000 advertise pu.bl.ic.ip:6000 alias=pcscf-ims.domain.net #!define WITH_NAT Any hints?

1 0

Problem reseting TMX statistics
by Átila de Moura Tavano Moretto 23 Sep '16

23 Sep '16

I'm currently using version 4.4.2 and am having trouble trying to reset the statistics in the TMX module. According to its documentation, all statistics except "inuse_transactions" can be reset. Checking the tmx_mod.c files for the statistics definition, I saw that the STAT_NO_RESET flag wasn't used for any statistic listed there (which seems odd because I think it should have been set for the inuse_transactions statistic): static stat_export_t mod_stats[] = { {"UAS_transactions", STAT_IS_FUNC, (stat_var**)tmx_stats_uas_trans }, {"UAC_transactions", STAT_IS_FUNC, (stat_var**)tmx_stats_uac_trans }, {"2xx_transactions", STAT_IS_FUNC, (stat_var**)tmx_stats_trans_2xx }, {"3xx_transactions", STAT_IS_FUNC, (stat_var**)tmx_stats_trans_3xx }, {"4xx_transactions", STAT_IS_FUNC, (stat_var**)tmx_stats_trans_4xx }, {"5xx_transactions", STAT_IS_FUNC, (stat_var**)tmx_stats_trans_5xx }, {"6xx_transactions", STAT_IS_FUNC, (stat_var**)tmx_stats_trans_6xx }, {"inuse_transactions", STAT_IS_FUNC, (stat_var**)tmx_stats_trans_inuse }, {"active_transactions", STAT_IS_FUNC, (stat_var**)tmx_stats_trans_active}, {"rpl_received", STAT_IS_FUNC, (stat_var**)tmx_stats_rcv_rpls }, {"rpl_absorbed", STAT_IS_FUNC, (stat_var**)tmx_stats_abs_rpls }, {"rpl_generated", STAT_IS_FUNC, (stat_var**)tmx_stats_rld_loc_rpls}, {"rpl_relayed", STAT_IS_FUNC, (stat_var**)tmx_stats_rld_rcv_rpls}, {"rpl_sent", STAT_IS_FUNC, (stat_var**)tmx_stats_rld_tot_rpls}, {0, 0, 0} }; All seems ok, but when I try to reset the stats, the command simply doesn't work. The values remain unaltered. The clear command doesn't work either, it just returns the current values. Can anyone help me trying to debug this problem, I'm very new to Kamailio and its code, and don't have a good understanding of how it the statistics gathering/reset works. Thanks. Atila Moretto

1 0

Location Table In db_mode 3
by Gholamreza Sabery 23 Sep '16

23 Sep '16

I have multiple Kamailio servers in an Active/Active scenario (the database is a MySQL server) using db_mode 3. But when the number of clients goes up writes on the location table become too much and it causes database problems. I even put my location table on a separate SSD disk on the server to separate it's IO but still at times I have problems. How can I move location table into memory? Is this a good solution? What other options do I have?

1 0

Dimension number of child
by Igor Potjevlesch 23 Sep '16

23 Sep '16

Hello, I'm wondering if there's a best practise to dimension the number of child to setup? I see, very often, Kamailio that stops to reply to SIP traffic for couple a seconds. This cause some calls drop if RE-INVITE are not processed for example. So, I'm wondering if the issue is not due to a lack of child for processing the SIP traffic. Thank you for your inputs. Regards, Igor.

1 0

failed to find command http_async_query - Even though module loaded
by Jonathan Hunter 23 Sep '16

23 Sep '16

Hi guys, I am using the http_async_client module to POST some requests to a Web Interface, and on server 1 it all works well. I have done a fresh install from github, version 4.4 on Centos 6.8 on server 2, and kamailio wont start. Its the same configuration file, however it states; failed to find command http_async_query (params 3) However the module is loaded, and when installing no errors were written when creating the module, and in terms of its links they are below and look ok to me. Is there anywhere else I should be looking? Thanks Jon ldd http_async_client.so linux-vdso.so.1 => (0x00007ffecedf1000) libcurl.so.4 => /usr/lib64/libcurl.so.4 (0x00007fdc5bd63000) libevent-2.0.so.5 => /usr/lib64/libevent-2.0.so.5 (0x00007fdc5bb1c000) libsrutils.so.1 => /usr/local/kamailio-4.4/lib64/kamailio/libsrutils.so.1 (0x00007fdc5b90c000) libkcore.so.1 => /usr/local/kamailio-4.4/lib64/kamailio/libkcore.so.1 (0x00007fdc5b6ff000) libc.so.6 => /lib64/libc.so.6 (0x00007fdc5b36a000) libidn.so.11 => /lib64/libidn.so.11 (0x00007fdc5b138000) libldap-2.4.so.2 => /lib64/libldap-2.4.so.2 (0x00007fdc5aee7000) librt.so.1 => /lib64/librt.so.1 (0x00007fdc5acde000) libgssapi_krb5.so.2 => /lib64/libgssapi_krb5.so.2 (0x00007fdc5aa9a000) libkrb5.so.3 => /lib64/libkrb5.so.3 (0x00007fdc5a7b3000) libk5crypto.so.3 => /lib64/libk5crypto.so.3 (0x00007fdc5a586000) libcom_err.so.2 => /lib64/libcom_err.so.2 (0x00007fdc5a382000) libz.so.1 => /lib64/libz.so.1 (0x00007fdc5a16c000) libssl3.so => /usr/lib64/libssl3.so (0x00007fdc59f28000) libsmime3.so => /usr/lib64/libsmime3.so (0x00007fdc59cfb000) libnss3.so => /usr/lib64/libnss3.so (0x00007fdc599bb000) libnssutil3.so => /usr/lib64/libnssutil3.so (0x00007fdc5978e000) libplds4.so => /lib64/libplds4.so (0x00007fdc5958a000) libplc4.so => /lib64/libplc4.so (0x00007fdc59385000) libnspr4.so => /lib64/libnspr4.so (0x00007fdc59146000) libpthread.so.0 => /lib64/libpthread.so.0 (0x00007fdc58f29000) libdl.so.2 => /lib64/libdl.so.2 (0x00007fdc58d25000) libssh2.so.1 => /usr/lib64/libssh2.so.1 (0x00007fdc58afc000) libm.so.6 => /lib64/libm.so.6 (0x00007fdc58878000) /lib64/ld-linux-x86-64.so.2 (0x00007fdc5c1f8000) liblber-2.4.so.2 => /lib64/liblber-2.4.so.2 (0x00007fdc58668000) libresolv.so.2 => /lib64/libresolv.so.2 (0x00007fdc5844e000) libsasl2.so.2 => /usr/lib64/libsasl2.so.2 (0x00007fdc58234000) libkrb5support.so.0 => /lib64/libkrb5support.so.0 (0x00007fdc58028000) libkeyutils.so.1 => /lib64/libkeyutils.so.1 (0x00007fdc57e25000) libssl.so.10 => /usr/lib64/libssl.so.10 (0x00007fdc57bb8000) libcrypto.so.10 => /usr/lib64/libcrypto.so.10 (0x00007fdc577d4000) libcrypt.so.1 => /lib64/libcrypt.so.1 (0x00007fdc5759d000) libselinux.so.1 => /lib64/libselinux.so.1 (0x00007fdc5737d000) libfreebl3.so => /lib64/libfreebl3.so (0x00007fdc5717a000)

2 2

Selective rport behaviour
by Alex Balashov 23 Sep '16

23 Sep '16

Hi, In some scenarios, it happens that a request is received with the following characteristics: IP xxx.xxx.xxx.xxx:14000 -> server:5060 Via: SIP/2.0/xxx.xxx.xxx.xxx:5060;branch=9hG4bKblah;rport Normally, we just force_rport() on all incoming requests so that we reply to the real source port of the request, since most endpoints on this installation are NAT'd. However, occasionally we run into a scenario where an ALG or misconfigured client incorrectly inserts an rport attribute into its topmost Via, and really expects to receive a response at the address/port indicated in the Via (i.e. 5060). Does Kamailio offer a means of dealing with these on a selective basis? Would refraining from calling force_rport() be enough? Or would it be necessary to set reply_to_via=1 as well, thus breaking symmetrical behaviour for the vast majority of the NAT'd endpoints? In other words, I'm not 100% clear on the following: 1) What impact does force_rport() have if an 'rport' attribute is sent by the client? In this case, there should be nothing to "force"; I assume that if the 'rport' attribute is placed by the client, then the proxy will return replies to the source port of the request even if force_rport() is not called, because that's the RFC 3581-compatible thing to do. Right? 2) Does reply_to_via=1 override the behaviour hypothesised in #1? 3) Does reply_to_via=1 override force_rport()? Thanks! -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

3 3

Killing dialog using kamcmd
by Laura 22 Sep '16

22 Sep '16

Dear list, i have developed a tool that automatically stop the call after some trigger i use.. For do that i use the kamcmd dlg.end_dlg command. It work perfectly but i have noticed a problem over the kamailio acc table. When i kill a specific call using this command.. the acc table don't write the BYE record for the dialog.. Is that correct ? how can i fix.. I'm using ngrep for verify the SIP dialog and i see the BYE command correctly on the A and B leg of the call.. Thanks for any idea. Laura

2 2

ndb_mongodb -Sending Query to Mongo
by Jonathan Hunter 22 Sep '16

22 Sep '16

Hi Guys, I am using the ndb_mongodb module, and Im just trying to return values from the dispatcher table, in particular the setid of a particular entry depending on its attrs value. If I query using mongodb shell command it works fine, here I enter test.com and it returns only the setid value, or 2; db.dispatcher.find( { attrs: "test.com" }, { setid: 1, _id: 0 } ){ "setid" : 2 } However I have been having issues getting this to work using mongodb_find_one, as I would of thought the following route return the same; if(mongodb_find_one("ml-mongodb", "kamailiosbc", "dispatcher", "{ \"attrs\" : \"test.com\" },{ \"setid\": \"1\", \"_id\": \"0\" }", "mgr1")) However it returns all the entries relating to that attrs columm from dispatcher table, instead of just setid. Is it possible to do this? I am stuggling to find the json format to acheive the same thing. Also in terms of the mgr1 response, I see its contained within the $mongodb(key), can I drill down to get specific fields from it or would I need to use transformations or jansson to do this? Many thanks Jon

2 2

STUN server - stun_allow_stun
by Hermann Norpois 22 Sep '16

22 Sep '16

Hello, I have some STUN questions. My plan is to build a P2P connection with ICE (psua2 or libnice). 1) According to the cookbook ( http://www.kamailio.org/wiki/cookbooks/4.0.x/core) one can activate an internal STUN server: stun_allow_stun = 0 | 1 (off | on); default 1 This is followed by: "Note: STUN support is only available if sip-router/Kamailio was compiled with STUN support (“make STUN=1 cfg”)" I did an installation with sudo apt-get ... Is STUN support availbable? How do I check for STUN support? 2) I ICE it seems to be usual that the body of the SIP request contains an SDP message that includes the IP addresses and ports to be used. So the results of the STUN request (and TURN request possibily) are already included. How does the kamailio STUN server work? Is it an independent STUN server (caller asks the kamailio STUN server - gets an answer - encodes an SDP with the external IP adress an port information - starts the SIP INVITE) ? Or does Kamailio change SDP lines after contacting the internal STUN server? 3) Is the kanailio STUN server prefered to other STUN servers, for instance stun.stunprotocol.org? Thanks Hermann

2 1

Problem using ds_is_from_list in failure_route
by Joel Serrano | VOZELIA 21 Sep '16

21 Sep '16

Hi all, I'm facing a strange problem using ds_is_from_list in the failure_route block. I have 4 servers in a dispatcher group 8201. Scenario: A call comes in, gets routed to a backend server from 8201 dispatcher group, the backend replies with a 480. Then we have: onreply_route[MANAGE_REPLY] { [...] if (ds_is_from_list("8201")) && t_check_status("480") { <<<<<< This gets executed, because the node is in 8201 and the reply was a 480 if (is_present_hf("X-UserID")) { $var(UserId) = $hdr(X-UserID); } } [...] } So far so good, next is: failure_route[MANAGE_FAILURE] { [...] if (ds_is_from_list("8201")) && (t_check_status("480")) { <<<<<< This does NOT get executed. append_hf("X-UserID: $var(UserId)\r\n"); if (!ds_select_domain("8205","4")) { xlog("L_ERR", "[$ci] - No additional gateways\n"); t_reply("502","Bad Gateway"); exit; } t_set_fr(0,2000); route(RELAY); } [...] } To find the problem, I added some logging in the failure_route before entering the IF statement: if (ds_is_from_list("8201")) xlog("L_NOTICE", "DS is in group 8201"); <<<<< No log, so did NOT match. if (t_check_status("480")) xlog("L_NOTICE", "Status is 480"); <<<<< Yes log, so DID match. According to the docs, ds_is_from_list can be used in any route block, so I think I am definitely doing something wrong... or maybe it's a bug?? The idea is to achieve this: Call comes in from softphone into proxy, proxy sends it to backend, if backend replies with a 480, send it to another dispatcher set (8205). If the 480 comes from somewhere that is not the backend (8201), then do nothing. Any suggestions? Thanks in advanced. Best regards, Joel. Este mensaje y, en su caso, los ficheros anexos son confidenciales, especialmente en lo que respecta a los datos personales, y se dirigen exclusivamente al destinatario referenciado.Les informamos que los datos contenidos en el presente mail, han sido facilitados por usted o recogidos de Fuentes Accesibles al Público. Dicha información personal, será incluida en un fichero bajo la responsabilidad de VOZELIA TELECOM S.L.Todos los ficheros que trata nuestra empresa se encuentran debidamente inscritos ante la Agencia Española de Protección de Datos.Vd. Como propio interesado podrá ejercitar sus derechos de Acceso, Cancelación, Rectificación u Oposición, en el mail: administracion(a)vozelia.com o en la dirección Avenida de Manoteras 8 escalera 3 2ºB, 28050 Madrid.Tal y como establece la Ley Orgánica de Protección de Datos 15/1999 de 13 de Diciembre.Para ejercer sus derechos desde Panama podrá hacerlo en la dirección Vía Argentina, Edificio Mediteranean Loft. Panama.En caso de no ser el destinatario y haya recibido este mensaje por error, agradeceremos que nos lo comunique inmediatamente al remitente sin difundir, almacenar o copiar su contenido.La divulgación o el suministro, en todo o en parte, a cualquier tercero, no podrá ser realizada sin el previo, expreso y escrito consentimiento de VOZELIA TELECOM S.L. .

2 3

Error 2002 (HY000) creating database kamailio
by Diogenes Marcano 21 Sep '16

21 Sep '16

Iam new in kamailio. When I try to create the data base with kamdbctl y have the following error root@diogenes-Inspiron-1525:/usr/local/sbin# kamdbctl create MySQL password for root: INFO: test server charset ERROR 2002 (HY000): Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2 "No such file or directory") ERROR 2002 (HY000): Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2 "No such file or directory") Usage: grep [OPTION]... PATTERN [FILE]... Try 'grep --help' for more information. /usr/local/lib/kamailio//kamctl/kamdbctl.mysql: line 111: [: =: unary operator expected INFO: creating database kamailio ... ERROR 2002 (HY000): Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2 "No such file or directory") ERROR: Creating database kamailio failed! root@diogenes-Inspiron-1525:/usr/local/sbin# mysql ERROR 2002 (HY000): Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2 "No such file or directory") Thank you for your hrlp Diogenes Marcano

2 1

parameter not found in module dialog
by Michael Jepson 21 Sep '16

21 Sep '16

Hi all, I am trying to limit concurrent calls, using dialog, or rather, I want to try it. I have now only added loadmodule "dialog.so" and a series of modparam commands to my config file and already I am running into problems: ERROR: <core> [modparam.c:139]: set_mod_param_regex(): parameter <toroute_column> of type <1> not found in module <dialog> It fails on the last modparam line, but if I remove it (by commenting it), it fails on whatever the latest modparam line is after that. I have tried removing multiple calls, but every time it fails on whatever is the last line. I have the required modules already loaded (tm.so, rr.so and pv.so), so that's not it I guess. Any idea what might be causing this issue? Also, I noticed (when calling locate dialog.so on the Linux machine) that there is also an ims_dialog.so, if I replace dialog with ims_dialog, I run into the exact same issues. Any help is greatly appreciated! Best regards, Michael Jepson CM Telecom

2 1

Kamailio invite fails with 403 at cscf
by Owais Ahmad 20 Sep '16

20 Sep '16

I am using kamailio 4.4.2 IMS Registration completes successfully, but S-CSCF gives the following error when P-CSCF relays INVITE to S-CSCF: 32(1349) ERROR: ims_dialog [dlg_handlers.c:190]: populate_leg_info(): bad sip message or missing Contact hdr 32(1349) ERROR: ims_dialog [dlg_handlers.c:1352]: dlg_onreply(): could not add further info to the dlg out Resultantly, P-CSCF returns a 403 with: "Forbidden - You must register first with a S-CSCF" It appears that pcscf_is_registered("location") is unable to lookup the location although the database seems to have valid entries. Any hints on how I can get the INVITE processed without this issue? Regards, Owais

3 4

.NET Core & Kamailio 5.0
by Grant Bagdasarian 20 Sep '16

20 Sep '16

Hi, I was reading some docs on Kamailio 5 and saw in the ToDo list an item about implementing the dynamic object mechanism for Mono. Would it be possible to implement it directly for .NET Core, since the official .NET support has been added to Linux some time ago? Regards, Grant

1 0

event_route[uac:reply] question
by Julia Boudniatsky 20 Sep '16

20 Sep '16

I send INVITE via uac_send_req() and receive "183" and "200"responses. Only "200" appears in the log of event_route[uac:reply]. Is event_route[uac:reply] executed only for final responses? Thank you, Julia

3 5

OOM Errors
by Phil Lavin 20 Sep '16

20 Sep '16

Hi all, Kamailio stopped performing some functions earlier today, citing OOM errors. An excerpt from the log is below. SHM usage, at the time, looked fine. Also see below for that output. A restart of Kamailio resolved the issue. Is this SHM related or OS memory manager related? root@ua-proxy-01:~# kamctl stats shmem shmem:fragments = 910 shmem:free_size = 104131224 shmem:max_used_size = 51590800 shmem:real_used_size = 30086504 shmem:total_size = 134217728 shmem:used_size = 19558800 Sep 19 11:35:56 ua-proxy-01 /sbin/kamailio[20218]: ERROR: <core> [data_lump.c:161]: insert_new_lump_before(): out of memory Sep 19 11:40:10 ua-proxy-01 /sbin/kamailio[20218]: ERROR: <core> [msg_translator.c:2390]: build_res_buf_from_sip_req(): out of memory; needs 423 Sep 19 11:40:16 ua-proxy-01 /sbin/kamailio[20218]: ERROR: <core> [msg_translator.c:2390]: build_res_buf_from_sip_req(): out of memory; needs 453 Sep 19 11:42:46 ua-proxy-01 /sbin/kamailio[20218]: ERROR: <core> [data_lump.c:136]: insert_new_lump_after(): out of memory Sep 19 11:42:46 ua-proxy-01 /sbin/kamailio[20218]: ERROR: <core> [ut.c:163]: as_asciiz(): Out of memory Sep 19 11:42:57 ua-proxy-01 /sbin/kamailio[20218]: ERROR: <core> [ut.c:163]: as_asciiz(): Out of memory Sep 19 11:43:12 ua-proxy-01 /sbin/kamailio[20218]: ERROR: <core> [data_lump.c:361]: anchor_lump(): out of memory Sep 19 11:43:12 ua-proxy-01 /sbin/kamailio[20218]: ERROR: <core> [data_lump.c:361]: anchor_lump(): out of memory Sep 19 11:43:12 ua-proxy-01 /sbin/kamailio[20218]: ERROR: <core> [data_lump.c:306]: del_lump(): out of memory Sep 19 11:43:12 ua-proxy-01 /sbin/kamailio[20218]: ERROR: <core> [data_lump.c:306]: del_lump(): out of memory Sep 19 11:43:12 ua-proxy-01 /sbin/kamailio[20218]: ERROR: <core> [msg_translator.c:326]: received_builder(): out of memory Sep 19 11:43:12 ua-proxy-01 /sbin/kamailio[20218]: ERROR: <core> [msg_translator.c:326]: received_builder(): out of memory Sep 19 11:43:12 ua-proxy-01 /sbin/kamailio[20218]: ERROR: <core> [msg_translator.c:326]: received_builder(): out of memory Sep 19 11:43:16 ua-proxy-01 /sbin/kamailio[20218]: ERROR: <core> [data_lump.c:136]: insert_new_lump_after(): out of memory Sep 19 11:43:16 ua-proxy-01 /sbin/kamailio[20218]: ERROR: <core> [ut.c:163]: as_asciiz(): Out of memory Sep 19 11:44:18 ua-proxy-01 /sbin/kamailio[20218]: ERROR: <core> [data_lump.c:161]: insert_new_lump_before(): out of memory Sep 19 11:45:07 ua-proxy-01 /sbin/kamailio[20218]: ERROR: <core> [ut.c:163]: as_asciiz(): Out of memory Sep 19 11:45:30 ua-proxy-01 /sbin/kamailio[20218]: ERROR: acc [acc_extra.c:267]: extra2strar(): extra2strar: out of memory. Sep 19 11:45:43 ua-proxy-01 /sbin/kamailio[20218]: ERROR: <core> [data_lump.c:186]: insert_subst_lump_after(): out of memory Sep 19 11:45:52 ua-proxy-01 /sbin/kamailio[20218]: ERROR: <core> [data_lump.c:161]: insert_new_lump_before(): out of memory

2 10

Presence: Empty reginfo NOTIFY bodies
by Stefan Kohlhauser 19 Sep '16

19 Sep '16

Hey everyone! Problem: At startup the presence module for reginfos can get into a state where it is not adding bodies to reginfo NOTIFYs. Setup: We are using the pua_reginfo to let the Kamailio (4.2.6) PUBLISH the current registration state to itself. The body is correctly added to the presentity-table in the Postgres. Analysis: However, we have identified two cases where after startup the Kamailio can reach a state where he won't add bodies to the NOTIFYs. And this does not fix itself over time. We are pretty confident that the problem lies in an inconsistency between the presentity table in the DB and the hashtable of the presence module. Case 1: The entry for a URI in the DB is expired. On startup of the Kamailio the hashtable remains empty for that URI. Maybe due to another bug the pua_reginfo occasionally sends PUBLISHes with an expired "SIP-If-Match"-header. In module/presence/presentity.c::update_presentity() the DB is updated. However, the hashtable is only updated if it is a new request (i.e. no "SIP-If-Match"-header and thus new_t not set). The hash table is not filled on startup, not on the first PUBLISH as it contains a (wrong) "SIP-If-Match"-header, and not on any subsequent PUBLISH as they contain correct "SIP-If-Match"-headers. Case 2: The presentity table contains at least one expired and one non-expired reginfo entry for a URI. The Kamailio seems to send a NOTIFY with a body within about a minute (probably the default clean period of 100 secs?). But after that short interval only NOTIFYs with empty bodies are sent. To us it looks like the cleanup function (module/presence/publish.c::msg_presentity_clean()) is cleaning up the expired DB entries correctly. But despite one still valid entry in the DB the entry in the hash table gets removed as well. And it wont get filled again (see case 1). This only seems to apply to the "reg" event, the "presence" event does not seem to be affected. We are aware aware of the publ_cach parameter, but we don't know if this has any performance impact on our system. If needed we can try to provide more insight on this problem. Some help on this or a fix would be much appreciated! BR, Stefan

1 0

Kamailio P-CSCF error on INVITE
by Owais 19 Sep '16

19 Sep '16

Hi guys, Kamailio P-CSCF is unable to return true for: pcscf_is_registered("location") on INVITE and responds to the INVITE with: 403 Forbidden - You must register first with a S-CSCF Registration completes fine and I can see database with valid location entries though... Any ideas? Regards, Owais -- View this message in context: http://sip-router.1086192.n5.nabble.com/Kamailio-P-CSCF-error-on-INVITE-tp1… Sent from the Users mailing list archive at Nabble.com.

2 3

how to access reply headers
by mjbemail2000-seti＠yahoo.co.uk 19 Sep '16

19 Sep '16

Hi, I have a registered handset which replies to an INVITE message from Kamailio with a "302 Moved Temporarily" message. The reason for this is because the user has set their handset to forward the call to another telephone number. In a wireshark trace I can see the forwarded telephone number in the Contact header of the SIP message but I can't find a way to access this header via any variables. In the config file I have tried this: # Manage incoming repliesonreply_route[MANAGE_REPLY] { ... if(t_check_status("3[0-9][0-9]")) { xlog("L_INFO", "test1: $hdr(Contact)"); xlog("L_INFO", "test2: $ct"); } ... } , but in both cases the value returned is the Contact header of the original INVITE message and not the Contact shown in the "302 Moved Temporarily" reply message. How do I access the Contact header in the reply message? Regards, Mark.

2 1

Debian 4.4.3 reproducable crashes in rtpengine.so
by Daniel Tryba 19 Sep '16

19 Sep '16

With kamctl fifo nh_enable_rtpp udp:rptipaddr:7723 1 kamailio will begin to crash: Sep 19 16:00:08 sipcluster-backend2 kernel: [1658593.859081] kamailio[22992]: segfault at 4 ip 00007f5a4d44e11f sp 0000 7ffcac5293e0 error 4 in rtpengine.so[7f5a4d446000+41000] Sep 19 16:00:12 sipcluster-backend2 /usr/sbin/kamailio[22999]: CRITICAL: <core> [pass_fd.c:275]: receive_fd(): EOF on 2 8 Sep 19 16:00:12 sipcluster-backend2 /usr/sbin/kamailio[22976]: ALERT: <core> [main.c:740]: handle_sigs(): child process 22992 exited by a signal 11 Sep 19 16:00:12 sipcluster-backend2 /usr/sbin/kamailio[22976]: ALERT: <core> [main.c:743]: handle_sigs(): core was gene rated Sep 19 16:00:12 sipcluster-backend2 /usr/sbin/kamailio[22976]: INFO: <core> [main.c:755]: handle_sigs(): terminating du e to SIGCHLD Sep 19 16:00:12 sipcluster-backend2 /usr/sbin/kamailio[22978]: INFO: <core> [main.c:810]: sig_usr(): signal 15 received .... Backtrace, Program terminated with signal SIGSEGV, Segmentation fault. #0 send_rtpp_command (node=node@entry=0x7f5a522b2140, dict=dict@entry=0x7f5a644b97c8, outlen=outlen@entry=0x7ffcac5294ec) at rtpengine.c:2619 2619 rtpengine.c: No such file or directory. (gdb) thread apply all bt full Thread 1 (Thread 0x7f5a66a49700 (LWP 22992)): #0 send_rtpp_command (node=node@entry=0x7f5a522b2140, dict=dict@entry=0x7f5a644b97c8, outlen=outlen@entry=0x7ffcac5294ec) at rtpengine.c:2619 addr = {sun_family = 12616, sun_path = "A\000\000\000\000\000t\326_", '\000' <repeats 13 times>, "\240ZGMZ\177\000\000`_GMZ\177\000\000\360\224R\254\374\177\000\000\a\000\000\000\000\000\000\000\374\257DMZ\177\000\000(\230KdZ\177\000\000\214\260DMZ\177\000\000\004\000\000\000\000\000\000\000\374\257DM\001\000\000\000\360\224R\254\374\177\000\000(\230KdZ\177"} fd = <optimized out> len = 0 i = <optimized out> vcnt = 6 cp = 0x7f5a4d687dc0 <buf> "" buf = '\000' <repeats 65535 times> fds = {{fd = 223, events = 0, revents = 0}} v = 0x7f5a644b98f6 out = {s = 0x0, len = 0} __FUNCTION__ = "send_rtpp_command" #1 0x00007f5a4d44f81c in rtpp_test_ping (node=0x7f5a522b2140) at rtpengine.c:1041 bencbuf = {pieces = 0x7f5a644b97b0, free_list = 0x0, error = 0} dict = 0x7f5a644b97c8 cp = <optimized out> ret = 0 #2 0x00007f5a4d454d89 in mi_enable_rtp_proxy (cmd_tree=0x1, param=0x1) at rtpengine.c:1137 crt_node = 0x0 rtpp_list = 0x7f5a522b1f80 root = 0x16 attr = 0x0 found = 0 __FUNCTION__ = "mi_enable_rtp_proxy" #3 0x00007f5a515f071e in run_mi_cmd (cmd=<optimized out>, cmd=<optimized out>, t=<optimized out>) at ../../lib/kmi/mi.h:77 No locals. #4 mi_fifo_server (fifo_stream=0x27be8a0) at fifo_fnc.c:509 line_len = 39 file_sep = 0x0 command = 0x7f5a644a2b41 "nh_enable_rtpp" f = 0x276f838 reply_stream = 0x27c4cf0 __FUNCTION__ = "mi_fifo_server" #5 0x00007f5a515e6345 in fifo_process (rank=1) at mi_fifo.c:270 fifo_stream = 0x27be8a0 __FUNCTION__ = "fifo_process" #6 0x00007f5a515e8f1e in mi_child_init (rank=0) at mi_fifo.c:234 pid = 0 #7 0x0000000000512ba9 in init_mod_child (m=0x1, rank=1) at sr_module.c:921 __FUNCTION__ = "init_mod_child" #8 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #9 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #10 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #11 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #12 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #13 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #14 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #15 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #16 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #17 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #18 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #19 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #20 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #21 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #22 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #23 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #24 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #25 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #26 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #27 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #28 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #29 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #30 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #31 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #32 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #33 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #34 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #35 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #36 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #37 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #38 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #39 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #40 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #41 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #42 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #43 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #44 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #45 0x0000000000510f18 in main_loop () at main.c:1696 si_desc = "udp receiver child=0 sock=127.0.0.1:5060\000\060\066\060\000\000\000\000\270\357DdZ\177\000\000\017\000\000\000\000\000\000\000n\000\000\000\000\000\000\000\255\370_\000\000\000\000\000\071\222\343p\000\000\000\000\000\071X\207\335c\334e\000\000\000\000\000\000\000\000\000\071X\207\335c\334e\210\263\246\000\000\000\000\000\005\000\000\000\000\000\000" woneinit = 1 __FUNCTION__ = "main_loop" #46 0x000000000041cefc in main (argc=3, argv=0x0) at main.c:2631 cfg_stream = 0x0 r = 0 tmp = 0x7ffcac52bee5 "" tmp_len = 0 port = 0 proto = 1722066384 ret = 0 seed = 2123298762 debug_save = 0 debug_flag = 8 dont_fork_cnt = 0 st = {st_dev = 15, st_ino = 12771, st_nlink = 2, st_mode = 16832, st_uid = 112, st_gid = 118, __pad0 = 0, st_rdev = 0, st_size = 100, st_blksize = 4096, st_blocks = 0, st_atim = {tv_sec = 1472635951, tv_nsec = 983999907}, st_mtim = {tv_sec = 1474293427, tv_nsec = 27546593}, st_ctim = {tv_sec = 1474293427, tv_nsec = 27546593}, __glibc_reserved = {0, 0, 0}} __FUNCTION__ = "main" (gdb)

1 0

yum install modules
by mjbemail2000-seti＠yahoo.co.uk 19 Sep '16

19 Sep '16

Hi, I've installed Kamailio on CentOS using yum but it's not installed the geoip2 module, amongst others. I'm I correct in saying that to install this module I'd need to install Kamailio from the source code? Regards, Mark.

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tls with ubuntu 16.04
by Jayesh Nambiar 19 Sep '16

19 Sep '16

Hi, Trying kamailio with Ubuntu 16.04 and I'm getting errors as follows: ERROR: tls [tls_init.c:490]: tls_pre_init(): Unable to set the memory allocation functions The error is identical to the one mentioned here: https://bugs.launchpad.net/ubuntu/+source/kamailio/+bug/1591992 Is there a solution to this. Thanks for any response. - Jayesh

2 21

Extract param from RPID header
by Grant Bagdasarian 19 Sep '16

19 Sep '16

Hello, I'm trying to extract the privacy param from the RPID header, but without success. Could someone point me in the right direction? Remote-Party-ID: "Cullen Jennings" <sip:+3123456789@1.1.1.1>;party=calling;privacy=off;screen=no. I've tried the following ways: 1) TEXTOPSX: $var(rpid_privacy_value) = @hf_value.remote_party_id.privacy; 2) Param Transformations: a. $(hdr(Remote-Party-ID){param.value,privacy}) b. $(re{param.value,privacy}) c. 4re{param.value,privacy} I can't seem to extract the value of privacy from the RPID header. Regards, Grant

2 2

Self challenged packets
by Torsten Hantzsche 19 Sep '16

19 Sep '16

Hi, while playing with the script from http://kb.asipto.com/kamailio:presence:k43-blf and "kamailio 4.4.3 (x86_64/freebsd) 5a2195" I could not see any NOTIFY packet going out to the subscribed watcher. "tcpdump -i lo0" shows self injected PUBLISH packets challenged with "407 Proxy Authentication Required" and there they got stuck. The PUBLISH packets are sent on lo0 from my network address to the same network address (here 10.0.0.106 to 10.0.0.106). The proxy challenge is also sent from 10.0.0.106 to 10.0.0.106. An xlog entry with the variable "$Ri" in route REQUEST shows 10.0.0.106 too. Since these packets are only seen on lo0, should "$Ri" not show 127.0.0.1? After disabling the WITH_AUTH variable, the watcher is getting its NOTIFY's. Now I wonder under which circumstances this script is supposed to function correctly without changes in the AUTH route? By the way, I had the same problem with the provided script of the msilo module (http://kamailio.org/docs/modules/4.4.x/modules/msilo.html#idp42867388) which too only works if there is no "auth_challenge" in the way. Could somebody with more insight in Kamailio internals give some hints on this, please? How can it get solved in an elegant way? Thanks, Torsten

3 3

Re: [SR-Users] help with kamailio rpm made from source
by Dmitry 18 Sep '16

18 Sep '16

Hello I tested a package http://download.opensuse.org/repositories/home:/kamailio:/v4.3.x-rpms/CentO… ( I downlowaded 4.3.6 version rpm and installed it. Kamailio restarts well. The behaviour is the same. the phone registeres without nonce. I install kamailio, kamailio-kazoo - 2 rpm зфслфпуы On Thursday, September 15, 2016 9:34 PM, Dmitry <mbike2000ru(a)yahoo.com> wrote: I made loadmodule and modparam("debugger", "cfgtrace", 1) but anyway - no logs when I register. As I understand - it's like no config file. On Thursday, September 15, 2016 6:01 PM, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: I am not familiar with kazoo configs, maybe asking on their mailing list can help you more. From Kamailio point of view, you can load debugger module and set its cfgtrace parameter to 1, then see what actions from config are executed and why is not getting to the authentication part. Cheers, Daniel On 15/09/16 14:09, Dmitry wrote: /etc/kazoo/kamailio/default.cfg - which containes all routes. 2600hz/kazoo-configs | | | | | | | | | | | 2600hz/kazoo-configs kazoo-configs - Kazoo Configuration Files for Software We Use | | | | I test on a working server (testing one) and a working config -- Daniel-Constantin Mierla http://www.asipto.com - http://www.kamailio.org http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda On Thursday, September 15, 2016 3:56 PM, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: Are you using default kamailio.cfg or another one? Cheers, Daniel On 15/09/16 12:39, Dmitry wrote: Hello I took this spec from suse. It generates no errors. When I installed from the RPM I had made - the phone register, but The phone sends a REGISTER and the KAmailio sends 200ok back to the phone (so no NONCE authorization) and no logs during it. In default.cfg I set L_DBG but no logs are generated during registering a phone. And I see no AMQP messages (I set to 1 Kazoo_enable in the SPEC file). Why there are no logs during register (I saw register in tcpdump)? On Wednesday, September 14, 2016 7:42 PM, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: Then you just need add those files in various packages inside the spec file, so they are not detected to be orphaned. Maybe you can inspire from: - https://build.opensuse.org/package/view_file/home:kamailio:v4.3.x-rpms/kama… Cheers, Daniel On 14/09/16 16:35, Dmitry wrote: 4.3.4 version is for Kazoo It is on production server currently. I need to rebuild the current RPM so as to apply patches. But first I want to get a working Kamailio and only after it I will apply the patches. I think I may take a list of modules from the production Kazoo-kamailio and rearchive the tar.gz because now the Fedora 18 rpmbuild gives a lot of files: RPM build errors: Installed (but unpackaged) file(s) found: /usr/lib64/kamailio/modules/auth_xkeys.so /usr/lib64/kamailio/modules/dmq_usrloc.so /usr/lib64/kamailio/modules/ims_charging.so /usr/lib64/kamailio/modules/jsonrpc-s.so /usr/lib64/kamailio/modules/mohqueue.so /usr/lib64/kamailio/modules/nosip.so /usr/lib64/kamailio/modules/rtjson.so /usr/lib64/kamailio/modules/rtpengine.so /usr/lib64/kamailio/modules/statsd.so /usr/lib64/kamailio/modules/tcpops.so /usr/lib64/kamailio/modules/tsilo.so /usr/share/doc/kamailio/modules/README.auth_xkeys /usr/share/doc/kamailio/modules/README.dmq_usrloc /usr/share/doc/kamailio/modules/README.ims_charging /usr/share/doc/kamailio/modules/README.ims_registrar_scscf /usr/share/doc/kamailio/modules/README.jsonrpc-s /usr/share/doc/kamailio/modules/README.mohqueue /usr/share/doc/kamailio/modules/README.nosip /usr/share/doc/kamailio/modules/README.rtjson /usr/share/doc/kamailio/modules/README.rtpengine /usr/share/doc/kamailio/modules/README.statsd /usr/share/doc/kamailio/modules/README.tcpops /usr/share/doc/kamailio/modules/README.tsilo /usr/share/snmp/mibs/KAMAILIO-MIB /usr/share/snmp/mibs/KAMAILIO-REG-MIB /usr/share/snmp/mibs/KAMAILIO-SIP-COMMON-MIB /usr/share/snmp/mibs/KAMAILIO-SIP-SERVER-MIB /usr/share/snmp/mibs/KAMAILIO-TC On Wednesday, September 14, 2016 5:05 PM, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: Hello, any reason not to use series 4.4.x? Iirc, the latest spec that got update on 4.4 are those for oracle enterprise linux, perhaps is something that you can reuse a lot for upgrading to the centos flavour. On the other hand, you can use opensuse build service if you want to build yourself, the spec from the kamailio project there is up to date and the ones for 4.3 should still be there. Anyhow, as I said, I would recommend to go with 4.4.x version. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com - http://www.kamailio.org http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 1

avp_db_query
by Angel Elena 18 Sep '16

18 Sep '16

Hi, good moorning. I have a Kamailio 4.0 with avp module working. Now, I need write some values in my database and I don't find and example for this. if (avp_db_query("update subscriber set bloqueado=0 where name=xxxxx")) { } Is possible make this operation with avp_db_query ? Thanks -------------------------------- Ángel Elena Medina _o) craem(a)craem.net / \\ http://blog.craem.net _(___V @craem_ --------------------------------

2 2

Replacing Asterisk with Kamailio
by Valter Nogueira 17 Sep '16

17 Sep '16

I use Asterisk for SIP and Media Proxy. Despite the fact that Asterisk is not a SIP Proxy at all. Customer registers in a SIP account, sends the invite and thru de context Asterisk dials out thru a SIP Trunk. Asterisk does the media proxy, since customer can't route directly to the SIP Trunk (altough it has a valida address, it don't have a public route allowed to it). I need limit customer concurrent calls, mangle some dial-in/dial-out numbers, keep track of ongoing call, control SIP dialog, retransmit correct hang-up causes and do media proxy (no transconding at all) After reading about Kamailio and Opensips, and due to the Kamailio Admin Book, I decided to go with Kamailio. Well, I understand that I have to use some kamailio modules, like auth, dialplan, rtpproxy and db_mysql. What make me stuck is how does everything fit together in kamailio.cfg and how do I get ongoing calls and CDR's? Can anyone point me a direction? Thanks

6 12

Commercial SBC or Kamailio
by Linux Vince 16 Sep '16

16 Sep '16

We are trying to setup VOIP infrastructure, mainly wholesale and retail. What is best option if money is not the problem? Developing our own infrastructure using Kamailio and other open source packages to act as SBC and switch or buy a commercial solution like GenBand/Sonus/Sansay. We are looking for high performance with scalability to handle thousands of call setups per second over the period of few years. I have no idea on how/if can kamailio outperform commercial solutions or not. Advantage of using kamailio is flexibility and possibility of customization as per requirement, I am new to this list so please advise if this is not a valid question to be asked here.

7 7

Kamailio fails to start when MySQL down
by Daniel Pocock 16 Sep '16

16 Sep '16

I recently set up a Kamailio instance using the default configuration for HOMER with MySQL[1] The database was not running and Kamailio refused to start. Would it be better for Kamailio to start anyway and go into a loop trying to connect to the database, just as if auto_reconnect was set? This is useful for people who start Kamailio from their init script and can't be sure if their database host is up before their Kamailio host boots 1. https://github.com/sipcapture/homer-api/blob/master/examples/sipcapture/sip…

3 2

Integration with Broadsoft and Config Question
by Daryn Johnson 16 Sep '16

16 Sep '16

I’m looking to briefly check the sanity of my configuration. I have been working w/Kamailio the last few months, and have read the SIP router ebook. Thank you for your efforts and the documentation. It has been a GREAT HELP! We are attempting to replace an Acme Packet SBC, and our setup is as follows: |ACCESSUSERS| ⇔ |KAMAILIO/SEMS| ⇔ |BROADSOFT| ⇔ |GENBAND| ⇔ |PSTN| Kamailio & SEMS are (currently) on the same box, and communicate using localhost via different ports. They also listen on the same private (core) IPs, using different ports. Since Broadsoft doesn’t support the PATH header, I manipulate the contact header w/the SEMS privateIP & port in the contact header. This causes it to be stored in the Broadsoft Registration string, and calls from PSTN route directly to SEMs to start calls etc… Registraton code looks like this: route[REGISTRATION] { remove_hf("Route"); add_path_received(); $avp(contact) = $ct; xlog("L_INFO", "found contact header $avp(contact)"); setflag(FLAG_REGISTRATION); if (nat_uac_test("20")){ xlog("L_INFO", "[REGISTRATION_ATTEMPT] [FROM_NAT] R=$ru ID=$ci UA='$ua'\n"); force_rport(); remove_hf("Contact"); append_hf("Contact: <sip:$fU" + "@" + "!SEMSPRIVATEIP!:5060;" + "alias=$si~$sp~1>\n"); setflag(FLT_NATS); } else { xlog("L_INFO", "[REGISTRATION_ATTEMPT] [PUBLIC_IP] \r\n[REGISTRATION_INFO] R=$ru UA='$ua'\r\n"); remove_hf("Contact"); append_hf("Contact: <sip:$fU@" + "!SEMSPRIVATEIP!:5060;" + "alias=$si~$sp~1>\n"); } $du = "sip:R_IP:REGISTRAR_PORT"; t_on_reply("REG"); route(RELAY); exit; } Then in the relay route, I force the registration out of the private interface & port of Kamailio if (isflagset(FLAG_REGISTRATION)){ $fs = "LISTEN_UDP_PRIVATE"; } Finally, if a registration is successful from BROADSOFT, I store it in the location table, to allow me to screen calls from access to ensure they are ‘registered’ onreply_route[REG]{ if(status=="200") { xlog("L_INFO", "REGISTRATION BE GOOD FOR $fu"); save("location", "0x04"); } } Calls from ACCESS, I check USRLOC for screening, then pass the call to SEMS via the localhost:port. From there, sems enacts the B2BUA/SBC module and routes the call to BROADSOFT, and along to our GENBAND Gateway… I could never get calls to work properly without SEMS, just using RTP PROXY/ENGINE, possibly because the RTP for the GENBAND & KAMAILIO/SEMS are on different private subnets. Our setup is fairly legacy ‘heavy’ and many things can’t be changed. Calls from PSTN route to GENBAND, then BROADSOFT, then SEMS, then KAMAILIO. Since I have manipulated the contact header, BROADSOFT maintains that in the INVITE and I can use the ‘handle_ruri_alias’ to route the call properly. SEMS sets the PUBLIC IP of the KAMAILIO/SEMS box in the SDP, and facilitates rtp relaying. if (isflagset(FLAG_TOUSER)){ xlogl("L_INFO", "TOUSERFLAGSET"); if(is_method("INVITE|BYE")){ handle_ruri_alias(); } /*if(has_body("application/sdp") && nat_uac_test("8")) { fix_nated_sdp("10"); xlog("L_INFO", "SDP CONTAINS NAT"); }*/ $fs = "LISTEN_UDP_PUBLIC"; } I’ve gone headlong into this setup, and it works! But, I wanted to get a ‘Sanity Check’ to determine if there is a better way to do this. We are at the point where I’m doing load testing and attempting to validate this for a ‘production’ ready state. If possible, I’d just like to ask the community if they have encountered a similar setup, and how they solved it? Thanks in advance!!! Daryn Johnson Senior VoIP Engineer 248.485.1109 www.telnetww.com 1175 W. Long Lake Rd. | Suite 101 | Troy, MI 48098

1 0

Kamailio Authentication failing
by Eric Koome 16 Sep '16

16 Sep '16

Hi all - my Kamailio - 4.1.6 is receiving this particular structured INVITES from multiple IPs, and for some reason it is not requesting authentication. I have AUTH & IPAUTH modules in use for two years now, but this is bypassing that and actually forwarding the invite to asterisk servers behind Kamailio. I notice from the invite that the contact (c=IN IP4 10.10.10.10) is unusual and in the private range. Is this what is bypassing Authentication? Any Pointers on how to stop this. This is flooding my boxes! ========================================== 2016/09/16 06:45:16.824768 83.48.28.27:63029 -> MYPUBLICIP:5060 INVITE sip:9441134018000@MYPUBLICIPSIP/2.0 Via: SIP/2.0/UDP 83.48.28.27:5060;branch=z9hG4bK898408493 Max-Forwards: 70 From: "442030930112" <sip:442030930112@83.48.28.27>;tag=5044916148 To: <sip:9441134018000@MYPUBLICIP> Contact: <sip:442030930112@83.48.28.27> Call-ID: 9xCjfQkrmDwT7D6AqM0BTYAgmWjG84nq CSeq: 102 INVITE User-Agent: SIP Call Date: Fri, 16 Sep 2016 05:45:48 +0000 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 308 v=0 o=442030930112 8982 1536 IN IP4 10.10.10.10 s=SIP c=IN IP4 10.10.10.10 t=0 0 m=audio 14806 RTP/AVP 0 3 8 4 18 101 a=rtpmap:0 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMU/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv Regards Eric

2 1

http_async_query - How to Allow SIP transaction to continue
by Jonathan Hunter 16 Sep '16

16 Sep '16

Hi Guys, I am just testing with jansson and http_async_query. Essentially I sent an INVITE into kamailio ,it hits CALL_START_REQUEST (see below) and the query allows the http site I send to, to provide routing details for kamailio. Logging the $rm that hits CALL_START_ REQUEST, it shows as INVITE, however when CALL_START_REQUEST_REPLY is hit, the message type is OPTIONS, is this expected behaviour? Essentially I want it to continue in the script with the same $ci,$rm so I can onwards route it for termination, but all I get are options messages being sent, any hints please? As Ideally I want the transaction to continue to a route where I can forward on modified request for termination. Thanks Jon Routes; # Send to FreeSWITCHroute[CALL_START_REQUEST]{xlog("L_INFO","route[CALL_START_REQUEST] Call received $ci,$rm,$ru,$du\n"); ###Build JSON request to send to MAP$var(params) = $null;jansson_set("string", "username", "xxx", "$var(params)");jansson_set("string", "password", "xxx", "$var(params)");jansson_set("string", "source", "$fU", "$var(params)");jansson_set("string", "route", "$var(carriername)", "$var(params)");jansson_set("string", "destination", "$rU", "$var(params)");/// other fields set herehttp_set_timeout(5000);http_set_method("POST");http_append_header("Content-Type: application/json");http_async_query("http://test.com", "$var(params)", "CALL_START_REQUEST_REPLY");xlog("L_INFO", "Checking here we have status=$var(status)\n");xlog("L_INFO", "Current Destination is $du\n");#return;#exit;}#########################route[CALL_START_REQUEST_REPLY]{xlog("L_INFO", "route[CALL_START_REQUEST_REPLY]: $ci,$rm,$ru,$du\n"); Logs show; route[CALL_START_REQUEST] Call received b087fa53-f5fc-1234-94ac-f68eb8da94f7,INVITE,sip:441624891676@8.8.8.8:5060;user=phone,<null> route[CALL_START_REQUEST_REPLY]: 123,OPTIONS,sip:G711A-282951200@kamailio.org,sip:G711A-282951200@8.8.8.8:5080

2 3

help with kamailio rpm made from source
by Dmitry 16 Sep '16

16 Sep '16

Hello I obtained the source from https://www.kamailio.org/pub/kamailio/4.3.4/src/kamailio-4.3.4_src.tar.gz I found a SPEC file there and I prepared a RPM with this spec file. When I start a Kamailio (from this rpm) - it starts but the LOG file gives only 1 line: INFO: <core> [tcp_main.c:4657]: init_tcp(): using epoll_lt as the io watch method (auto detected) I want to test this rpm and if successfull i willl add patches and rebuild the rpm. How to get correct RPM?

3 17

Branch variables in branch_failure_route
by Paul Smith 14 Sep '16

14 Sep '16

Hi, I am struggling to figure out variable scope in branches and branch_failure_routes. Is there a way to store a variable in a branch route so that I can then read it from the branch failure route if that branch fails? Also does writing $ru in branch_failure_route append a branch? For example I have multiple SIP devices registered to Kamailio on the same account credentials. The devices are behind same NAT, so same public IP. Some devices have SRTP required and some cannot use SRTP at all, Kamailio does not know in advance of the call whether the call will use SRTP or RTP. When I send a call to that user Kamailio correctly parallel forks to all the registered devices. I use rtpengine offer to send SDP with RTP/SAVP in the initial invites. If one of the devices sends a 488 due to not being able to handle the RTP/SAVP I then want to run rtpengine_offer with RTP/AVP and send another invite to the same device. In the branch failure route I seem to have to read the current $ru and then write it back into $ru before calling t_relay() or I get Sep 14 14:56:06 registrar-secure /usr/sbin/kamailio[10168]: ERROR: tm [t_fwd.c:1771]: t_forward_nonack(): ERROR: t_forward_nonack: no branches for forwarding This code seems to work, but it feels like I’m doing something very wrong…. in particular $avp(triedrtpout) is set for the transaction, I should be setting it separately for each branch. event_route[tm:branch-failure:SRTP] { # Handle failure response xlog("L_INFO", "Handling $T_reply_code response to $rm to ru: <$ru> and du: $du\n"); if(t_check_status("488") && is_method("INVITE")) { if ($avp(triedsrtpout)==1) { # set $ru seems to create a new branch ... even when we set it to its current value. # we want to generate a new SIP INVITE to the same AOR but with different SDP $avp(buffer)=$ru; $ru=$avp(buffer); xlog("L_INFO","488 caught. Resending to $ru"); # try with unencrypted RTP/AVP rtpengine_delete(); rtpengine_offer("to-tag trust-address replace-origin replace-session-connection ICE=force RTP/AVP"); if (!t_relay()) { sl_reply_error(); } } } } I am using kamailio 4.2 debian package for this registrar. Sorry for the rambling question … I don’t understand enough to write the short version! Paul

3 3

Kamailio v4.4.3 Released
by Daniel-Constantin Mierla 14 Sep '16

14 Sep '16

Hello, Kamailio SIP Server v4.4.3 stable release is out. This is a maintenance release of the latest stable branch, 4.4, that includes fixes since the release of v4.4.2. There is no change to database schema or configuration language structure that you have to do on previous installations of v4.4.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.4.3. For more details about version 4.4.3 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2016/09/kamailio-v4-4-3-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

jansson_get path to access a JSON reponse containing an array
by Jonathan Hunter 14 Sep '16

14 Sep '16

Hi guys, Sorry for the question but Its not clear to me using json_path and the jansson module how to get the contents of a response containing an array. Please see the returned response below that I am getting on kamailio; {"status":"success","destination"[{"msisdn":"4412345566","route":"Carrier1"}],"maxduration":"60","source":"447777777","message_id":"c41c1823aafb80f44620f4cdc6802c0d"} I can access any of the fields status,destination,maxduration,source,message_id fine, however you will notice with destination it has an array (as multiple destinations with other queries could return more than one choice). So currently when I do jansson_get_field("$http_rb", "destination", "$var(mapdestination)"); I get; [{"msisdn":"447107110194","route":"carrier1"}] How do I get into the contents of the array in the first place so I can return the msisdn value itself, and the route? This is not a problem if its not an array, however if the result is contained within [] how do I indicate the path? I have tried destination[0], destination.msisdn with no luck, and Im sure I am missing something so any advice would be great. Thanks Jon

2 2

Planning the release of Kamailio v4.4.3
by Daniel-Constantin Mierla 14 Sep '16

14 Sep '16

Hello, now that the summer in the northern hemisphere is gone and many of us returned at the work, I am considering to release a new version of Kamailio out of the latest stable branch 4.4. The plan is to have it out next week, either on Wednesday Sep 14 or Thursday, the following day. If the are issues not reported on the bug tracker from github project, it is better to open one item. Do it also if you reported something on sr-users or sr-dev in the past two months and it is not actual discussion about it, likely it was forgotten or not noticed during the holidays and digesting the archive is going to take a while. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com - http://www.kamailio.org http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 1

Kamailio as Proxy with public and private interface
by gmele 13 Sep '16

13 Sep '16

Hello, we are building a new SIP service composed by a 2 SIP Proxies and 2 SIP Registrars. There is only one proxy active (controlled by keepalived). To secure our SIP infra, we have decided to publish a TLS public interface on the proxy side. The SIP registrar will not be reachable from the outside, so we have added a provate interface on the SIP proxy and another one on the SIP registrar (192.168.1.x). In brief: SIP Proxy, 2 I/F, 213.xxx.xxx.xxx (TLS public) and 192.168.1.x (UDP private) SIP Registrar, 1 I/F, 192.168.1.x (UDP private). Is this architecture possible with kamailio? We tried to implement this, but it seems that all requests incoming from the public IP address are correctly routed to the SIP registrar, but with the public IP instead of the private one, resulting in the impossibility to send the SIP requests to the registrar. Should we add a VIA toute in the SIP message before forwarding it to the SIP registrar? How can this be done? Thx for your help Giovanni -- View this message in context: http://sip-router.1086192.n5.nabble.com/Kamailio-as-Proxy-with-public-and-p… Sent from the Users mailing list archive at Nabble.com.

1 1

Issues on getting Notify Body data and parse it
by José Seabra 13 Sep '16

13 Sep '16

Hello there, I'm facing some issues trying to get the attributes from notify Body that contains information that i need for my calls . First i save the $rb into $var then i try to replace the characters #012 that seems to be the separator of all attributes in the body, i have tried it using s.select, s.strip, s.replace but without any success in all of them. The body msg is something like this: Calling-Name-Status: available#012Calling-Name: "teste XX" < sip:XXXXXXXXX@domain.com>#012Presentation-Indicator: allowed Anyone have any idea how can I get the value of each attribute: e.g: Calling-Name-Status: available Calling-Name: "teste XX" <sip:XXXXXXXXX@domain.com> Presentation-Indicator: allowed Best Regards -- José

2 3

Dump running config out of Kamailio into a file
by Jesus Tovar 13 Sep '16

13 Sep '16

Hi, I currently have a situation with my Kamailio servers sitting in production, the folder containing all configuration files at /etc/kamailio was deleted from the server and there is not backup. Is there a possibility to show or dump the current running configuration into a file so I could restore the configuration file from it. If it happens these servers get restarted then we are going to be in real trouble. Thanks for your help Kind regards, Albert T. Jesus Tovar | Senior Network Engineer | The Summit Group jtovar(a)thesummitgroup.com.au thesummitgroup.com.au 1300 049 749 Level 1, 39 Railway Road, Blackburn VIC 3130 The Summit Group

1 0

Radius Authentication for kamailio-4.4.2
by Chan Kwang Mien 12 Sep '16

12 Sep '16

Hi, I have installed Kamailio-4.4.2 and radiusclient-ng-0.5.6 for Radius Authentication to a Free Radius Server. I have used the steps in http://www.kamailio.org/docs/openser-radius-1.0.x.html to enable Radius Authentication for Kamailio-4.4.2. However, when I restart kamailio and my SIP phone register to Kamailio, Kamailio did not Radius Authenticate to Free Radius Server. Can anyone advise me on other document for Radius Authentication for Kamailio ? Thanks. Regards, Kwang Mien

2 1

pua_dialoginfo: Tags only in "early" state PUBLISH
by Stefan Kohlhauser 12 Sep '16

12 Sep '16

Hello everyone! I've been playing around with the pua_dialoginfo. I noticed that the From- and To-Tags are only included in the PUBLISH's dialog-info of the "early" state. This is also true if include_tags of pua_dialoginfo is explicitly set to '1'. The reason for this behaviour is probably that in pua_dialoginfo.c in the function __dialog_sendpublish() the call to dialog_publish_multi() only happens with the tag parameters set for the "early" state. For all other dialog states the tag parameters are statically set to '0'. Even though the local- and from-tags are optional I tend to regard this as a bug. I would have expected for the tags to be included in the dialog-info for all states when include_tags is set. Or is there a reason for this behaviour? Best regards Stefan

2 1

Kamailio RFC/3GPP standards
by Owais Ahmad 12 Sep '16

12 Sep '16

Hello, I can only see the 3GPP specification mentioned in ims_qos module. Please have the other IMS modules also updated with the respective 3gpp versions. Thanks, Owais

2 1

Re: [SR-Users] Kamailio - Multiple NAT
by Gholamreza Sabery 12 Sep '16

12 Sep '16

The best way to see what is happening is to capture some packets on the Kamailio system on which calls happen and see if any IP address or port number issues exist in body of SIP messages or not. You can simple use Wireshark to see what is going on. Regards On Mon, Sep 12, 2016 at 4:58 PM, Marino Mileti <marino.mileti(a)alice.it> wrote: > Hi guys, > > > > i'm going in trouble with multinat scenario and Kamailio J > > > > I've this scenario... > > > > Two kamailio boxes with the same configuration (KAM_A and KAM_B). Each box > has a WIFI in AP_MODE with the same address (but different SSID obviously) > 192.168.254.1. Finally there are 2 clients...CLIENT_1 connected to wifi > with KAM_A, and CLIENT_2 connected using wifi with KAM_B. Each box has an > instance of RTPEngine configured. > > KAM_A & KAM_B are also connected using wired connection with other clients > in order to allow that a calls to KAM_B (or KAM_A) generate a forking > scenario and all clients connected starts to RINGS (wired & wireless client) > > > > The mobile client connected on KAM_A makes a call to KAM_B... the forking > scenario starts and all clients on KAM_B starts to ring. > > If any wired client pick up the call everything works fine. If a wireless > client pick up the call...no media L > > > > Is it possible? Should I've to instruct RTPENGINE in any particular mode? > > > > > > *Marino Maria Mileti* > > *marino.mileti(a)alice.it <marino.mileti(a)alice.it>* > > > > *[image: cid:006a01cb6b0e$67eecdae$_CDOSYS2.0]**Reduce your energy > consumption and keep polar bears on ice!* > > > > > ------------------------------ > [image: Avast logo] > <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> > > Questa e-mail è stata controllata per individuare virus con Avast > antivirus. > www.avast.com > <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> > > > _______________________________________________ > 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 > >

1 0

Kamailio - Multiple NAT
by Marino Mileti 12 Sep '16

12 Sep '16

Hi guys, i'm going in trouble with multinat scenario and Kamailio J I've this scenario... Two kamailio boxes with the same configuration (KAM_A and KAM_B). Each box has a WIFI in AP_MODE with the same address (but different SSID obviously) 192.168.254.1. Finally there are 2 clients...CLIENT_1 connected to wifi with KAM_A, and CLIENT_2 connected using wifi with KAM_B. Each box has an instance of RTPEngine configured. KAM_A & KAM_B are also connected using wired connection with other clients in order to allow that a calls to KAM_B (or KAM_A) generate a forking scenario and all clients connected starts to RINGS (wired & wireless client) The mobile client connected on KAM_A makes a call to KAM_B... the forking scenario starts and all clients on KAM_B starts to ring. If any wired client pick up the call everything works fine. If a wireless client pick up the call...no media L Is it possible? Should I've to instruct RTPENGINE in any particular mode? Marino Maria Mileti <mailto:marino.mileti@alice.it> marino.mileti(a)alice.it cid:006a01cb6b0e$67eecdae$_CDOSYS2.0Reduce your energy consumption and keep polar bears on ice! --- Questa e-mail è stata controllata per individuare virus con Avast antivirus. https://www.avast.com/antivirus

1 0

Advice for routing and sbc functions
by Oliver Roth 12 Sep '16

12 Sep '16

Hi all I am using kamailio for several years now and I am really happy with it! Now an new scenario to solve: Sip-carrier A --> Kamailio loadbalancer --> multiple kamailio routing gateways --> freeswitch/sbc/media-gw/whatever --> sip - carriers B,C & D Kamailio gateways: They are doing different checks, manipulate the numbers (national /international format) and they use the carrierroute module freeswitch/sbc/media-gw/whatever: how can I use these services with the routing information from the carrierroute module? These media-gws will do the connection and maybe transcoding to the carriers. Can you please help me with the interaction within the systems "routing gateways" and the "media/sbc gateways"? Do I have to add all the carriers in the carrierroute as "routes" with the ip of the media gw/freeswitch and then splitting it up on the freeswitch again? Thanks for helping Oli

1 0

DSCP options
by Marcelo Bonin 09 Sep '16

09 Sep '16

I need to set DSCP values in Kamailio, I couldn't found a way to set in the configuration. Any help will be appreciated. Thanks.

2 1

4.4 topos ignoring Record-Route?
by Daniel Tryba 09 Sep '16

09 Sep '16

Looking into the topos module of 4.4.2 I see something unexpected, ACK and BYE are routed directly to the Contact in the 200 OK on an INVITE where the 200 OK has a Record-Route. Without the topos module the message flow is as expected: 109.235.32.57<->185.61.68.106<->109.235.32.55 With topos the ACK and BYE take a different route: 109.235.32.57<->185.61.68.106<->109.235.32.55 <->109.235.32.48 109.235.32.5x are proxies. 185.61.68.106 is the registrar To my understanding, in the callflow below the ACK should be send via the proxy that inserted Record-Route: <sip:109.235.32.55;lr> in the 200 OK. Am I wrong to conclude this or the topos module ignoring the routing? Also note the To/From are passed untouched in the ACK/BYE and the BYE to the called endpoint hasn't had its Via headers stripped. Call scenario: tryba/+31407110xxx calls 0880100XXX. Both are registered to 185.61.68.106 via proxies (resp. 109.235.32.57 and 109.235.32.55) >From 109.235.32.57 -> 185.61.68.106: INVITE sip:0880100XXX@sip.itco.nl SIP/2.0 Record-Route: <sip:109.235.32.57;r2=on;lr> Record-Route: <sip:109.235.32.57;transport=tcp;r2=on;lr> Via: SIP/2.0/UDP 109.235.32.57;branch=z9hG4bK7dd7.e15a4ffb2c3a7f882117b80c85f75223.0;i=60c Via: SIP/2.0/TCP 10.0.3.169:5067;rport=5067;received=109.235.34.226;branch=z9hG4bKb5004b60de287221 Route: <sip:109.235.32.57;transport=tcp;lr> From: <sip:tryba@sip.itco.nl>;tag=576aff337d33fbe5 To: <sip:0880100XXX@sip.itco.nl> Contact: <sip:tryba@10.0.3.169:5067;transport=tcp;alias=109.235.34.226~5067~2> >From 185.61.68.106 -> 109.235.32.55: INVITE sip:+31880100XXX@109.235.32.48 SIP/2.0 Via: SIP/2.0/UDP 185.61.68.106;branch=z9hG4bK7dd7.d4cdfe8014e0d5d806b4dc4a7422c8e0.1 Route: <sip:loadbalancer@109.235.32.55;lr;received=sip:109.235.32.48:5060> From: <sip:+31407110XXX@sip.itco.nl>;tag=576aff337d33fbe5 To: <sip:+31880100XXX@sip.itco.nl> Contact: <sip:btpsh-57b6cc75-f42d-1@185.61.68.106> >From 109.235.32.55 -> 185.61.68.106: SIP/2.0 200 OK Via: SIP/2.0/UDP 185.61.68.106;rport=5060;branch=z9hG4bK7dd7.d4cdfe8014e0d5d806b4dc4a7422c8e0.1 Record-Route: <sip:109.235.32.55;lr> From: <sip:+31407110XXX@sip.itco.nl>;tag=576aff337d33fbe5 To: <sip:+31880100XXX@sip.itco.nl>;tag=as61fc6a47 Contact: <sip:%2b31880100XXX@109.235.32.48> >From 185.61.68.106 -> 109.235.32.57 SIP/2.0 200 OK From: <sip:tryba@sip.itco.nl>;tag=576aff337d33fbe5 To: <sip:0880100XXX@sip.itco.nl>;tag=as61fc6a47 Via: SIP/2.0/UDP 109.235.32.57;branch=z9hG4bK7dd7.e15a4ffb2c3a7f882117b80c85f75223.0;i=60c,SIP/2.0/TCP 10.0.3.169:5067;rport=5067;received=109.235.34.226;branch=z9hG4bKb5004b60de287221 Contact: <sip:atpsh-57b6cc75-f42d-2@185.61.68.106> Record-Route: <sip:109.235.32.57;r2=on;lr>,<sip:109.235.32.57;transport=tcp;r2=on;lr> >From 109.235.32.57 -> 185.61.68.106: ACK sip:atpsh-57b6cc75-f42d-2@185.61.68.106 SIP/2.0 Via: SIP/2.0/UDP 109.235.32.57;branch=z9hG4bK7dd7.a49ec0c055166a1935ff84cfca4d1a5c.0;i=60c Via: SIP/2.0/TCP 10.0.3.169:5067;rport=5067;received=109.235.34.226;branch=z9hG4bK24aff2df2ee7b43f From: <sip:tryba@sip.itco.nl>;tag=576aff337d33fbe5 To: <sip:0880100XXX@sip.itco.nl>;tag=as61fc6a47 Contact: <sip:tryba@10.0.3.169:5067;transport=tcp;alias=109.235.34.226~5067~2> >From 185.61.68.106 -> 109.235.32.48 ACK sip:%2b31880100XXX@109.235.32.48 SIP/2.0 Via: SIP/2.0/UDP 185.61.68.106;branch=z9hG4bK7dd7.ac1b9a45042f9c26277afb9ec32bebe0.0 From: <sip:tryba@sip.itco.nl>;tag=576aff337d33fbe5 To: <sip:0880100XXX@sip.itco.nl>;tag=as61fc6a47 Contact: <sip:btpsh-57b6cc75-f42d-1@185.61.68.106> >From 109.235.32.57 -> 185.61.68.106 BYE sip:atpsh-57b6cc75-f42d-2@185.61.68.106 SIP/2.0 Via: SIP/2.0/UDP 109.235.32.57;branch=z9hG4bK4dd7.24da86af0a970ee399c9504933d8d751.0;i=60c Via: SIP/2.0/TCP 10.0.3.169:5067;rport=5067;received=109.235.34.226;branch=z9hG4bK0d8db1380c5011b6 From: <sip:tryba@sip.itco.nl>;tag=576aff337d33fbe5 To: <sip:0880100XXX@sip.itco.nl>;tag=as61fc6a47 >From 185.61.68.106 -> 109.235.32.48 BYE sip:%2b31880100XXX@109.235.32.48 SIP/2.0 Via: SIP/2.0/UDP 185.61.68.106;branch=z9hG4bK4dd7.4916bf7bcaa3b0346e43c3785a9be782.0 Via: SIP/2.0/UDP 109.235.32.57;branch=z9hG4bK4dd7.24da86af0a970ee399c9504933d8d751.0;i=60c Via: SIP/2.0/TCP 10.0.3.169:5067;rport=5067;received=109.235.34.226;branch=z9hG4bK0d8db1380c5011b6 From: <sip:tryba@sip.itco.nl>;tag=576aff337d33fbe5 To: <sip:0880100XXX@sip.itco.nl>;tag=as61fc6a47

2 19

Video calls with delay on one side
by Yuriy Gorlichenko 09 Sep '16

09 Sep '16

Hi. I usein kamailio 4.4 +rtpengine 4.5 for making videocalls though Web And have issue with it: I have one way audio and video till video not started. For example i calling form A point to B point B point accept call and have Audio and Video flow from the point A but point A have no any media flow. For now i using scheme when kamailio runs together with asterisk but also i had this issue with calls without asterisk. There is a link on issue i discribed before https://github.com/sipwise/rtpengine/issues/260 I use settings at the kamailio rtpengine_manage("force trust-address replace-origin replace-session-connection RTP/SAVPF") for making UDP call to WS And rtpengine_manage("trust-address replace-origin replace-session-connection ICE=remove DTLS=passive RTP/AVP media-address=MY_IP_ADDR"); For converting to back I use folloving scheme (Call from A to B) WSphone(A)->(1-st leg)-> kamailio+rtpeinge->(2-nd leg->)asterisk asterisk ->(3-st leg)-> kamailio+rtpeinge->(4-th leg)->WSphone(B) At the example below rtpengine (with kamailio) an asteirsk uses same server But connection made thoufh external interface. not local (it is made because in a production scheme asteirsk will be in a different server, but actuaaly it is does not have any matter, just for understanding topology) So at the logs after call i see next things First call from point A to asterisk (first leg and second leg) [1473403379.000108] INFO: [ga5c8qb2uaasr42j63h4]: ------ Media #1 <https://github.com/sipwise/rtpengine/issues/1> (audio over RTP/SAVPF) using opus/48000/2 [1473403379.000108] INFO: [ga5c8qb2uaasr42j63h4]: --------- Port my.ser.ver.ip:30170 <> A.po.int.ip:51918, 244 p, 22944 b, 0 e, 1473403349 last_packet [1473403379.000108] INFO: [ga5c8qb2uaasr42j63h4]: --------- Port my.ser.ver.ip:30171 <> A.po.int.ip:51920 (RTCP), 4 p, 196 b, 0 e, 1473403349 last_packet [1473403379.000108] INFO: [ga5c8qb2uaasr42j63h4]: ------ Media #2 <https://github.com/sipwise/rtpengine/issues/2> (video over RTP/SAVPF) using VP8/90000 [1473403379.000108] INFO: [ga5c8qb2uaasr42j63h4]: --------- Port my.ser.ver.ip:30200 <> A.po.int.ip:51922, 209 p, 164499 b, 0 e, 1473403349 last_packet [1473403379.000108] INFO: [ga5c8qb2uaasr42j63h4]: --------- Port my.ser.ver.ip:30201 <> A.po.int.ip:51924 (RTCP), 129 p, 4292 b, 0 e, 1473403349 last_packet [1473403379.000108] INFO: [ga5c8qb2uaasr42j63h4]: --- Tag 'as76205e6a', created 0:39 ago for branch '', in dialogue with '5aalmrbaek' [1473403379.000108] INFO: [ga5c8qb2uaasr42j63h4]: ------ Media #1 <https://github.com/sipwise/rtpengine/issues/1> (audio over RTP/AVP) using opus/48000/2 [1473403379.000108] INFO: [ga5c8qb2uaasr42j63h4]: --------- Port my.ser.ver.ip:30156 <> my.ser.ver.ip:29236, 242 p, 23861 b, 0 e, 1473403349 last_packet [1473403379.000108] INFO: [ga5c8qb2uaasr42j63h4]: --------- Port my.ser.ver.ip:30157 <> my.ser.ver.ip:29237 (RTCP), 0 p, 0 b, 0 e, 1473403340 last_packet [1473403379.000108] INFO: [ga5c8qb2uaasr42j63h4]: ------ Media #2 <https://github.com/sipwise/rtpengine/issues/2> (video over RTP/AVP) using VP8/90000 [1473403379.000108] INFO: [ga5c8qb2uaasr42j63h4]: --------- Port my.ser.ver.ip:30184 <> my.ser.ver.ip:27252, 148 p, 152972 b, 0 e, 1473403349 last_packet [1473403379.000108] INFO: [ga5c8qb2uaasr42j63h4]: --------- Port my.ser.ver.ip:30185 <> my.ser.ver.ip:27253 (RTCP), 0 p, 0 b, 0 e, 1473403340 last_packet call from asterisk to point B (3-d leg and 4-th leg) [1473403379.000108] INFO: [7742bfcf0e9570ab6d7d40a52cd4cd4a@my.ser.ver.ip:5999]: ------ Media #1 <https://github.com/sipwise/rtpengine/issues/1>(audio over RTP/AVP) using opus/48000/2 [1473403379.000108] INFO: [7742bfcf0e9570ab6d7d40a52cd4cd4a@my.ser.ver.ip:5999]: --------- Port my.ser.ver.ip:30238 <> my.ser.ver.ip:29958, 239 p, 24902 b, 0 e, 1473403349 last_packet [1473403379.000108] INFO: [7742bfcf0e9570ab6d7d40a52cd4cd4a@my.ser.ver.ip:5999]: --------- Port my.ser.ver.ip:30239 <> my.ser.ver.ip:29959 (RTCP), 0 p, 0 b, 0 e, 1473403340 last_packet [1473403379.000108] INFO: [7742bfcf0e9570ab6d7d40a52cd4cd4a@my.ser.ver.ip:5999]: ------ Media #2 <https://github.com/sipwise/rtpengine/issues/2>(video over RTP/AVP) using VP8/90000 [1473403379.000108] INFO: [7742bfcf0e9570ab6d7d40a52cd4cd4a@my.ser.ver.ip:5999]: --------- Port my.ser.ver.ip:30264 <> my.ser.ver.ip:40374, 205 p, 163663 b, 0 e, 1473403349 last_packet [1473403379.000108] INFO: [7742bfcf0e9570ab6d7d40a52cd4cd4a@my.ser.ver.ip:5999]: --------- Port my.ser.ver.ip:30265 <> my.ser.ver.ip:40375 (RTCP), 0 p, 0 b, 0 e, 1473403340 last_packet [1473403379.000108] INFO: [7742bfcf0e9570ab6d7d40a52cd4cd4a@my.ser.ver.ip:5999]: --- Tag 'fgl9f33gf0', created 0:39 ago for branch '', in dialogue with 'as5f26dbd9' [1473403379.000108] INFO: [7742bfcf0e9570ab6d7d40a52cd4cd4a@my.ser.ver.ip:5999]: ------ Media #1 <https://github.com/sipwise/rtpengine/issues/1>(audio over RTP/SAVPF) using opus/48000/2 [1473403379.000108] INFO: [7742bfcf0e9570ab6d7d40a52cd4cd4a@my.ser.ver.ip:5999]: --------- Port my.ser.ver.ip:30218 <> B.po.int.ip:26138, 242 p, 21481 b, 0 e, 1473403349 last_packet [1473403379.000108] INFO: [7742bfcf0e9570ab6d7d40a52cd4cd4a@my.ser.ver.ip:5999]: --------- Port my.ser.ver.ip:30219 <> B.po.int.ip:30161 (RTCP), 4 p, 188 b, 0 e, 1473403349 last_packet [1473403379.000108] INFO: [7742bfcf0e9570ab6d7d40a52cd4cd4a@my.ser.ver.ip:5999]: ------ Media #2 <https://github.com/sipwise/rtpengine/issues/2>(video over RTP/SAVPF) using VP8/90000 [1473403379.000108] INFO: [7742bfcf0e9570ab6d7d40a52cd4cd4a@my.ser.ver.ip:5999]: --------- Port my.ser.ver.ip:30246 <> B.po.int.ip:31174, 198 p, 163302 b, 0 e, 1473403349 last_packet [1473403379.000108] INFO: [7742bfcf0e9570ab6d7d40a52cd4cd4a@my.ser.ver.ip:5999]: --------- Port my.ser.ver.ip:30247 <> B.po.int.ip:59057 (RTCP), 12 p, 540 b, 0 e, 1473403349 last_packet As i see there is no any lost pakets and errors and all packets going through but at the asteirsk log i see that Sent RTP P2P packet to my.ser.ver.ip:30238 (type 107, len 000102) Sent RTP P2P packet to my.ser.ver.ip:30156 (type 111, len 000071) Sent RTP P2P packet to my.ser.ver.ip:30238 (type 107, len 000079) when packets sended from asterisk to rtpengine they have no response So it looks like that asterisk have no response from rtpengine when sending him any IP. Also i checked at the webphone side but it sending pakents. So at this side no trouble... Thanks for advices!

1 0

app_lua with lua5.3
by Péter Barabás 08 Sep '16

08 Sep '16

Hi, I would use lua5.3 with app_lua module. Is it possible since it is not compiled with lua5.1? Thanks. Peter

2 2

Call forking with delay
by Igor Potjevlesch 08 Sep '16

08 Sep '16

Hello list, I have a proxy that manage fix calls and a proxy that manage mobile calls. I want to implement call forking so when I call to a fix phone the call goes to the fix proxy and it forks to the mobile proxy who manage the call and viceversa. I want the fix to ring 2 or 3 times before the mobile. In order to do so in my fix proxy I added a diversion field with the reason "forking" and in my mobile proxy I check if the reason is "forking" to wait with the fonction "async_route" from the ASYNC module. When I call the fix I can fork to the mobile with delay without any problem because in that case my fix proxy manage one call and my mobile proxy the other one. But when I call the mobile the call doesn't even get to my fix proxy because my mobile proxy has to make the original call to wait while he is forking the fix call to my other proxy. And i think that calling the async_route fonction makes both calls to wait. My code is: Fix proxy: if($tU==123456789){ add_diversion("forking"); append_branch("sip:987654321@proxy_mobile"); } Mobile proxy: if($tU==987654321){ $var(z) = "1"; append_branch("sip:123456789@proxy_fix"); } route[INVITE]{ if($dir == "forking"){ async_route("RELAY", "7"); }else if($var(z) == "1"){ async_route("RELAY", "7"); }else{ route(RELAY); exit; } } How can I differentiate in the second scenario when I call the mobile phone to fork to the fix phone? Thank you for your help. Igor.

2 3

Kamailio send TCP packets from random ports
by Yuriy Gorlichenko 08 Sep '16

08 Sep '16

Hi. I try to make working kamailio on TCP infront of asterisk Before to send to asteisk any packet i added $fs=ip.add.re.ss:port Also discribed listen=tcp:ip.add.re.ss:port But kamailio send outgoing packets from random prot throug TCP Presume i configured 5060 port but it send from 35410 port. googling on a core functions didn gives me any answer. May be i forget something to set at the config?

3 12

Re: [SR-Users] Kamailio send TCP packets from random ports
by Yuriy Gorlichenko 08 Sep '16

08 Sep '16

I know about dispatcher but it not always canbe heplfull. Sometimes i need my own logic that not implemented at the dispatcher. For checking kamailio live from asterisk im use qualfy for keepalives, if i use asterisk only as media server but it not always usefull. Sometimes scenarios need to use another logic. 2016-09-08 18:24 GMT+03:00 Yuriy Gorlichenko <ovoshlook(a)gmail.com>: > I know about dispatcher but it not always canbe heplfull. Sometimes i need > my own logic that not implemented at the dispatcher. > > For checking kamailio live from asterisk im use qualfy for keepalives, if > i use asterisk only as media server but it not always usefull. Sometimes > scenarios need to use another logic. > > 2016-09-08 17:50 GMT+03:00 Federico Cabiddu <federico.cabiddu(a)gmail.com>: > >> The issue with dispatcher is that, in case of TCP transport, you cannot >> set the sending socket for the same reason. >> Basically, each time a client TCP socket is open by Kamailio, the SO >> select the port, due to the lack of support for SO_REUSEPORT. >> >> Cheers, >> >> Federico >> >> On Thu, Sep 8, 2016 at 4:33 PM, Daniel Tryba <d.tryba(a)pocos.nl> wrote: >> >>> On Thu, Sep 08, 2016 at 03:38:32PM +0300, Yuriy Gorlichenko wrote: >>> > I didnt thought about keepalive. I suppose it can help. >>> >>> Better than using qualify in asterisk is to use the dispatcher module in >>> kamailio. The idea is the same, but more configurable and it is just 1 >>> keepalive mechanisme so the amount of keepalives doesn't scale linear >>> with the number of registered 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 >>> >> >> >

1 0

Kamailio 4.4.2 S-CSCF DB error
by Owais 08 Sep '16

08 Sep '16

I can see queries by S-CSCF like: The subscriber table doesn't contain presentity_uri as per: http://kamailio.org/docs/db-tables/kamailio-db-4.4.x.html#idp58608 <http://kamailio.org/docs/db-tables/kamailio-db-4.4.x.html#idp58608> Please share a valid mysql schema for tables used by s-cscf and presence. Thanks, Owais -- View this message in context: http://sip-router.1086192.n5.nabble.com/Kamailio-4-4-2-S-CSCF-DB-error-tp15… Sent from the Users mailing list archive at Nabble.com.

1 0

UAC reg_contact_addr with rinstance
by DanB 08 Sep '16

08 Sep '16

Hey Guys, Have hit an issue when using uacreg. When working with Broadsoft (I know, life is not perfect), it appears that their SBC sitting in front of registration server is caching REGISTRATION requests and it will not be able to re-REGISTER in case of credential changes for example. This will break the registration process in registrar since that one flushes the registrations on password change. Their "standard" workaround it is using "rinstance" uri parameter in contact (each contact should be unique in order to be cached by their SBC). Checking it appears that I can only add rinstance statically in reg_contact_addr configuration parameter. Is there any chance to provide things like $avp or selects which I can later modify via mi maybe? Thanks in advance for any kind of tip! DanB

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Query - REST API calls in async
by Agalya RAMACHANDRAN 08 Sep '16

08 Sep '16

Hi team, I am Agalya and I work for Comcast. We are trying to use kamailio as proxy server for our project. I have a query regarding the same. We are trying to use REST API calls asynchronously from the config file. I see that there is a module "http_async_client" which is used to handle REST services asynchronously. Found that libcurl is used to perform these operations. My question is: 1) How much time kamailio is taking to send the HTTP request out after TCP port is opened? Is there any time dependency we need to wait on libcurl to send HTTP request out? B'coz in opensips I see a delay time and libcurl is used in-turn. I found kamailio doing similar to opensips and hence am posting this question. 2) We can also use PUT service by using http_set_method(PUT)? Should we add code to support PUT service? Thanks in advance. Regards, Agalya

2 1

Mitigating DDOS attacks from carrier based on ani or dnis by limitating cps or blocking when detecting
by anfecora 08 Sep '16

08 Sep '16

Hi is there any way to use pike and htable to mitigate ddos or flood attack from trusted trunks. I need help to build it the same way kamailio control registrations. Case a carrier trunk star calling several users from the system to more than 50 CPS(calls per second), it will affect the system but cannot block the trunk since it is pstn traffic coming from a sip provider, therefore we need to find a way to identify this traffic based on ANI or DNIS or any other header and then blocked for a time just like pike does with registrations, then start the cycle all over. in less words make kamailio be aware of invite request rate, then verify that is from the same source means ani or dnis then tagged as bad traffic them star dropping it for a specified time, while normal traffic still flowing unaffected. any recommendations will be highly appreciated. thank you

2 3

New links for "IMS getting started"
by Franz Edler 07 Sep '16

07 Sep '16

Hi, Due to some storage limits I recently encountered I had to temporarily remove the image from my drop-box. Now this issue is fixed and I have uploaded the image again. Here are the links: The VMware image: https://www.dropbox.com/s/swhlgzpflgl3p3e/Debian_8.2-step5-basicKamailio%26b asicIMScore.zip?dl=0 The config-files: https://www.dropbox.com/s/nm51v4orxrox4ac/Kamailio-IMS%20config-files.zip?dl =0 Good luck. Franz DI Franz Edler University of Applied Sciences VIENNA, Austria

4 3

deb.kamailio.org down?
by Carsten Bock 06 Sep '16

06 Sep '16

Hi, currently, deb.kamailio.org is down. Is anyone taking care of this? Thanks, Carsten -- Carsten Bock CEO (Geschäftsführer) ng-voice GmbH Millerntorplatz 1 20359 Hamburg / Germany http://www.ng-voice.com mailto:carsten@ng-voice.com Office +49 40 5247593-40 Fax +49 40 5247593-99 Sitz der Gesellschaft: Hamburg Registergericht: Amtsgericht Hamburg, HRB 120189 Geschäftsführer: Carsten Bock Ust-ID: DE279344284 Hier finden Sie unsere handelsrechtlichen Pflichtangaben: http://www.ng-voice.com/imprint/

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