- sr-users - kamailio.org

Kamailio start up issue
by NITESH BANSAL 01 Apr '16

01 Apr '16

Hello, I installed Kamailio 4.3 on debian wheezy machine using debian package from https://www.kamailio.org/wiki/packages/debsThe installation went fine, but I'm unable to start kamailio. It seems that Kamailio is having issue locating the issues.Kamailio installed all the modules in location '/usr/lib/x86_64-linux-gnu/kamailio/modules', when building from source, it usedto copy modules in '/usr/lib/kamailio/modules'.Any suggestions on how can I fix it? Thanks,Nitesh

1 1

Framework/Lib to create SIP packets
by Igor Olhovskiy 01 Apr '16

01 Apr '16

Hi, all Not a Kamailio-related question, but. I want to form SIP-packets with info I need and receive answers. At first iteration in would be PUBLISH (with XML info) messages (PUBLISH - 200 OK) sent to Kamailio. I want to use Python (more preferable) or Lua (less preferable). Is there any lib I can create messages with, or forget and just use plain text + UDP socket? -- Best regards, Igor

5 13

HOW send 4XX to the caller when the callee response to bye is 481 Call leg/transaction does not exist
by Humberto H 01 Apr '16

01 Apr '16

Hello; [image: Inline image 1] HOW can I send 4XX to the caller when the callee response to a BYE message is: *481 Call leg/transaction does not exist. BUT the caller is waiting.* Before this message. I drop the bye sent by the callee, my purpose here is "*Do not send the BYE TO THE CALLER"* JUST send a message in order to the caller do failover, If I send *BYE* to the caller it can't do failover Thanks in advance. 10.929596| | BYE | | |SIP Request BYE CSeq:104 | | |(5060) ------------------> (5060) | | |10.929981| | 481 Call leg/transac | |SIP Status 481 Call leg/transaction does not exist | | |(5060) <------------------ (5060) | | |10.929983| | 481 Call leg/transac | |SIP Status 481 Call leg/transaction does not exist Call leg/transaction does not exist | | |(5060) ------------------> (5060) | | |10.930445| | 481 Call leg/transac | |SIP Status 481 Call leg/transaction does not exist | | |(5060) ------------------> (5060) | | |10.930451| | 481 Call leg/transac | |SIP Status 481 Call leg/transaction does not exist

2 1

How can I send 404 to the caller instead of BYE?
by Humberto H 01 Apr '16

01 Apr '16

Hello I am trying to send 404 to the caller instead of BYE. Using the configuration showned bellow. when the callee send BYE Kamailio sends -- *404* to the callee (to the source of the BYE signaling) --- and *BYE* to the caller But I need to send 404 response to the caller only Thanks in advance event_route[dialog:end] { if(is_method("BYE") && ((src_ip==XXXXXXX) )){ send_reply("404", "Not found"); xlog("L_INFO","ENVIO 404 ON BYE \n"); } exit; }

2 1

stateful messages in a dialog
by Al S 01 Apr '16

01 Apr '16

Hi, I would like to know how I can make subsequent messages in a dialog (ACK and BYE) to be stateful.Your help will be greatly appreciated. // reroute INVITE after timeout.failure_route[MANAGE_FAILURE]{ if (t_branch_timeout()) { # change to new destination IP $rd = "10.10.10.10"; t_relay(); exit; }...} Thanks,AS

2 1

Record-Route header
by jignesh gandhi 01 Apr '16

01 Apr '16

Hello All, I have a need to save the Record-Route header from the INVITE and use it as a route header in requests like BYE. PROXY --> Kamailio(as PROXY) --> Asterisk I am using rr module to add a record_route when passing the INVITE to Asterisk. Thanks, Jignesh Gandhi

2 4

Re: [SR-Users] Proxy configuration
by Jignesh Gandhi 31 Mar '16

31 Mar '16

Thanks Alex for you reply. I understand the dispatcher module can be used to load balance inbound requests. But my questions is more related with outbound requests thru Kamailio. Are you saying that in-dialog request can be routed automatically by Kamailio to the correct Asterisk instance on the back end ? And this is included in the stock config ? I was just looking for an example. Thanks, --Jignesh On 3/30/16, 6:00 AM, "sr-users on behalf of sr-users-request(a)lists.sip-router.org" <sr-users-bounces(a)lists.sip-router.org on behalf of 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. Destination Timeout (ACK message) (Al S) > 2. Proxy configuration (Jignesh Gandhi) > 3. Re: Proxy configuration (Alex Balashov) > 4. Local Kamailio share a hosted Kamailio/Asterisk sip server > (S?bastien Brice) > 5. SQlite DB Lock with Presence (Kristian Kielhofner) > 6. Kamailio local and hosted (S?bastien Brice) > 7. Re: Releasing Kamailio v4.4.0 (Daniel-Constantin Mierla) > 8. Re: SQlite DB Lock with Presence (Daniel-Constantin Mierla) > > >---------------------------------------------------------------------- > >Message: 1 >Date: Tue, 29 Mar 2016 09:27:56 -0400 >From: Al S <alirsm(a)outlook.com> >To: "sr-users(a)lists.sip-router.org" <sr-users(a)lists.sip-router.org> >Subject: [SR-Users] Destination Timeout (ACK message) >Message-ID: <BAY177-W1513432E7A3AFDAAB1DE8DD4870(a)phx.gbl> >Content-Type: text/plain; charset="iso-8859-1" > >Hi, >The destination node is down. By adding the following code Kamailo re-routing INVITE to a new destination after getting 408 timeout (as below). > >failure_route[MANAGE_FAILURE]{ if (t_branch_timeout()) { # change to new destination IP $rd = "10.10.10.10"; t_relay(); exit; }...} >However, the ACK message coming from source is still being send to the old destination. How can I reroute other messages in the dialog to be sent to this new destination.. >Thank, >AS >-------------- next part -------------- >An HTML attachment was scrubbed... >URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20160329/b91e2df…> > >------------------------------ > >Message: 2 >Date: Tue, 29 Mar 2016 14:24:01 +0000 >From: Jignesh Gandhi <Jignesh.Gandhi(a)MoviusCorp.com> >To: "sr-users(a)lists.sip-router.org" <sr-users(a)lists.sip-router.org> >Subject: [SR-Users] Proxy configuration >Message-ID: <011222F3-5820-41FB-A0D8-8D727EA121FA(a)moviuscorp.com> >Content-Type: text/plain; charset="utf-8" > >Hello, > >Is it possible to use kamailio to be a stateful outbound proxy ? > >The use case is kamailio is an outbound proxy for multiple instances of Asterisk. >Thus all responses need to go back to the same instance of Asterisk that generated them. > >After the call is established. >How can I ensure that the Re-Invites, BYEs, etc are sent to the same instance of Asterisk that generated the outbound Invite ? > >Any help is appreciated. > >Thanks, >--Jignesh >-------------- next part -------------- >An HTML attachment was scrubbed... >URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20160329/5081718…> > >------------------------------ > >Message: 3 >Date: Tue, 29 Mar 2016 10:29:19 -0400 >From: Alex Balashov <abalashov(a)evaristesys.com> >To: sr-users(a)lists.sip-router.org >Subject: Re: [SR-Users] Proxy configuration >Message-ID: <56FA913F.6080903(a)evaristesys.com> >Content-Type: text/plain; charset=windows-1252; format=flowed > >Jignesh, > >Everything you're asking for is part of Kamailio's core functionality: > >1. Most useful applications of Kamailio involve stateful relay, and the >stock included configurations are stateful. > >2. The loose-routing section of the configuration, and, more broadly, >RFC 3261 in general, take care of routing in-dialog requests (ACKs, >reinvites, BYEs, etc.) to the appropriate dialog party. > >3. Load-balancing/distribution among Asterisk instances can be done with >the 'dispatcher' module; there are many examples floating around, as >it's one of the most common applications for Kamailio. > >-- Alex > >-- >Alex Balashov | Principal | Evariste Systems LLC >1447 Peachtree Street NE, Suite 700 >Atlanta, GA 30309 >United States > >Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) >Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ > > > >------------------------------ > >Message: 4 >Date: Tue, 29 Mar 2016 16:32:36 +0200 (CEST) >From: S?bastien Brice <sebastien.brice(a)jvs.fr> >To: sr-users <sr-users(a)lists.sip-router.org> >Subject: [SR-Users] Local Kamailio share a hosted Kamailio/Asterisk > sip server >Message-ID: <2011796741.12246629.1459261956126.JavaMail.zimbra(a)jvs.fr> >Content-Type: text/plain; charset=utf-8 > > >Hi everyone >At my workplace there's plenty of hardphone directly connected to an hosted sbc in the cloud. >As an added value i proposed to add locally Kamailio as a proxy server to 'federate' all the sip traffic. > >ipphone behind nat (A) <---> sip proxy local 192.168.x.x (B)<----> sbc registrar (mydomain.com) and sip proxy with public IP (C) > >when hardphone try to register, B rewrite (with rewritehost) the ip public of (C). B then uses exported function of TM to t_relay_to(C). >to route INVITE?from A to C i use proxy_authenticate("mydomain.com") function (on B) from auth_db to pass the calls inbetween. > >This works well, but i am wondering if that makes sense if i get several phones behind B. >Should stateless forward works better in this kind of scenario? > >In a near future i plan to offer disaster recovery, and i want all users binding to C (users(a)mydomain.com)instead bind on B (in case link B---C get down)and assume some kind of telephone functionnality >Is that possible ? >Do you have some sort of example Kamailio do it good On-premise/localy ? >How to deal with nat since all hardphones have rfc1918?address? > >thx you > >PS: as an sbc (C) i did setup kamailio/asterisk in a real-time configuration. > > > >S?bastien BRICE VoIP, Support et Int?gration > > > >------------------------------ > >Message: 5 >Date: Tue, 29 Mar 2016 20:50:14 -0400 >From: Kristian Kielhofner <kris(a)kriskinc.com> >To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> >Subject: [SR-Users] SQlite DB Lock with Presence >Message-ID: > <CAKDfjge4kLhdLjV5V7gsOOsLsdByeLgtwnqE-EEXHhxbVeV8Ag(a)mail.gmail.com> >Content-Type: text/plain; charset=UTF-8 > >Hello everyone, > > I'm trying to use BLF with the config here (except with SQlite): > >http://kb.asipto.com/kamailio:presence:k43-blf > > BLF seems to work perfectly except after the call is ended I get >"sqlite commit failed: database is locked" error messages as seen >here: > >http://pastebin.com/xVijj98H > >BLF is then broken at this point. > >kamailio -v >version: kamailio 4.3.5 (x86_64/linux) 950657 >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_PTHREAD_MUTEX, 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: 950657 >compiled on 01:33:01 Mar 29 2016 with x86_64-openwrt-linux-gnu-gcc 4.8.3 > >Has anyone ever seen this before? > >Thanks! > >-- >Kristian Kielhofner > > > >------------------------------ > >Message: 6 >Date: Tue, 29 Mar 2016 14:38:40 +0200 (CEST) >From: S?bastien Brice <sebastien.brice(a)jvs.fr> >To: sr-users <sr-users(a)lists.sip-router.org> >Subject: [SR-Users] Kamailio local and hosted >Message-ID: <1194659979.12170808.1459255120868.JavaMail.zimbra(a)jvs.fr> >Content-Type: text/plain; charset=utf-8 > >Hi everyone >At my workplace there's plenty of hardphone directly connected to an hosted sbc in the cloud. >As an added value i proposed to add locally Kamailio as a proxy server to 'federate' all the sip traffic. > >ipphone behind nat (A) <---> sip proxy local 192.168.x.x (B)<----> sbc registrar (mydomain.com) and sip proxy with public IP (C) > >when hardphone try to register, B rewrite (with rewritehost) the ip public of (C). B then uses exported function of TM to t_relay_to(C). >to route INVITE?from A to C i use proxy_authenticate("mydomain.com") function (on B) from auth_db to pass the calls inbetween. > >This works well, but i am wondering if that makes sense if i get several phones behind B. >Should stateless forward works better in this kind of scenario? > >In a near future i plan to offer disaster recovery, and i want all users binding to C (users(a)mydomain.com)instead bind on B (in case link B---C get down)and assume some kind of telephone functionnality >Is that possible ? >Do you have some sort of example Kamailio do it good On-premise/localy ? >How to deal with nat since all hardphones have rfc1918?address? >, >thx you > >PS: as an sbc (C) i did setup kamailio/asterisk in a real-time configuration. > > > >S?bastien BRICE VoIP, Support et Int?gration > > > >------------------------------ > >Message: 7 >Date: Wed, 30 Mar 2016 10:02:02 +0200 >From: Daniel-Constantin Mierla <miconda(a)gmail.com> >To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.sip-router.org>, sr-dev <sr-dev(a)lists.sip-router.org>, > "business(a)lists.kamailio.org" <business(a)lists.kamailio.org> >Subject: Re: [SR-Users] Releasing Kamailio v4.4.0 >Message-ID: <56FB87FA.8080304(a)gmail.com> >Content-Type: text/plain; charset=utf-8 > >Hello, > >the final packaging for 4.4.0 is expected to start after 14:00UTC today. >Backports to branch 4.4 can be done up to 12:00UTC, after that let's >sync on sr-dev mailing list, because I will start testing how it builds >on different platforms and it is good to have the a consistent source code. > >Cheers, >Daniel > >On 29/03/16 09:04, Daniel-Constantin Mierla wrote: >> Hello, >> >> refreshing about the major release 4.4.0 planned for tomorrow -- if any >> dev has patches to backport to branch 4.4 should do it as soon as possible. >> >> Cheers, >> Daniel >> >> On 18/03/16 12:16, Daniel-Constantin Mierla wrote: >>> Hello, >>> >>> with no major issues reported lately, I think we are on a good track for >>> releasing stable v4.4.0. Next week a lot of people prepare for Easter, >>> therefore I propose to do the release on Wednesday, March 30, 2016. If >>> new things pop up, the exact date can be adjusted a bit before or after >>> the current proposal. >>> >>> Any new discovered issue should be reported to bug trucker: >>> >>> * https://github.com/kamailio/kamailio/issues >>> >>> Help to complete the upgrade guide would be very appreciated: >>> >>> * https://www.kamailio.org/wiki/install/upgrade/stable-to-devel >>> >>> A draft list of what is new in 4.4 is already available at: >>> >>> * https://www.kamailio.org/wiki/features/new-in-4.4.x >>> >>> Cheers, >>> Daniel >>> > >-- >Daniel-Constantin Mierla >http://www.asipto.com >http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com > > > > >------------------------------ > >Message: 8 >Date: Wed, 30 Mar 2016 11:37:39 +0200 >From: Daniel-Constantin Mierla <miconda(a)gmail.com> >To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.sip-router.org> >Subject: Re: [SR-Users] SQlite DB Lock with Presence >Message-ID: <56FB9E63.7040104(a)gmail.com> >Content-Type: text/plain; charset=windows-1252 > >Hello, > >not using sqlite here, but could be related to presence modules doing db >transactions and not releasing the locks. Do you get the issue after the >first call, or it runs for a while and at some point it breaks? > >Cheers, >Daniel > >On 30/03/16 02:50, Kristian Kielhofner wrote: >> Hello everyone, >> >> I'm trying to use BLF with the config here (except with SQlite): >> >> http://kb.asipto.com/kamailio:presence:k43-blf >> >> BLF seems to work perfectly except after the call is ended I get >> "sqlite commit failed: database is locked" error messages as seen >> here: >> >> http://pastebin.com/xVijj98H >> >> BLF is then broken at this point. >> >> kamailio -v >> version: kamailio 4.3.5 (x86_64/linux) 950657 >> 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_PTHREAD_MUTEX, 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: 950657 >> compiled on 01:33:01 Mar 29 2016 with x86_64-openwrt-linux-gnu-gcc 4.8.3 >> >> Has anyone ever seen this before? >> >> Thanks! >> > >-- >Daniel-Constantin Mierla >http://www.asipto.com >http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com > > > > >------------------------------ > >Subject: Digest Footer > >_______________________________________________ >sr-users mailing list >sr-users(a)lists.sip-router.org >http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > >------------------------------ > >End of sr-users Digest, Vol 130, Issue 29 >*****************************************

2 1

timeout function
by Al S 31 Mar '16

31 Mar '16

Hi, I am routing call traffic thru proxy to destination A. If destination A is down, we need to reroute to destination B. What function we can use to process timeout so we can reroute the calls to different node? Thanks,ARM

2 1

TCP connection select problem
by 张顺通 31 Mar '16

31 Mar '16

my program structure is LinPhone------TCP------KA_EDGE------UDP------KA_REGISTER------UDP------FS KA_EDGE:all clients TCP connection is in this server KS_REGSITER:all register info is process here when User A want to use program,Linphone send SIP REGISTER to KA_EDGE by TCP(and TCP is always present), and KA_EDGE send to KA_REGISTER by UDP. when FS want to Call A, FS generate SIP: INVITE A@KA_REGISTER_IP KA_REGISTER recive INVITE, lookup the register list, send to KA_EDGE INVITE TCP_IP:TCP_PORT Route:KA_EDGE_IP:KA_EDGE_PORT and LinPhone recive that INVITE. *my question is:* LinPhone A and LinPhone B both register in KA_REGISTER. FS send INVITE: INVITE B: KA_REGISTER_IP and KA_REGISTER lookup correct，send to KA_EDGE INVITE TCP_IP_B:TCP_PORT_B Route:KA_EDGE_IP:KA_EDGE_PORT but KA_EDGE is not correct.sip is correct,* but KA_EDGE send the INVITE to A' TCP connnection* INVITE TCP_IP_B:TCP_PORT_B Route:KA_EDGE_IP:KA_EDGE_PORT Someone had this problem?Is this a bug?

4 19

db_cassandra status and plans
by Andreas Granig 31 Mar '16

31 Mar '16

Hi all, I was checking db_cassandra with Kamailio 4.3 recently to try making it work as a usrloc back-end, and my observation is that the module has quite some issues. In general, the module states that it's only tested with cassandra 1.0.6, which is really out-dated. I tried with the latest version of the 1.0 series (1.0.12) and even with lots of fiddling I didn't manage to make it work, could be my fault though. There has not been any work for 4 years other than cosmetical documentation fixes two years ago. Documentation is missing important information how to set up the version table, that could be fixed though. The schema definition in the documentation is missing new columns which have been introduced, that could be fixed as well. The module uses the legacy thrift interface, whereas nowadays (since cassandra 1.2+) a native cql driver is available as open-source. The module can not be used together with nathelper for nat pings, since the queries don't provide a key, which is required for cassandra. Our plan is to revamp the whole module and make it work with recent cassandra versions and remove the nat ping limitations. Main question is whether to go for a new module or replacing the old one, since it doesn't seem to work anymore anyways. Feedback whether you got it working with Kamailio 4.3+ and your views on future plans is much appreciated! Andreas

3 5

acc_cdrs empty
by Nelson Migliaro 31 Mar '16

31 Mar '16

Hello, I am not able to set up cdr. I see acc table is updated with 2 new entries after each call (INVITE and BYE) but acc_cdrs is empty. There are errors in the log. I think that I set up everything but is not working, - load the module, set module params and set the flag? Am I missing something, should I use a store procedure or something? ------------------------------------------------------------------------------------------------------------ loadmodule "acc.so" modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) modparam("acc", "detect_direction", 1) modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) 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") if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } route[WITHINDLG] { if (has_totag()) { if (loose_route()) { route(DLGURI); if (is_method("BYE")) { CREATE TABLE `acc_cdrs` ( `id` int(10) unsigned NOT NULL AUTO_INCREMENT, `start_time` varchar(32) NOT NULL DEFAULT '', `end_time` varchar(32) NOT NULL DEFAULT '', `duration` varchar(32) NOT NULL DEFAULT '', PRIMARY KEY (`id`), KEY `start_time_idx` (`start_time`) ) ENGINE=MyISAM AUTO_INCREMENT=2 DEFAULT CHARSET=latin1$$ Thank you.

1 1

UAC Register Asterisk
by Alberto Sagredo 31 Mar '16

31 Mar '16

Hi Im trying to modify example for Asterisk/Kamailio integration that miconda did, and trying to handle Forwarded register to Asterisk in an authenticated way instead removing secret in asterisk. I have made this changes route[REGFWD] { if(!is_method("REGISTER")) { return; } $var(rip) = $var(astip); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $var(astport); $uac_req(furi)="sip:" + $(au{s.substr,8,0}) + "@" + $var(rip); $uac_req(turi)="sip:" + $(au{s.substr,8,0}) + "@" + $var(rip); $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n"; if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n"; xlog("Dentro de REGFWD"); t_on_failure("REG_AST"); uac_req_send(); } failure_route[REG_AST] { if (t_is_canceled()) { exit; } xlog("Dentro de REG AST"); if(t_check_status("401|407")) { xlog("Dentro de 401"); $avp(auser) = "test"; $avp(apass) = "test"; uac_auth(); t_relay(); exit; } } What i see on traces: U x.x.x..240:5060 -> x.x.x..56:5060 REGISTER sip:x.x.x.56:5060 SIP/2.0. Via: SIP/2.0/UDP x.x.x.240;branch=z9hG4bKfb46.a47535d4000000000000000000000000.0. To: <sip:200@x.x.x.56>. From: <sip:200@x.x.x.56>;tag=533cb9e91f4b999cf76861cbb9ed54ed-6a4f. CSeq: 10 REGISTER. Call-ID: 09dc120d65932ed7-16044(a)127.0.0.1. Max-Forwards: 70. Content-Length: 0. User-Agent: ANY. Contact: <sip:emp_001_200@x.x.x.240:5060>. Expires: 3600. . # U x.x.x.56:5060 -> x.x.x..240:5060 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP x.x.x..240 ;branch=z9hG4bKfb46.a47535d4000000000000000000000000.0;received=x.x.x..240 ;rport=5060. From: <sip:200@x.x.x.56>;tag=533cb9e91f4b999cf76861cbb9ed54ed-6a4f. To: <sip:200@x.x.x.56>;tag=as62b47138. Call-ID: 09dc120d65932ed7-16044(a)127.0.0.1. CSeq: 10 REGISTER. Server: FPBX-2.11.0(11.17.1). Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="59e39692". Content-Length: 0. . But Kamailio is not sending authentication . Any idea what could be wrong? Or any tip to do what i try. The idea is that a sip endpoint register with authentication on kamailio and it sends register with authentication to asterisk. Thanks in advance

4 7

Kamailio v4.4.0 Released - new major version is out
by Daniel-Constantin Mierla 30 Mar '16

30 Mar '16

Kamailio v4.4.0 is out – it comes with 9 new modules and a consistent set of improvements touching more than 60 existing modules. You can read detailed release notes at: * https://www.kamailio.org/w/kamailio-v4-4-0-release-notes/ Many thanks to all developers and community members that made possible this release. Enjoy Kamailio v4.4.0! The development will start to focus on 5.0.0 - join the discussion on sr-users mailing list to design and decide what should Kamailio 5.0.0 look like - some notes are already collected at: * https://www.kamailio.org/wiki/devel/kamailio-5.0-design Thank you for flying Kamailio! Daniel -- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com

1 0

Releasing Kamailio v4.4.0
by Daniel-Constantin Mierla 30 Mar '16

30 Mar '16

Hello, with no major issues reported lately, I think we are on a good track for releasing stable v4.4.0. Next week a lot of people prepare for Easter, therefore I propose to do the release on Wednesday, March 30, 2016. If new things pop up, the exact date can be adjusted a bit before or after the current proposal. Any new discovered issue should be reported to bug trucker: * https://github.com/kamailio/kamailio/issues Help to complete the upgrade guide would be very appreciated: * https://www.kamailio.org/wiki/install/upgrade/stable-to-devel A draft list of what is new in 4.4 is already available at: * https://www.kamailio.org/wiki/features/new-in-4.4.x Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com

1 2

Kamailio local and hosted
by Sébastien Brice 29 Mar '16

29 Mar '16

Hi everyone At my workplace there's plenty of hardphone directly connected to an hosted sbc in the cloud. As an added value i proposed to add locally Kamailio as a proxy server to 'federate' all the sip traffic. ipphone behind nat (A) <---> sip proxy local 192.168.x.x (B)<----> sbc registrar (mydomain.com) and sip proxy with public IP (C) when hardphone try to register, B rewrite (with rewritehost) the ip public of (C). B then uses exported function of TM to t_relay_to(C). to route INVITE from A to C i use proxy_authenticate("mydomain.com") function (on B) from auth_db to pass the calls inbetween. This works well, but i am wondering if that makes sense if i get several phones behind B. Should stateless forward works better in this kind of scenario? In a near future i plan to offer disaster recovery, and i want all users binding to C (users(a)mydomain.com)instead bind on B (in case link B---C get down)and assume some kind of telephone functionnality Is that possible ? Do you have some sort of example Kamailio do it good On-premise/localy ? How to deal with nat since all hardphones have rfc1918 address? , thx you PS: as an sbc (C) i did setup kamailio/asterisk in a real-time configuration. Sébastien BRICE VoIP, Support et Intégration

1 1

Proxy configuration
by Jignesh Gandhi 29 Mar '16

29 Mar '16

Hello, Is it possible to use kamailio to be a stateful outbound proxy ? The use case is kamailio is an outbound proxy for multiple instances of Asterisk. Thus all responses need to go back to the same instance of Asterisk that generated them. After the call is established. How can I ensure that the Re-Invites, BYEs, etc are sent to the same instance of Asterisk that generated the outbound Invite ? Any help is appreciated. Thanks, --Jignesh

2 1

Destination Timeout (ACK message)
by Al S 29 Mar '16

29 Mar '16

Hi, The destination node is down. By adding the following code Kamailo re-routing INVITE to a new destination after getting 408 timeout (as below). failure_route[MANAGE_FAILURE]{ if (t_branch_timeout()) { # change to new destination IP $rd = "10.10.10.10"; t_relay(); exit; }...} However, the ACK message coming from source is still being send to the old destination. How can I reroute other messages in the dialog to be sent to this new destination.. Thank, AS

1 0

SIP / WebRTC / GSoC proposals
by Daniel Pocock 29 Mar '16

29 Mar '16

Hi all, We had a few GSoC applications[1] from students who would like to do work relating to SIP and WebRTC this summer. The students have a range of skills. Some are interested in the server side (C, C++) code. Some are interested in Android and Java projects (e.g. the Lumicall and CSipSimple dialers) and some are web developers, interested in making practical use of WebRTC. Would anybody else from the community be interested in helping to mentor one of these projects? Prospective mentors can review the proposals privately before making any firm commitment, please contact me for details. Mentors can have some input in the direction of the students' work, just as long as all code is released publicly. Regards, Daniel 1. http://danielpocock.com/gsoc-2016-voice-video-chat

2 2

XML-RPC reply doesn't contain any data
by Arsen 29 Mar '16

29 Mar '16

Hi guys! We have an issue with XML-RPC module. >From this doc: http://kamailio.org/docs/modules/4.3.x/modules/xmlrpc.html "Success replies always contain at least one return value. In our case the simplest success replies contain single boolean with value 1" We are trying to reload domain module after adding/removing new domain, so we send the following request: XX - web server; YY - kamailio T XX.XX.XX.XX:55028 -> YY.YY.YY.YY:8080 [AP] POST /RPC2 HTTP/1.1. Host: YY.YY.YY.YY:8080. Accept: */*. Accept-Encoding: deflate, gzip. User-Agent: cURL. Accept-Charset: UTF-8. Content-Length: 120. Content-Type: application/x-www-form-urlencoded. . <?xml version="1.0" encoding="iso-8859-1"?> <methodCall> <methodName>domain.reload</methodName> <params/> </methodCall> T YY.YY.YY.YY:8080 -> XX.XX.XX.XX:55028 [AP] HTTP/1.1 200 OK. Sia: SIP/2.0/TCP XX.XX.XX.XX:55028. Server: kamailio (4.3.4 (x86_64/linux)). Content-Length: 108. . <?xml version="1.0"?> <methodResponse> <params> <param> <value></value> </param> </params> </methodResponse> As you can see in the response from the kamailio there is no any data, so our script is waiting until timeout... Please advice. Thanks, -- Regards, Arsen.

4 6

Timeout Handling (update)
by Al S 28 Mar '16

28 Mar '16

Hi, I am trying to re-route to a different $ru after timeout and right before Kamailio reply with a 408.I made the changes right before sl_reply_error(), however it does not work as I expected. Any help will be appreciated. Also, $rm represents the request message received by Kamailio. Is there a variable represents the messages leaving Kamailio?. route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|SUBSCRIBE")) { t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { // re-route to a new $ru before sending 408 sl_reply_error(); } exit;} Thanks,AS

2 2

using DMQ , Kamailio shutsdown
by Sunil More 28 Mar '16

28 Mar '16

Hello All, I am running two instances of kamailio and using dmq. Kamailio starts good. However one of the node shutsdown unexpectedly. Unsure what is going wrong. Please let me know what could be wrong. below is a the conf detail. loadmodule "dmq.so" modparam("dmq", "server_address", "sip:sunilmore.in:5060") modparam("dmq", "notification_address", "sip:sunilmore.in:5064") loadmodule "usrloc.so" modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", DBURL2) modparam("usrloc", "use_domain", 1) modparam("usrloc", "matching_mode", 0) modparam("usrloc", "desc_time_order", 1) modparam("usrloc", "nat_bflag", FLB_NATB) modparam("usrloc", "db_insert_null", 1) loadmodule "dmq_usrloc.so" modparam("dmq_usrloc", "enable", 0) Below is the backtrace of the logs. ar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14039]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank 11: kex Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14039]: DEBUG: kex [kex_mod.c:161]: child_init(): rank is (11) Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14039]: DEBUG: <core> [sruid.c:106]: sruid_init(): root for sruid is [srid-56f4d496-36d7-] (0 / 19) Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14039]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank 11: mi_rpc Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14039]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank 11: ndb_redis Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14013]: DEBUG: <core> [main.c:1723]: main_loop(): Expect maximum 2177 open fds Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14040]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank 12: htable Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14040]: DEBUG: htable [htable.c:237]: child_init(): rank is (12) Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14040]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank 12: websocket Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14040]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank 12: mi_datagram Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: tm Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: tm [callid.c:134]: child_init_callid(): DEBUG: callid: ' 0e4b331663a1f7ae-14041(a)sunilmore.in' Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: tmx Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: tmx [tmx_mod.c:272]: child_init(): rank is (-4) Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: sl Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: tls Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: ndb_mongodb Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: dmq Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: usrloc Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sruid.c:106]: sruid_init(): root for sruid is [ulcx-56f4d496-36d9-] (0 / 19) Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14040]: DEBUG: <core> [local_timer.c:61]: init_local_timer(): timer_list between 0xa762c8 and 0xaba2c8 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: dmq_usrloc Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14040]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa76100, 40, 1, (nil)), fd_no=0 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14039]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank 11: htable Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14039]: DEBUG: htable [htable.c:237]: child_init(): rank is (11) Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14039]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank 11: websocket Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14039]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank 11: mi_datagram Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: dmq_usrloc [dmq_usrloc.c:110]: child_init(): child_init PROC_INIT Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: registrar Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sruid.c:106]: sruid_init(): root for sruid is [uloc-56f4d496-36d9-] (0 / 19) Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: nathelper Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: nat_traversal Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: mi_fifo Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: avpops Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: rtimer Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: dialog Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: domain Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: uac Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: presence Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: kex Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: kex [kex_mod.c:161]: child_init(): rank is (-4) Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sruid.c:106]: sruid_init(): root for sruid is [srid-56f4d496-36d9-] (0 / 19) Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: mi_rpc Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: ndb_redis Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: htable Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: htable [htable.c:237]: child_init(): rank is (-4) Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: websocket Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [sr_module.c:914]: init_mod_child(): rank -4: mi_datagram Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14039]: DEBUG: <core> [local_timer.c:61]: init_local_timer(): timer_list between 0xa762c8 and 0xaba2c8 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14039]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa76100, 38, 1, (nil)), fd_no=0 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [local_timer.c:61]: init_local_timer(): timer_list between 0xa32028 and 0xa76028 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 8, 1, 0x7fab28a78438), fd_no=0 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 9, 1, 0x7fab28a786a8), fd_no=1 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 10, 4, 0x7fab082bb21c), fd_no=2 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 11, 4, 0x7fab082bb2a8), fd_no=3 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 12, 4, 0x7fab082bb334), fd_no=4 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 13, 4, 0x7fab082bb3c0), fd_no=5 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 14, 4, 0x7fab082bb44c), fd_no=6 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 15, 4, 0x7fab082bb4d8), fd_no=7 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 16, 4, 0x7fab082bb564), fd_no=8 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 17, 4, 0x7fab082bb5f0), fd_no=9 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 18, 4, 0x7fab082bb67c), fd_no=10 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 19, 4, 0x7fab082bb708), fd_no=11 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 20, 4, 0x7fab082bb794), fd_no=12 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 21, 4, 0x7fab082bb938), fd_no=13 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 22, 4, 0x7fab082bb9c4), fd_no=14 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 23, 4, 0x7fab082bba50), fd_no=15 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 24, 4, 0x7fab082bbadc), fd_no=16 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 27, 4, 0x7fab082bbb68), fd_no=17 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 25, 4, 0x7fab082bbbf4), fd_no=18 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 26, 4, 0x7fab082bbc80), fd_no=19 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 29, 4, 0x7fab082bbd0c), fd_no=20 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 31, 4, 0x7fab082bbd98), fd_no=21 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 28, 3, 0x7fab28d647f0), fd_no=22 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 30, 3, 0x7fab28d64810), fd_no=23 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 36, 3, 0x7fab28d64830), fd_no=24 Mar 25 06:03:02 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0xa31e60, 39, 3, 0x7fab28d64850), fd_no=25 Mar 25 06:03:03 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:458]: wsconn_get_list(): wsconn_get_list Mar 25 06:03:03 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:502]: wsconn_get_list(): wsconn_get_list returns list [(nil)] with [0] members Mar 25 06:03:04 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:458]: wsconn_get_list(): wsconn_get_list Mar 25 06:03:04 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:502]: wsconn_get_list(): wsconn_get_list returns list [(nil)] with [0] members Mar 25 06:03:05 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:458]: wsconn_get_list(): wsconn_get_list Mar 25 06:03:05 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:502]: wsconn_get_list(): wsconn_get_list returns list [(nil)] with [0] members Mar 25 06:03:06 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:458]: wsconn_get_list(): wsconn_get_list Mar 25 06:03:06 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:502]: wsconn_get_list(): wsconn_get_list returns list [(nil)] with [0] members Mar 25 06:03:07 P172 /usr/local/kamailio/sbin/kamailio[14027]: DEBUG: tm [h_table.c:127]: free_cell_helper(): freeing transaction 0x7fab0853a9d8 from timer.c:648 Mar 25 06:03:07 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:458]: wsconn_get_list(): wsconn_get_list Mar 25 06:03:07 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:502]: wsconn_get_list(): wsconn_get_list returns list [(nil)] with [0] members Mar 25 06:03:08 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:458]: wsconn_get_list(): wsconn_get_list Mar 25 06:03:08 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:502]: wsconn_get_list(): wsconn_get_list returns list [(nil)] with [0] members Mar 25 06:03:09 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:458]: wsconn_get_list(): wsconn_get_list Mar 25 06:03:09 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:502]: wsconn_get_list(): wsconn_get_list returns list [(nil)] with [0] members Mar 25 06:03:10 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:458]: wsconn_get_list(): wsconn_get_list Mar 25 06:03:10 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:502]: wsconn_get_list(): wsconn_get_list returns list [(nil)] with [0] members Mar 25 06:03:11 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:458]: wsconn_get_list(): wsconn_get_list Mar 25 06:03:11 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:502]: wsconn_get_list(): wsconn_get_list returns list [(nil)] with [0] members Mar 25 06:03:12 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:458]: wsconn_get_list(): wsconn_get_list Mar 25 06:03:12 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:502]: wsconn_get_list(): wsconn_get_list returns list [(nil)] with [0] members Mar 25 06:03:13 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:458]: wsconn_get_list(): wsconn_get_list Mar 25 06:03:13 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:502]: wsconn_get_list(): wsconn_get_list returns list [(nil)] with [0] members Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:458]: wsconn_get_list(): wsconn_get_list Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14034]: DEBUG: websocket [ws_conn.c:502]: wsconn_get_list(): wsconn_get_list returns list [(nil)] with [0] members Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14041]: CRITICAL: <core> [pass_fd.c:275]: receive_fd(): EOF on 21 Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [tcp_main.c:3448]: handle_ser_child(): dead child 14, pid 14031 (shutting down?) Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14041]: DEBUG: <core> [io_wait.h:598]: io_watch_del(): DBG: io_watch_del (0xa31e60, 21, -1, 0x0) fd_no=26 called Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14013]: ALERT: <core> [main.c:739]: handle_sigs(): child process 14031 exited by a signal 11 Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14013]: ALERT: <core> [main.c:742]: handle_sigs(): core was generated Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14013]: INFO: <core> [main.c:754]: handle_sigs(): terminating due to SIGCHLD Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14013]: DEBUG: <core> [main.c:756]: handle_sigs(): terminating due to SIGCHLD Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14038]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14040]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14037]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received Mar 25 06:03:14 P172 /usr/local/kamailio/sbin/kamailio[14035]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received There were no UAs that tried to register while this error occured. Thanking You, Sunil More Ph : 9503338275

2 4

Kamailio Dispatcher error msg 407
by Rex Lin (林昱頡) 28 Mar '16

28 Mar '16

Hi admins, I'm kinda new in this group and I'm not quite sure if I'm posting in a correct way, advise if I'm wrong. below is my post. ------------------------------------------------------------------------------------------------------------------------------------ I would like to activate Dispatcher module for apply Load Balance functionality purpose. but it's been hard to complete so far, I wonder if you can give me some advise in below situation. we have 2 Kamailios, the version is 4.3.3 ; we use domain for the SIP registration. we want to make Kamailios work in double active mode for SIP load balancing, we have a F5 to do the domain name round robin to make two UAC regitering under two UAS. I have put the "route(DISPATCH) b4 PSTN in request_route. when I try to make UAC-A call to UAC-B, it appears 407 proxy authentication required. we wiresharked, UAS has sent INVITE to UAS-B, but UAS-B sent the error code 407 to UAS-A Is the configuration has to be set to "UAS-B accepting INVITE of UAS-A" ? or the UAS-A should carrying the auth info into UAS-B ? detail configuration as below. ------------------------------------------------------------------------------------------------------------------------------------ dispatcher params modparam("dispatcher", "list_file", "/usr/local/etc/kamailio/dispatcher.list") modparam("dispatcher", "flags", 1) modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") modparam("dispatcher", "sock_avp", "$avp(AVP_SOCK)") modparam("dispatcher", "ds_ping_interval", 30) modparam("dispatcher", "ds_ping_from", "sip:kamailio1@awesomedomain.com") modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_probing_threshold", 1) DISPATCHER Routing route[DISPATCH] { # round robin dispatching on gateways group '1' if(!ds_select_dst("1", "4")) { send_reply("404", "No destination"); exit; } xdbg("--- SCRIPT: going to <$ru> via <$du>\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } # Re-route in case of failure failure_route[RTF_DISPATCH] { if (t_is_canceled()) { exit; } # next node - only for 500 or local timeout if (t_check_status("500") || (t_branch_timeout() && !t_branch_replied())) { if(ds_next_dst()) { t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } Best Regards, Rex Lin

2 3

Custom Authentication module with Java Code
by Jungtae Kim 28 Mar '16

28 Mar '16

Hello, I'm new to Kamailio. I'm trying to make my own auth module to check if incoming SIP messages are sighed with a correct key. I want to do this by calling a Java code snippet. I found app_java module but I can't figure out if this will support my usecase. Does app_java allow me to call Java code from another module? If not, what can I do to meet my use case? My token verification code is written in Java, so I'd like to reuse my code if possible. Thanks

2 1

Kamailio return response
by Shiv Patidar 28 Mar '16

28 Mar '16

i run client socket program for send SIP request to kamailio server kamailio registration is done successfully with 200OK and i want to store 200OK in client.c program in structure like struct reg_db { char *to char *from char ..... ....... } please help me how to store ?

2 1

Two different Database at the same time
by Barış Şekerciler 28 Mar '16

28 Mar '16

Hello,I want to use Cassandra and PGSQL at the same time in Kamailio. Is that possible?For example, for CDR records I want to use Cassandra and for other things I want to use PostgreSQL. Regards.

2 4

Re: [SR-Users] Kamailio PostgreSQL Asking Password Everytime
by Barış Şekerciler 26 Mar '16

26 Mar '16

Hello Daniel,Thanks for your reply. So I can't see any password for root user option in the kamctlrc file by default.How can I add? Regards.Barış. To: sr-users(a)lists.sip-router.org From: miconda(a)gmail.com Date: Wed, 23 Mar 2016 08:26:58 +0100 Subject: Re: [SR-Users] Kamailio PostgreSQL Asking Password Everytime Hello, I haven't used the postgres in a long while, but I think it should ask only once. Can you try by setting the password inside the kamctlrc file and see if works straight away? Cheers, Daniel On 22/03/16 11:52, Barış Şekerciler wrote: Hello, I have one postgres server at one server and Kamailio at another server. So I believe I did true configuration to the Kamailio. Actually I made some tests and I could create user and authenticated etc. But I have a problem. When I tried to present "kamdbctl create" Kamailio asks password 50 times (almost, much or less idk certainly). After 50 times Db created successfully and works. And it looks like this: root@bs-kamailio:/home/bs-kamailio# kamdbctl create INFO: creating database kamailio ... Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: So how can I solve this situation? I want Kamailio to asks password one time and handle it's job. Regards. _______________________________________________ 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://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.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 3

Kamailio PostgreSQL Asking Password Everytime
by Barış Şekerciler 25 Mar '16

25 Mar '16

Hello,I have one postgres server at one server and Kamailio at another server.So I believe I did true configuration to the Kamailio. Actually I made some tests and I could create user and authenticated etc. But I have a problem. When I tried to present "kamdbctl create" Kamailio asks password 50 times (almost, much or less idk certainly).After 50 times Db created successfully and works. And it looks like this: root@bs-kamailio:/home/bs-kamailio# kamdbctl create INFO: creating database kamailio ... Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: Password for user postgres: So how can I solve this situation? I want Kamailio to asks password one time and handle it's job. Regards.

3 2

Custom Authentication module with Java code
by Jungtae Kim 25 Mar '16

25 Mar '16

Hello, I'm new to Kamailio. I'm trying to make my own auth module to check if incoming SIP messages are sighed with a correct key. I want to do this by calling a Java code snippet. I found app_java module but I can't figure out if this will support my usecase. Does app_java allow me to call Java code from another module? If not, what can I do to meet my use case? My token verification code is written in Java, so I'd like to reuse my code if possible. Thanks

1 0

Re: [SR-Users] Multiple LCR instances with LCR module
by DanB 25 Mar '16

25 Mar '16

Hey Juha, Many thanks for so quick feedback! Changing the module param did the trick. Have a good one! DanB

1 0

Problem with authdb and view(?)
by Jon Bonilla (Manwe) 25 Mar '16

25 Mar '16

Hi all I've created a "subscriber" table which is a view of a asterisk sippeers table. The view is created as follows: create view subscriber as select id, name as username, fromdomain as domain, secret as password, email as email_address, md5secret as ha1, md5secret as ha1b, NULL as rpid from sippeers where type="friend"; and one of the resulting subscribers looks like this: *************************** 5. row *************************** id: 100022 username: user domain: domain.com password: plain or empty email_address: user(a)domain.com ha1: 3a4f5cc8dc0efb0d899cawe4dec9ea08 ha1b: 3a4f5cc8dc0efb0d899cawe4dec9ea08 rpid: NULL But when I try to register I get ERROR in the logs: New request - M=REGISTER R=sip:domain.com F=sip:user@domain.com ERROR: db_mysql [km_dbase.c:123]: db_mysql_submit_query(): driver error on query: Unknown column 'md5secret' in 'field list' ERROR: <core> [db_query.c:132]: db_do_query_internal(): error while submitting query ERROR: auth_db [authorize.c:187]: get_ha1(): failed to query database Is there any problem with views? Why the logs talks about md5secret column when it should see ha1? cheers, Jon

1 1

Multiple LCR instances with LCR module
by DanB 25 Mar '16

25 Mar '16

Hey Guys, I was wondering if any of you use multiple LCR instances with LCR module since I could see that the documentation allows it, however for some reason the queries to load data (both on restart or kamcmd lcr.reload) somehow hardcode the value of 1 inside: """ select `gw_name`,`ip_addr`,`port`,`uri_scheme`,`transport`,`params`,`hostname`,`strip`,`prefix`,`tag`,`flags`,`defunct`,`id` from `lcr_gw` where `lcr_id`=0 select `gw_name`,`ip_addr`,`port`,`uri_scheme`,`transport`,`params`,`hostname`,`strip`,`prefix`,`tag`,`flags`,`defunct`,`id` from `lcr_gw` where `lcr_id`=1 select `id`,`prefix`,`from_uri`,`stopper`,`enabled`,`request_uri` from `lcr_rule` where `lcr_id`=1 select `rule_id`,`gw_id`,`priority`,`weight` from `lcr_rule_target` where `lcr_id`=1 """ My kamailio version: """ # kamailio -V version: kamailio 4.3.5 (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 4.9.2 """ Many thanks in advance for any kind of tip! DanB

2 1

kamailio memory issue
by Volodymyr Rykhva 25 Mar '16

25 Mar '16

Hi, I have started using kamailio recently. Added to db almost 100K users. And now I am trying to increase shared and private memory with this command when server is running: kamailio -M 12 -m 128. But then I check it with `kamailio -l` and the same values (64 and 8 respectively) are there. Can I change it with config? or what I do wrong. And what's the purpose of child process? - Will kamailio process requests faster if I set child=32 (for example)? Thanks and best regards, Volodymyr

2 1

Re: [SR-Users] htable - banning users
by meres 24 Mar '16

24 Mar '16

Thank u Vasiliy and Daniel, I used -3 condition as you suggested! Regards, meres

1 0

dynamic weighting for dispatcher
by Brooks Bridges 24 Mar '16

24 Mar '16

So I'm building a load balancer solution that needs to monitor remote devices at a specific interval via snmp for certain data (channels and cps) and use that data to create a dynamically updated weight for each of the destinations. I can't find any reasonable mechanism for updating the weight attribute in the dispatcher module itself without having to do some hacky "update the database column then trigger dispatcher.reload" script, which I very much want to avoid. Is this something I should just be thinking of ditching the dispatcher module and rolling my own design or am I missing something obvious? Brooks Bridges | Sr. Voice Services Engineer O1 Communications 5190 Golden Foothill Pkwy El Dorado Hills, CA 95762 office: 916.235.2097 | main: 888.444.1111, Option 2 email: bbridges(a)o1.com<mailto:bbridges@o1.com> | web: www.o1.com<http://www.o1.com/>

2 2

Dispatching Registrations to Weight based servers
by SamyGo 24 Mar '16

24 Mar '16

Hi All, I'm having a wee little bit of difficulty in trying to load-balance registrations to multiple servers based on their weights. Kind of similar scenario as discussed in this thread: http://lists.sip-router.org/pipermail/sr-users/2014-December/086235.html The problem I'm facing is that dispatcher is always choosing the first destination from the group. I do suspect that dispatcher is not considering registrations as load at all and only counts INVITES as load. If above is correct then is there a way to use dispatcher to count registrations as load ? Thanks, Sammy

2 3

Kamailio 5.0 - B2BUA
by Alex Balashov 24 Mar '16

24 Mar '16

Hi, I wanted to raise the possibility of an inline signalling-only B2BUA component to Kamailio. I know that's an extremely poor fit for Kamailio, and not at all what it's supposed to do. And there are many things about the OpenSIPS B2BUA module that reveal how awkwardly it is situated, as a square peg in a round hole. I myself am philosophically opposed to a B2BUA in Kamailio to the threshold of physical violence. However, the reason I bring it up for discussion is that there are very few viable, high-performance signalling-only B2BUA alternatives that are FOSS. There is enormous demand for a B2BUA, mainly for topology concealment purposes, from the short-duration traffic industry, which many users of Kamailio deal with. It's why a lot of them end up going to OpenSIPS; they need the B2BUA, and prefer to consolidate on one OpenSER brand. Anyone who has tried to run high-CPS traffic through the existing FOSS signalling-only B2BUAs out there situated in front of Kamailio has discovered, sometimes in a very financially painful way, that: - FreeSWITCH falls over at around +/- 300 CPS. This is with RTP relay disabled -- signalling-only. It requires horizontally scaling a large number of FreeSWITCH boxes to meet a capacity requirement of, say, 2000 CPS, which is unfortunate given the favourable proposition offered by Kamailio for the infrastructure unit economics. In other words, it's ironic to have to build a fleet of 10 FreeSWITCH boxes for the 1% problem of topology concealment when Kamailio can otherwise churn through 2000 CPS with no issues. - SEMS' 'sbc' module is a good candidate and can handle the load, but Frafos offers practically no support for it, with all efforts focused on their commercial ABC SBC product. That's very understandable, but just not practical given the high technical knowledge SEMS requires to deploy and maintain in this capacity. - None of the other userspace B2BUA folk traditions can handle the load. So, like I said, I personally recoil in shock and horror at the idea of introducing a B2BUA into something that was designed to be anything but a B2BUA. But, there is a huge market opportunity for this functionality in North America, and at the moment, users who need it are mostly ending up in OpenSIPS land. -- Alex -- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

8 15

Presence Publish expires
by Igor Olhovskiy 24 Mar '16

24 Mar '16

Hi! Is there any possibility to set Expires in PUBLISH message to never in Presence module? I want to have BLF’s that indicates states, that driven via PUBLISH, but can be really irregular (let’s say once a week) -- Best regards, Igor

3 5

dialog rpc functions.
by Yasin CANER 24 Mar '16

24 Mar '16

1 0

kamailio in a multi-site configuration
by Sébastien Brice 24 Mar '16

24 Mar '16

Hi everyone i am installing Kamailio to scale asterisk on a remote hosted server in a way to manage calls dialogs in a real-time fashion. (and to manage many customers) Several users (maybe 80) are working at the head-office and i want them register their hardphone on the remote Kamailio Ipbx, to free them of ours end-of-life PABX (alcatel-Lucent). Can kamailio help me to achieve that ? What kind of software have I to install locally (mysql to replicate the master mysql db? a replicated asterisk instance?) ? Taking into consideration disaster recovery plan What could be my options to maintain internal call if my internet provider get down by exemple. thx you for your idea. Sébastien BRICE VoIP, Support et Intégration

3 2

set_advertised_address() / set_advertised_port() + PVs?
by Alex Balashov 23 Mar '16

23 Mar '16

Hello, I have a need to deploy Kamailio in AWS in a scenario of this sort: [Public Internet] <---> Kamailio <---> (Internal AWS servers) In such a scenario, Kamailio would be multihomed. Furthermore, in addition to sending out of two distinct network interfaces, the networking idiosyncrasies of AWS needs to be taken into account. AWS hosts only have an RFC1918 address homed natively, front-ended by 1:1 NAT externally. Normally, this is taken care of by advertised_address, which allows Kamailio to make outward representations about the network address by which it can be reached that are different to the IP to which it is bound. The problem here is that I need to do this conditionally, only on traffic going out the public interface. It should not be done on messages going to the "internal AWS servers". So, the questions that arise are: 1. Is this sane? Any unforeseen effects, e.g. vis-a-vis RR, provided enable_double_rr is enabled and that two genuinely different network interfaces are used? 2. Do set_advertised_address()/set_advertised_port() accept PV arguments, or are they pre-PV "core function folk traditions" in the same way as rewritehostport() and force_send_socket()? Thanks! -- Alex -- Alex Balashov | Principal | Evariste Systems LLC 1447 Peachtree Street NE, Suite 700 Atlanta, GA 30309 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

3 10

Git clone failing?
by Richard Good 23 Mar '16

23 Mar '16

Hi Is anyone else struggling with git clone? richard@richard-laptop-new:~/Smile-dev/git_code$ git clone --depth 1 --no-single-branch git://git.kamailio.org/kamailio kamailio Cloning into 'kamailio'... fatal: read error: Connection reset by peer Tried on a few machines our side and keep getting connection reset. Regards *Richard Good* *Senior Manager: Applications & Services* *Smile Communications Pty (Ltd)* *Mobile:* +27 (0) 72 389 8365 * Skype:* richard.a.good richard.good(a)smilecoms.com <name.surname(a)smilecoms.com> www.smilecoms.com -- This email is subject to the disclaimer of Smile Communications at http://www.smilecoms.com/home/email-disclaimer/ <http://www.smilecoms.com/disclaimer>

2 1

Random number generation kamailio
by Cibin Paul 23 Mar '16

23 Mar '16

Hi, How can I generate a random number say of length 10-14 in Kamailio. Can I use cfgutils for this. Please advise Regards Cibin

4 11

Feature request for Dialog terminate
by Yasin CANER 23 Mar '16

23 Mar '16

1 0

Kamailio Bridge
by Nelson Migliaro 23 Mar '16

23 Mar '16

Hello, I am trying to set up a Kamailio in Bridge Mode with two interfaces using rtpproxy. The problem I have is SDP information port and IP is not updated from outside to inside. But on the other hand the SDP is rewrited from inside to outside. In the rtpproxy log I am seeing this error: Mar 21 14:21:21 kamailio rtpproxy[6119]: ERR:create_twinlistener: can't bind to the IPv4 port 23414: Cannot assign requested address Mar 21 14:21:21 kamailio rtpproxy[6119]: ERR:handle_command: can't create listener and in kamailio log: ERROR: rtpproxy [rtpproxy.c:2727]: force_rtp_proxy(): incorrect port 0 in reply from rtp proxy Something strange and interesting is the port that shows up in the RTPPROXY error (23414) does not corresponds to any ports used. Thank you Nelson.- --------------------------------------------------------------------------------------------------------------------------------------------- Everything is set up this way: Kamailio Asterisk (asterisk-inside-ip) <-> Kamailio (inside-kamailio-ip) / Kamailio (outside-kamailio-ip) <-> sip-vendor-ip listen=inside-kamailio-ip listen=outside-kamailio-ip mhomed=1 if(src_ip=="asterisk-inside-ip"){ rtpproxy_manage("faie"); } if(src_ip=="sip-vendor-ip"){ rtpproxy_manage("faie"); } rtpproxy OPTIONS="rtpproxy -l inside-kamailio-ip/outside-kamailio-ip -m 20000 -M 30000 -u rtpproxy:rtpproxy -p /var/run/rtpproxy/rtpproxy.pid -s udp: 127.0.0.1:7722 -d DBUG:LOG_LOCAL6" ---------------------------------------------------------------------------------------------------------------------------------------------- SIP TRACES 2016/03/21 14:46:52.121472 asterisk-inside-ip:5060 -> inside-kamailio-ip:5060 INVITE sip:999999999@inside-kamailio-ip SIP/2.0 Via: SIP/2.0/UDP asterisk-inside-ip:5060;branch=z9hG4bK7f1ece8e;rport Max-Forwards: 70 From: 111111111<sip:1000@asterisk-inside-ip>;tag=as497457e6 To: <sip:999999999@inside-kamailio-ip> Contact: <sip:1000@asterisk-inside-ip:5060> Call-ID: 57bd4ca457972f5a42502ba34bba4500@asterisk-inside-ip:5060 CSeq: 102 INVITE User-Agent: Abantix Date: Mon, 21 Mar 2016 13:46:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Remote-Party-ID: <sip:888888888@local-domain ;user=phone>;privacy=off;party=calling Content-Type: application/sdp Content-Length: 251 v=0 o=root 478097443 478097443 IN IP4 asterisk-inside-ip ########## ASTERISK IP s=asterisk c=IN IP4 asterisk-inside-ip ########## ASTERISK IP t=0 0 m=audio 17146 RTP/AVP 8 101 ########## ASTERISK PORT a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv -------------------------------------------------------------------------------------------------------------------------------------------------------- 2016/03/21 14:46:52.139435 outside-kamailio-ip:5060 ->sip-vendor-ip:5060 INVITE sip:999999999@sip-vendor-ip SIP/2.0 Record-Route: <sip:outside-kamailio-ip;r2=on;lr=on;ftag=as497457e6;vsf=AAAAAAMEAAAACXIEAW5fXkFYWUwAUkM-;vst=AAAAAAcHAw8MDgsAAHZBKUEeQF5DRUFMGEtCMDY-;nat=yes> Record-Route: <sip:inside-kamailio-ip;r2=on;lr=on;ftag=as497457e6;vsf=AAAAAAMEAAAACXIEAW5fXkFYWUwAUkM-;vst=AAAAAAcHAw8MDgsAAHZBKUEeQF5DRUFMGEtCMDY-;nat=yes> Via: SIP/2.0/UDP outside-kamailio-ip;branch=z9hG4bK0bfd.7bd4d3ae01e5850c14a3a939a612a05a.0 Via: SIP/2.0/UDP asterisk-inside-ip:5060;received=asterisk-inside-ip;branch=z9hG4bK7f1ece8e;rport=5060 Max-Forwards: 69 From: 1111111111 <sip:1111111111@sip-vendor-ip>;tag=as497457e6 To: <sip:999999999@sip.sip-vendor-ip> Contact: <sip:1000@asterisk-inside-ip:5060> Call-ID: 57bd4ca457972f5a42502ba34bba4500@asterisk-inside-ip:5060 CSeq: 102 INVITE User-Agent: Abantix Voice Sevices Date: Mon, 21 Mar 2016 13:46:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Remote-Party-ID: <sip:888888888@local-domain ;user=phone>;privacy=off;party=calling Content-Type: application/sdp Content-Length: 271 v=0 o=root 478097443 478097443 IN IP4 outside-kamailio-ip ########## OUTSIDE KAMAILIO IP REWRITED BY RTPPROXY s=asterisk c=IN IP4 outside-kamailio-ip ########## OUTSIDE KAMAILIO IP REWRITED BY RTPPROXY t=0 0 m=audio 24122 RTP/AVP 8 101 ########## OUTSIDE KAMAILIO PORT REWRITED BY RTPPROXY a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv -------------------------------------------------------------------------------------------------------------------------------------------------------- 2016/03/21 14:46:57.393518sip-vendor-ip:5060 -> outside-kamailio-ip:5060 SIP/2.0 180 Ringing Via: SIP/2.0/UDP outside-kamailio-ip;rport=5060;branch=z9hG4bK0bfd.7bd4d3ae01e5850c14a3a939a612a05a.1 Via: SIP/2.0/UDP asterisk-inside-ip:5060;received=asterisk-inside-ip;branch=z9hG4bK7f1ece8e;rport=5060 From: 1111111111 <sip:1111111111@sip-vendor-ip>;tag=as497457e6 To: <sip:999999999@sip.sip-vendor-ip>;tag=fkii1xm4-CC-26 Call-ID: 57bd4ca457972f5a42502ba34bba4500@asterisk-inside-ip:5060 CSeq: 102 INVITE Record-Route: <sip:sip-vendo-ip;lr;ftag=as497457e6;vsf=AAAAAAEECQkCAgsNAXBeLxwZHQxBH11lcw--;did=0e.126;nat=yes> Record-Route: <sip:outside-kamailio-ip;r2=on;lr=on;ftag=as497457e6;vsf=AAAAAAMEAAAACXIEAW5fXkFYWUwAUkM-;vst=AAAAAAcHAw8MDgsAAHZBKUEeQF5DRUFMGEtCMDY-;nat=yes> Record-Route: <sip:inside-kamailio-ip;r2=on;lr=on;ftag=as497457e6;vsf=AAAAAAMEAAAACXIEAW5fXkFYWUwAUkM-;vst=AAAAAAcHAw8MDgsAAHZBKUEeQF5DRUFMGEtCMDY-;nat=yes> Contact: <sip:999999999@sip-vendor-carrier:5060;user=phone;transport=udp> Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Content-Length: 237 Content-Type: application/sdp v=0 o=HuaweiSoftX3000 71167517 71167517 IN IP4sip-vendor-ip ########## SIP VENDOR IP s=Sip Call c=IN IP4sip-vendor-ip ########## SIP VENDOR IP t=0 0 m=audio 44862 RTP/AVP 8 101 ########## SIP VENDOR PORT a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=ptime:20 a=fmtp:101 0-15 a=nortpproxy:yes -------------------------------------------------------------------------------------------------------------------------------------------------------- 2016/03/21 14:46:57.395029 inside-kamailio-ip:5060 -> asterisk-inside-ip:5060 SIP/2.0 180 Ringing Via: SIP/2.0/UDP asterisk-inside-ip:5060;received=asterisk-inside-ip;branch=z9hG4bK7f1ece8e;rport=5060 From: "1000" <sip:1000@asterisk-inside-ip>;tag=as497457e6 To: <sip:999999999@inside-kamailio-ip>;tag=fkii1xm4-CC-26 Call-ID: 57bd4ca457972f5a42502ba34bba4500@asterisk-inside-ip:5060 CSeq: 102 INVITE Record-Route: <sip:sip-vendo-ip;lr;ftag=as497457e6;vsf=AAAAAAEECQkCAgsNAXBeLxwZHQxBH11lcw--;did=0e.126;nat=yes> Record-Route: <sip:outside-kamailio-ip;r2=on;lr=on;ftag=as497457e6;vsf=AAAAAAMEAAAACXIEAW5fXkFYWUwAUkM-;vst=AAAAAAcHAw8MDgsAAHZBKUEeQF5DRUFMGEtCMDY-;nat=yes> Record-Route: <sip:inside-kamailio-ip;r2=on;lr=on;ftag=as497457e6;vsf=AAAAAAMEAAAACXIEAW5fXkFYWUwAUkM-;vst=AAAAAAcHAw8MDgsAAHZBKUEeQF5DRUFMGEtCMDY-;nat=yes> Contact: <sip:999999999@sip-vendor-carrier:5060;user=phone;transport=udp> Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Content-Length: 237 Content-Type: application/sdp v=0 o=HuaweiSoftX3000 71167517 71167517 IN IP4sip-vendor-ip ########## SIP VENDOR IP NOT REWRITED BY RTPPROXY s=Sip Call c=IN IP4sip-vendor-ip ########## SIP VENDOR IP NOT REWRITED BY RTPPROXY t=0 0 m=audio 44862 RTP/AVP 8 101 ########## SIP VENDOR PORT NOT REWRITED BY RTPPROXY a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=ptime:20 a=fmtp:101 0-15 a=nortpproxy:yes

2 2

Kamailio forward REGISTER return "general error"
by Cockhootec Yahrabee 23 Mar '16

23 Mar '16

Hi everyone, I am trying to forward registration from one kamailio instance to another one eg. UA1 <---> kamailioB <------> kamailioA I am trying to register on `kamailioA` with UA1. I have UA1 registered on kamailioA. I am routing SIP messages by adding this to `request_route` block: if (uri != myself) { forward(otherKamailioIP, 5060); } Where I am using as `otherKamilioIP on kamailioA the kamailioB IP address and vice versa. I can clearly see in Wireshark, that the SIP messages are routed correctly but I am getting 401 Unauthorized which is sent from kamailioA to kamailioB and then from kamailioB to UA1. I printed value of `auth_check` following this http://lists.sip-router.org/pipermail/sr-users/2014-November/085851.html and I am getting return code of `-1` which is according to the documentation http://kamailio.org/docs/modules/4.2.x/modules/auth_db.html#auth_db.f.www_a… the GENERIC ERROR. I don't know what to do next. Does anyone encounter this kind of error before? Thanks in advance

2 3

processing ACK
by jaflong jaflong 23 Mar '16

23 Mar '16

Hi Everyone I am manually processing sdp and handling media so I want to repond to ACK myself instead of forwarding on for media server. I catch the method=="ACK" condition. From within this condition how do I retrieve avp variables that were set in the INVITE. Requesting $avp("myvar") comes out as null. Also what function can I run so the ACK is replied to properly so it stops resending 200 OK of the Invite.

5 6

for kamailio registration
by Shiv Patidar 23 Mar '16

23 Mar '16

For registration , we would like to see the complete traces at all nodes whoever processing UE sending REGISTER method. could you please let me know how can we check the logging at all IMS nodes? i am able to successfully register and got 200OK but i want understand what and all processing has been done by server? can't i see default messages with the way install step i followed i need this for test/development

4 4

Kamailio 4.3.x kamailio-redis RPM
by Jay Patel 23 Mar '16

23 Mar '16

Is there any trusted yum repo which builds redis module RPM for EL6/CentOS? Seems like default make file does not build redis and it'snot published on rpm.kamailio.org. cnxcc will be added bonus. Thanks in advance. Regards, Jay

2 3

Kamctl dispatcher reload
by Alberto Sagredo 22 Mar '16

22 Mar '16

I have installed Kamailio 4.4.x and im getting this error when trying to reload dispatcher? Any idea where to look? kamctl dispatcher reload 500 ERROR Reloading data Also kamcmd dispatcher.reload gave me same result.

1 1

Kamailio World 2016 for students
by Daniel-Constantin Mierla 22 Mar '16

22 Mar '16

Hello, continuing the program from last years, based on the roots and the tight relation of Kamailio project with the academic environment, we are offering three seats at Kamailio World Conference, May 18-20, 2016, in Berlin, to students enrolled in universities or research institutes (both bachelor and PhD programs qualify). Last year we had students participating from Austria, Slovakia and Netherlands. If you are a student and want to participate, email me directly (miconda(a)gmail.com) or to registration(a)kamailio.org . Participation to all the content of the event (workshops, conference and social event) is free, but you will have to take care of expenses for traveling and accommodation. Write a short description about your interest in real time communications and what is the university or the research institute you are affiliate to. Also, if you are not a student, but you are in touch with some or have access to students forums/mailing lists, it will be very appreciated if you forward these details. More information about Kamailio World is available on the web site: - http://www.kamailioworld.com Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 18-20, 2016 Berlin, Germany - http://www.kamailioworld.com

1 0

Database engine not found
by Shiv Patidar 22 Mar '16

22 Mar '16

when i run this command kamctl fifo debug 6 then i got WARNING: database engine not found tried 'MYSQL'

2 1

Kamailio 5.0 - Better way of dealing with stateless CANCELs
by Alex Balashov 21 Mar '16

21 Mar '16

Currently (AFAIK), restarting Kamailio amidst production call processing is basically "safe"; except for no availability during the few seconds it takes to restart (which will just result in retransmissions until it is available), most things will happen "correctly" after restart even though TM state has been lost: (1) Initial requests will be routed as initial requests always were; (2) In-dialog requests will be loose-routed as sequential requests always were; (3) Replies to open transactions will fall back to stateless routing but will be delivered correctly to their destinations based on SIP fundamentals (i.e. Via). (4) rtpproxy & rtpengine control messages are grouped by Call-ID, so also stateless. If the proper destroy/remove functions are not called from failure_route[] due to lack of TM state, it's not so bad; rtpproxy & rtpengine will see an RTP timeout after a while and expire the bindings on their own. If dialog state is used, it will be lost, but assuming one is willing to live with that, it's okay. I don't know if there has been any work done to create a persistence layer for dialog that can be re-read completely on startup, and if it actually works - does it? - but it's a relatively small price to pay if it's important to integrate a change into production in the middle of the day. The one exception is CANCEL handling. CANCEL is a special animal, since it's a hop-by-hop (branch-level) request, so CANCELs sent from a caller apply to the 'caller -> Kamailio' branch. Kamailio generates separate CANCELs endogenously for one or more 'Kamailio -> gateway' branches. Stateful CANCEL handling with TM is implemented using t_check_trans() or t_relay_cancel(). For example, in the stock config[1] # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } Or, as in our case, more folklorically: if(is_method("CANCEL")) { if(!t_relay_cancel()) { # Corresponding INVITE transaction found, but error # occurred. sl_send_reply("500", "Internal Server Error"); exit; } # Corresponding INVITE transaction for CANCEL was not # found. exit; } In both cases, the corresponding INVITE transaction must exist. Unfortunately, there's no good alternative. According to RFC 3261 Section 16.11 ("Stateless Proxy"): Stateless proxies MUST NOT perform special processing for CANCEL requests. They are processed by the above rules as any other requests. So, in other words, route_logic(CANCEL) == route_logic(initial INVITE). Sometimes, this is possible - with considerable config logic labour - but other times the path taken by the CANCEL is not so deterministic, as for example with round-robin load balancing, random distribution, complex LCR, etc. One is basically left in these situations with the choice of implementing one's own Call-ID/branch => destination persistent state database of some kind, which is, to put it mildly, complicated and undesirable. Now, if the INVITE transaction receives a final negative reply, this will get back to the calling UAC, and it will process it correctly. However, some calls get answered with 2xx. Many UACs will behave reasonably in this situation: when they don't receive a 200 OK for their CANCELs but later receive an answer, they will go ahead and send the end-to-end ACK, then BYE the call. However, they cannot be reliably counted upon to do this. Some simply get drop the INVITE transaction after their CANCEL has gone unreplied for a short time, regardless of whether they receive a final negative reply for it. Is there a better way? Perhaps a feature can be devised by which Kamailio keeps some kind of lightweight and restart-persistent map to which to send the CANCELs? Or perhaps TM is due for a feature that allows the shm transaction table to be dumped to disk and persisted across restarts? Comments welcome. Also, if I'm missing something, please let me know! -- Alex [1] https://github.com/kamailio/kamailio/blob/master/etc/kamailio.cfg#L466 -- Alex Balashov | Principal | Evariste Systems LLC 1447 Peachtree Street NE, Suite 700 Atlanta, GA 30309 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

3 8

Missing pipelimit values in data returned by JSONRPC-S
by Brooks Bridges 21 Mar '16

21 Mar '16

When trying to pull a list of pipes and their current values by using pl.dump via the JSONRPC-S module, I am only getting the last result in the reply as opposed to an array. If I use the CLI RPC interface I get the data I expect. Here's a sanitized example: [root@server1 ~]# kamcmd pl.stats PIPE: id=user_1 load=0 counter=29 PIPE: id=user_2 load=0 counter=108 PIPE: id=user_3 load=0 counter=18 PIPE: id=user_4 load=0 counter=0 PIPE: id=user_5 load=0 counter=1 PIPE: id=user_6 load=0 counter=2 PIPE: id=user_7 load=0 counter=0 PIPE: id=user_8 load=0 counter=84 PIPE: id=user_9 load=0 counter=26 PIPE: id=user_10 load=0 counter=0 PIPE: id=user_11 load=0 counter=23 PIPE: id=user_12 load=0 counter=9 PIPE: id=user_13 load=0 counter=23 PIPE: id=user_14 load=0 counter=0 PIPE: id=user_15 load=0 counter=107 PIPE: id=user_16 load=0 counter=10 PIPE: id=user_17 load=0 counter=28 PIPE: id=user_18 load=0 counter=45 PIPE: id=user_19 load=0 counter=7 PIPE: id=user_20 load=0 counter=0 PIPE: id=user_21 load=0 counter=12 PIPE: id=user_22 load=0 counter=94 PIPE: id=user_23 load=0 counter=14 PIPE: id=user_24 load=0 counter=7 PIPE: id=user_25 load=0 counter=18 PIPE: id=user_26 load=0 counter=46 PIPE: id=user_27 load=0 counter=45 PIPE: id=user_28 load=0 counter=50 PIPE: id=user_29 load=0 counter=2 PIPE: id=user_30 load=0 counter=2 [root@server1 ~]# curl -s --header 'Content-Type: application/json' --data-binary '{"jsonrpc": "2.0", "method": "pl.stats"}' http://1.2.3.4:5060/jsonrpc {"jsonrpc":"2.0","result":"PIPE: id=user_30 load=0 counter=2"} [root@server1 ~]# The jsonrpc config and event route is about as basic as you can get: modparam("jsonrpc-s", "pretty_format", 0) modparam("jsonrpc-s", "transport", 1) event_route[xhttp:request] { xlog("L_INFO", "[R-HTTP-REQUEST:$si] -> Got HTTP request\n"); if($hu =~ "^/jsonrpc") { xlog("L_INFO", "[R-XHTTP-REQUEST:$si] -> Handling JSON RPC request: $hu\n"); jsonrpc_dispatch(); } else { xhttp_reply("404", "Not Found", "text/html", ""); } } Should the JSONRPC-S module be returning this as an array instead of just a single value? Brooks Bridges | Sr. Voice Services Engineer O1 Communications 5190 Golden Foothill Pkwy El Dorado Hills, CA 95762 office: 916.235.2097 | main: 888.444.1111, Option 2 email: bbridges(a)o1.com<mailto:bbridges@o1.com> | web: www.o1.com<http://www.o1.com/>

3 10

Removing Via and Record-Route headers
by Marrold 21 Mar '16

21 Mar '16

Hi, Is there any way to remove Via and Record-Route headers from requests sent to an endpoint and update the contact header, but have Kamailio statefully remember where the replies need to route? I imagine this would involve mangling the packets slightly to behave similar to a B2BUA. My motivations, in order of priority - 1) Reduce UDP packet size to avoid MTU limitations 2) Improve interoperability, don't expect end points to be able to properly parse / process many via / record-route headers 3) Hide topology. If someone could point me in the right direction in terms of configuration or modules, that would be great. Thanks

4 10

What to read after RFC 3261 to get started with Kamailio.
by Oivvio Polite 21 Mar '16

21 Mar '16

I want to get started with Kamailio. So far I've understood that I really need to grok SIP before tackling Kamailio so I'm reading the book "SIP demystified" as the starting point of my Kamailio journey. After that I will read RFC 3261. After that I think it's time to read something that will give me good understanding of how to actually implement applications with Kamailio. Searching Amazon for Kamailio returns "Building Telephony Systems with OpenSER" as the results that seems to be most specific to Kamailio. This book is from 2008. Is it still relevant or is it better to go with the online docs? regards, Oivvio

3 4

problem with relaying after t_susped() and t_continue()
by Uri Shacked 21 Mar '16

21 Mar '16

I have a problem when using t_suspend() and t_continue(). Let's say I would like to have a 5 sec delay before relaying an INVITE. For example the following code: * route[MYROUTE] {* * t_newtran();* * if (is_method("INVITE")) {* * t_reply("100", "Trying");* * t_reply("180", "In Queue");* * $sht(a=>$ci::t_index) = $T(id_index);* * $sht(a=>$ci::t_label) = $T(id_label);* * $sht(a=>$ci::t_suspended) = 1;* * xlog("L_WARN","SUSPENDING t_index = $sht(a=>$ci::t_index) and t_label = $sht(a=>$ci::t_label)");* * t_suspend();* * sleep(5);* * if (t_is_canceled()) drop;* * $sht(a=>$ci::t_suspended) = 0;* * t_continue("$avp(t_index)","$avp(t_label)","1");* * }* * route(RELAY);* * exit;* * }* * route[1] {* * xlog("L_WARN","After transaction continue\n");* * route(RELAY);* * exit;* * }* When using this code, I get the following error: Mar 9 16:49:32 mysip /usr/local/sbin/kamailio[5260]: WARNING: <script>: After transaction continue Mar 9 16:49:32 mysip /usr/local/sbin/kamailio[5260]: ERROR: <core> [resolve.c:1693]: sip_hostport2su(): could not resolve hostname: "mysip" Mar 9 16:49:32 mysip /usr/local/sbin/kamailio[5260]: ERROR: tm [ut.h:319]: uri2dst2(): failed to resolve "mysip" Mar 9 16:49:32 mysip /usr/local/sbin/kamailio[5260]: ERROR: tm [t_fwd.c:1711]: t_forward_nonack(): ERROR: t_forward_nonack: failure to add branches Mar 9 16:49:32 mysip /usr/local/sbin/kamailio[5260]: ERROR: sl [sl_funcs.c:363]: sl_reply_error(): ERROR: sl_reply_error used: Unresolvable destination (478/SL) Mar 9 16:49:32 mysip /usr/local/sbin/kamailio[5260]: BUG: tm [t_lookup.c:1481]: t_unref(): tm: t_unref: REQ_ERR DELAYED should have been caught much earlier for 0x7fd4e727a468: 27 (hex 1b) If I try to t_continue() without any route: t_continue("$avp(t_index)","$avp(t_label)",""); The call is working great and the following error pops up: Mar 9 17:10:36 mysip /usr/local/sbin/kamailio[5830]: ERROR: tmx [tmx_mod.c:635]: w_t_continue(): empty action lists in route block [] Am I doing something wrong? My other question is regarding CANCEL. Let's assume I suspended the INVITE and in these 5 sec I received CANCEL for it. Is there a short and elegant way to remove the transaction from suspension? For now I do: * if (is_method("CANCEL")) {* * if (t_check_trans()) {* * if($sht(a=>$ci::t_suspended) == 1) {* * t_continue("$sht(a=>$ci::t_index)","$sht(a=>$ci::t_label)","");* * xlog("L_WARN","After transaction continue\n");* * }* * route(RELAY);* * }* * exit;* * }* Thanks, Uri

2 1

LCR prefix based
by Roberto Innaimi 21 Mar '16

21 Mar '16

Hi to all, i want to use prefix based LCR routing, but for some reason, something is not working as expected. Can anyone help me? This is the LCR configuration: #LCR MODULE loadmodule "lcr.so" modparam("lcr", "db_url", "mysql://kamailioro:kamailioro@localhost/kamailio") modparam("lcr", "lcr_gw_table","lcr_gw") modparam("lcr", "id_column", "id") modparam("lcr", "gw_name_column", "gw_name") modparam("lcr", "ip_addr_column", "ip_addr") modparam("lcr", "hostname_column", "hostname") modparam("lcr", "port_column", "port") modparam("lcr", "params_column", "params") modparam("lcr", "uri_scheme_column", "uri_scheme") modparam("lcr", "transport_column", "transport") modparam("lcr", "strip_column", "strip") modparam("lcr", "tag_column", "tag") modparam("lcr", "flags_column", "flags") modparam("lcr", "defunct_column", "defunct") modparam("lcr", "lcr_rule_table", "lcr_rule") modparam("lcr", "prefix_column", "prefix") modparam("lcr", "from_uri_column", "from_uri") modparam("lcr", "request_uri_column", "request_uri") modparam("lcr", "stopper_column", "stopper") modparam("lcr", "enabled_column", "enabled") modparam("lcr", "lcr_rule_target_table", "lcr_rule_target") modparam("lcr", "rule_id_column", "id") modparam("lcr", "gw_id_column", "gw_id") modparam("lcr", "priority_column", "priority") modparam("lcr","weight_column", "weight") modparam("lcr", "gw_uri_avp", "$avp(i:709)") modparam("lcr", "ruri_user_avp", "$avp(i:500)") modparam("lcr", "tag_avp", "$avp(lcr_tag)") modparam("lcr", "flags_avp", "$avp(i:712)") modparam("lcr", "lcr_id_avp", "$avp(s:lcr_id_avp)") modparam("lcr", "defunct_gw_avp", "$avp(s:defunct_gw_avp)") route[LCR] { if (status=="200") { xlog("LCR: Inside the LCR route\n"); } if(method=="INVITE") { xlog("We got an invite"); if(!load_gws(1, $rU, $var(caller_uri))) { xlog("Couldn't load gateways"); sl_send_reply("500", "Server Internal Error - Cannot load gateways"); exit; } else { xlog("GW Selected '$avp(i:709)'\n"); xlog("Domain of destination: $dd\n"); xlog("To URI: $tu\n"); } if(!next_gw()) { xlog("Couldn't proceed to next gateway"); sl_send_reply("503", "Service not available, no gateways found"); exit; } else { xlog("Calling the first matched gateway\n"); xlog("ruri_user_avp: '$avp(i:500)'\n"); xlog("To URI after next_gw: $tu\n"); xlog("Request URI: $rU\n"); } } } AND this is the error which i have got: 17(7483) ERROR: <script>: We got an invite17(7483) ERROR: <script>: GW Selected '1|sip:|3|||2873019861||5060||;transport=udp|1' 17(7483) ERROR: <script>: Domain of destination: <null> 17(7483) ERROR: <script>: To URI: sip:001326212223313@10.10.0.10 17(7483) ERROR: <script>: Calling the first matched gateway 17(7483) ERROR: <script>: ruri_user_avp: '001326212223313' 17(7483) ERROR: <script>: To URI after next_gw: sip: 001326212223313(a)10.10.0.10 17(7483) ERROR: <script>: Request URI: 326212223313 -- Roberto Innaimi roberto.innaimi(a)gmail.com

2 1

Problem with Freeswitch with IP public and Kamailio behind NAT
by Hai Bui Duc Ha 21 Mar '16

21 Mar '16

Hi all, I work on Kazoo system. I have NAT problem when configure Freeswitch and Kamailio. As the old network diagram: <https://lh3.googleusercontent.com/-dVJit4UPAL8/VuI4z1LBdKI/AAAAAAAABDo/zSG0…> I configure dispatcher to FS 1,2 with IP private and map IP public <-> IP private for each server via Router. This work fine. Then now, I want to improve FS performance so that I set up new FS with IP public: <https://lh3.googleusercontent.com/-YZl8d3HBo8E/VuI7ChgTGzI/AAAAAAAABD0/AQQW…> I configure dispatcher FS3 with IP public, connection between FS3 & ecallmgr via public IP is Okay but have problem with NAT: *When Kamailio foward SIP INVITE message to FS3 then FS3 response 200 OK message but it send to Ip private of Kamailio not IP public.* As Kamailio I have already configure advertise IP public: local.cfg listen=UDP_SIP advertise 125.212.212.40:5060 listen=TCP_SIP advertise 125.212.212.40:5060 View console log on Freeswitch, I see the ecallmgr response to FS3: 1. bridge({outbound_redirect_fatal="false",call_timeout=20, originate_timeout=20,local_var_clobber="true"}[ecallmgr_Authorizing-ID= "71310939684a2c8acc4e0e6dff1be6c2",ecallmgr_Owner-ID= "188cd101dcea060d080261b328fdf1ca",ecallmgr_Account-ID= "3f221d1ce6959ec04acd372923e253b9",sdp_secure_savp_only="false", sip_invite_domain="hahai1412.anttel-pro.ab-kz-02.antbuddy.com", presence_id="sharklasers(a)hahai1412.anttel-pro.ab-kz-02.antbuddy.com <https://happy.antbuddy.com/sharklasers@hahai1412.anttel-pro.ab-kz-02.antbud…> ",sip_h_X-KAZOO-AOR=sip:sharklasers@hahai1412.anttel-pro.ab-kz- 02.antbuddy.com,absolute_codec_string="^^:PCMU:VP8",leg_timeout="20", effective_callee_id_number="sharklasers",effective_callee_id_name= "sharklasers ",origination_callee_id_number="sharklasers", origination_callee_id_name="sharklasers ",ecallmgr_Realm=" hahai1412.anttel-pro.ab-kz-02.antbuddy.com",ecallmgr_Username= "sharklasers"]sofia/sipinterface_1/sharklasers@113.161.89.159:5914 <https://happy.antbuddy.com/sharklasers@113.161.89.159:5914>;transport=ws ;*fs_path=sip:10.127.15.36:5060*;lr;received="sip:113.161.89.159:5914 ;transport=ws") I research fs_path parameter: https://wiki.freeswitch.org/wiki/Sofia-SIP#Specifying_SIP_Proxy_With_fs_path I guess this is broken It should be Public IP: *fs_path=sip:125.212.212.40:5060*. But I don't know how I can change this parameter ! Anybody faced that problem ? Any solution ? Regards, Hai Bui -- Hai Bui VoIP engineer, Cvoice team, HTK-HCM Office Mobile: +84-165-618-9876

2 1

Kamailio behind NAT
by Safdar Khan 21 Mar '16

21 Mar '16

Hello I have been running kamailio on local network including rtpengine and freeswitch. Now i am configuring kamailio with my public IP(trying with stock kamailio configuration to get started) #define WITH_NAT listen=udp:192.168.3.32:5060 listen=tcp:192.168.3.32:5060 advertised_address= "122.xx.xx.xx" advertised_port="5060" and also using rtpengine module at udp:192.168.3.32:22222 now the problems i am facing are 1.> UA can't register with UDP transport(not even recieving any packets at server ,testing with IMSdroid). 2.>TCP is registering the user, but users can't make calls . i've also noticed this error in /var/log/syslog when tcp ca't make calls ERROR: <core> [tcp_main.c:4250]: tcpconn_main_timeout(): connect 100.68.181.111:50842 failed (timeout) please help me out

2 1

srv_query question
by Vasiliy Ganchev 18 Mar '16

18 Mar '16

Hi folk! I am trying to implement srv_query usage in my script (server is based on Kamailio+Asterisk). The task: when outgoing REGISTER comes from Asterisk (outgoing trunk to operator), and R-URI consists domain name, need to store all ip-addresses of remote side (for following use) Currently, this works using dns_query, and storing in htable structure with remote IPs. But if remote side use SRV, I do not have all IPs Trying to use srv_query function from ipops, I am stuck with following: - srv_query(srvcname, pvid) to get srvcname - for example, "_sip._tcp.example.com", I need to provide, NAPTR record lookup (that is according to RFC3263, and Kamailio does this in core, while request processing) but I see no way how to make NAPTR lookup from script (not involving non Kamailio tools) Looking in mail list archive, found nothing that could help. What I am missing, is there an easy solution, how to cope with my case? Would appreciate any suggestion or idea. Cheers! -- View this message in context: http://sip-router.1086192.n5.nabble.com/srv-query-question-tp146546.html Sent from the Users mailing list archive at Nabble.com.

2 3

PostgreSQL - How To
by Barış Şekerciler 17 Mar '16

17 Mar '16

Hello everyone,I have Kamailio server which runs at 192.168.2.170 address and have PostgreSQL server which runs at 192.168.2.174.So I tried to arrange Kamailio for working with PostgreSQL but I think I give something miss. I couldnt handle it.Do you have any tutorial for this? Or can you tell me how can I exactly do this?Regards.

2 1

xhttp_rpc commands ?
by Uri Shacked 17 Mar '16

17 Mar '16

Hi, Can any one help me with an example for RPC command sent over http? My configuration works OK, I get the http request fine. I lack the knowledge of how to send commands..... For example, how do I send a command to list dialogs (dlg.list)? Or to profile_get_size ? Http://myserver:myport/http_rpc/dlg/????????? Thanks, Uri

4 10

htable - banning users
by meres gr 17 Mar '16

17 Mar '16

I am using the following script to ban users failing to authenticate with fail2ban: My problem is that it logs attempts of users only existing to the database I would like to modify it in order to log all FAILED authentication attempts Is this possible? # Authentication route route[AUTH] { #!ifdef WITH_AUTH if(is_present_hf("Authorization")) { if($sht(a=>$au::auth_count)==3) { $var(exp) = $Ts - 900; if($sht(a=>$au::last_auth) > $var(exp)) { sl_send_reply("403", "Try later"); exit; } else { $sht(a=>$au::auth_count) = 0; } } if(!www_authenticate("$td", "subscriber")) { switch ($retcode) { case -1: sl_send_reply("403", "Forbidden"); exit; case -2: #fixed null -> $null: if($sht(a=>$au::auth_count) == $null) $sht(a=>$au::auth_count) = 0; $sht(a=>$au::auth_count) = $sht(a=>$au::auth_count) + 1; xlog("User $fu Authenticating from ip $si\n"); if($sht(a=>$au::auth_count) == 3) xlog("User $fU Authenticating for the 3d time from ip $si\n"); $sht(a=>$au::last_auth) = $Ts; break; } www_challenge("$td"/*realm*/,"0"/*qop*/); exit; } $sht(a=>$au::auth_count) = 0; } else { www_challenge("$td","0"); exit; } } #!endif

3 2

SIP reg request not print in kamailio.log
by Shiv Patidar 15 Mar '16

15 Mar '16

i send sip reg request to kamailio then i dump tcp thent i got sip request like(INVITE ,TO,FROM ,via header file) but in kamailio.log file i am not got sip request what i need to do please help me my kamailio.log print this log in my terminal /usr/local/sbin/kamailio[25086]:INFO:<core>[main.c:798]:sig_usr():Signal 15 received /usr/local/sbin/kamailio[15085]:INFO:<core>[main.c:798]:sig_usr():Signal 15 received /usr/local/sbin/kamailio[15079]:INFO:<core>[sctp_core.c:53]:sctp_core_destroy():SCTP API not initialized /usr/local/sbin/kamailio[15079]:INFO:<core>[sctp_core.c:75]:sctp_core_check_support():SCTP API not enabled-if you want to use it load sctp module /usr/local/sbin/kamailio[23190]:INFO:rr[../outbound/api.h:54]:ob_load_api():failed to import bind_ob /usr/local/sbin/kamailio[23190]:INFO:rr[rr-mod.c:174]:mod_init():outbound module not available /usr/local/sbin/kamailio[23190]:INFO:usrloc[hslot.c:51]:ul_init_locks():locks aaray size 1024 /usr/local/sbin/kamailio[23190]:INFO:<core>[udp_server.c:150]:probe_max_receive_buffer():SO_RCVBUF is initially 163840 /usr/local/sbin/kamailio[23190]:INFO:<core>[udp_server.c:150]:probe_max_receive_buffer():SO_RCVBUF is finally 327680 /usr/local/sbin/kamailio[23198]:INFO:ctl[io_listener.c:210]:io_listen_loop():io_listen_loop:using epoll_lt as the io watch method (auto detected) i want to print INVITE TO FROM Via header file in my kamailio.log

2 5

Kamailio and UCARP with Multiple VIPs defined
by Jonathan Hunter 15 Mar '16

15 Mar '16

Hi Guys, Using Centos 6.7, kamailio 4.2 and Ucarp I am testing multiple VIPs on the same server and having some issues. I have ucarp running, with 1 Public and 1 Private VIP defined, and both are pingable and failover in a cluster. The Public is for Subscribers to Register with, and the Private for Communication to the Asterisk media servers behind it. My problem is kamailio when started states its listening on both the VIPS, however when I try and force_socket over the Private IP address, it states ERROR: <core> [forward.c:218]: get_out_socket(): no socket foundMar 1 20:04:35 bt1proxy1 /usr/sbin/kamailio[20372]: ERROR: <core> [forward.c:220]: get_out_socket(): no corresponding socket found for(udp:10.70.0.72:5080)Mar 1 20:04:35 bt1proxy1 /usr/sbin/kamailio[20372]: ERROR: tm [t_fwd.c:520]: prepare_new_uac(): ERROR: can't fwd to af 2, proto 1 (no corresponding listening socket) This is when listening on the VIP which is 10.70.0.75 and I also look to force_socket out of this IP. It just doesnt work, however if I use the physical interface on the server, that being 10.70.0.70, it works fine and forwards requests on the Private LAN. I have added net.ipv4.ip_nonlocal_bind = 1 to sysctl.conf as well but no joy. If 1 VIP is defined I have no issue, its just with 2, has anyone had this issue before ? Any help would be great. Many thanks Jon

2 9

RTPPROXY error message
by Kevin Greene 15 Mar '16

15 Mar '16

Hi: I am sure this has been answered before on this list and I have tried the basic answers I got from Google without any luck. Any suggests would be great. The Kamailio host is running Oracle Linux 6.7 (64 bit) and is fully patched as of today. The Kamailio version is: Server:: kamailio (4.3.4 (x86_64/linux)) Build:: mi_core.c compiled on 14:25:16 Nov 25 2015 with gcc 4.4.7 I have two major issues: I am seeing this on my Kamailio log: Mar 14 21:55:33 sipserver /usr/sbin/kamailio[24005]: ERROR: rtpproxy [rtpproxy.c:1681]: send_rtpp_command(): can't send command to a RTP proxy Mar 14 21:55:33 sipserver /usr/sbin/kamailio[24005]: ERROR: rtpproxy [rtpproxy.c:1716]: send_rtpp_command(): proxy <udp:127.0.0.1:7722> does not respond, disable it Mar 14 21:55:33 sipserver /usr/sbin/kamailio[24005]: WARNING: rtpproxy [rtpproxy.c:1573]: rtpp_test(): can't get version of the RTP proxy Mar 14 21:55:33 sipserver /usr/sbin/kamailio[24005]: WARNING: rtpproxy [rtpproxy.c:1610]: rtpp_test(): support for RTP proxy <udp:127.0.0.1:7722> has been disabled temporarily So the initial thought is that the proxy is not running, however I can see the proxy is alive: [root@sipserver boot]# ps -All | grep rtpproxy 5 S 498 2843 1 1 80 0 - 32524 hrtime ? 01:29:06 rtpproxy When I do "netstat -a", I can see 24 established connections. This is a small sample: udp 0 0 localhost:47937 localhost:rtpproxy ESTABLISHED udp 0 0 localhost:38746 localhost:rtpproxy ESTABLISHED udp 0 0 localhost:33660 localhost:rtpproxy ESTABLISHED So I am not sure what is going on. The second problem: I see this in the Kamailio log: Mar 14 22:05:33 sipserver /usr/sbin/kamailio[24000]: ERROR: db_mysql [km_dbase.c:124]: db_mysql_submit_query(): driver error on query: PROCEDURE kamailio.kamailio_rating does not exist (1305) Mar 14 22:05:33 sipserver /usr/sbin/kamailio[24000]: ERROR: <core> [db_query.c:181]: db_do_raw_query(): error while submitting query Mar 14 22:05:33 sipserver /usr/sbin/kamailio[24000]: ERROR: sqlops [sql_api.c:265]: sql_do_query(): cannot do the query [call kamailio_rating('default')] When I tried to create the database originally, the script that was part of the package would not work. I therefore manually created the database so I am thinking that something got left out. I have been over the schema and I am not sure what is missing. The database is MySQL (5.5) running in a separate host. Does anyone have an idea where to look what is missing? I was told that there is some sort of script that can "repair" the Kamailio schema in the database but I have not been able to find this magical script. Thanks. Kevin

2 1

kamailio with external database to replace from uri
by Thibault Gueslin 14 Mar '16

14 Mar '16

I am using kamailio without database. It works fine. For some scenario, I would like to replace the from uri (using mac_replace_from works fine) but using data I can accessed with a REST API. I want to have different origination depending on the sipusername. I need to have some synchronous request. Have someone already these type of behaviour ? Regards Thibault

2 1

Get a random available port to bind to for media sdp
by jaflong jaflong 14 Mar '16

14 Mar '16

Hi everyone How can I generate a random port number, one that is currently available to bind to. I want to assign this to the media parameter in the sdp thanks

2 1

Re: [SR-Users] Registration Failure - IMS
by Franz Edler 11 Mar '16

11 Mar '16

Hi Cristina, > But I have another problem related to registration process. After REGISTER- > 401 Unauthorized-REGISTER-200 OK, the IMS client sends the SUBSCRIBE > message (for the "reg" event package subscription) to the S-CSCF, the latter > replies with NOTIFY and the client correctly responds with 200 OK. In order to > be notified on any change of registration state for the client, also the P-CSCF > sends the SUBSCRIBE message to the S-CSCF, the S-CSCF sends a NOTIFY to > the proxy but, instead of replying with 200 OK, the P-CSCF replies with 404- > Not Here (like it doesn't recognize that the recipient of the NOTIFY in the > Req-URI is the P-CSCF itself). > > Have you ever seen similar issue? I have now spent some time to reproduce the issue. First of all: I had to adapt module reg_mod.c in ims_registrar_pcscf, because it containes a hardcoded P-CSCF address: str pcscf_uri = str_init("sip:pcscf.ims.smilecoms.com:4060"); which is only valid for smilecoms. Did you also change that? Then I got exactly the same problem. I found that the P-CSCF config causes in the part "# Check for Subsequent requests:" a reject " sl_send_reply("404","Not here");" I have doubts that this is correctly designed and have to dig deeper into the logic of the config-file. Fact is: the NOTIFY request does not contain a Route header and as it not an ACK it goes straight to the reject. Maybe any IMS expert can shed some light on this code as shown below: # 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 { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; BR Franz

3 3

UDP send: Operation not permitted
by Sebastian Damm 11 Mar '16

11 Mar '16

Hi, we have a load balancer which is handling a lot of SIP traffic all day. There's always 20-40 Mbit SIP traffic going through. From time to time we see in our logs messages like these: Sep 16 09:46:28 ecker /usr/sbin/kamailio[25505]: ERROR: <core> [udp_server.c:591]: udp_send(): ERROR: udp_send: sendto(sock,0x7f2d9d6b3ce0,1321,0,46.237.225.126:5060,16): Operation not permitted(1) Sep 16 09:46:38 ecker /usr/sbin/kamailio[25194]: ERROR: <core> [udp_server.c:591]: udp_send(): ERROR: udp_send: sendto(sock,0x7efc982b8fc8,420,0,82.113.121.183:35794,16): Operation not permitted(1) Sep 16 09:46:40 ecker /usr/sbin/kamailio[25505]: ERROR: <core> [udp_server.c:591]: udp_send(): ERROR: udp_send: sendto(sock,0x7f2d9d6b3ce0,1338,0,5.158.137.9:55067,16): Operation not permitted(1) Sep 16 09:46:44 ecker /usr/sbin/kamailio[25183]: ERROR: <core> [udp_server.c:591]: udp_send(): ERROR: udp_send: sendto(sock,0x7efc982d9f48,450,0,178.165.131.197:37515,16): Operation not permitted(1) Sep 16 09:46:49 ecker /usr/sbin/kamailio[25643]: ERROR: <core> [udp_server.c:591]: udp_send(): ERROR: udp_send: sendto(sock,0x7f93fb624530,496,0,172.56.7.69:25643,16): Operation not permitted(1) Sep 16 09:46:55 ecker /usr/sbin/kamailio[25335]: ERROR: <core> [udp_server.c:591]: udp_send(): ERROR: udp_send: sendto(sock,0x7f41632cda98,598,0,80.215.234.139:3396,16): Operation not permitted(1) Sep 16 09:46:56 ecker /usr/sbin/kamailio[25345]: ERROR: <core> [udp_server.c:591]: udp_send(): ERROR: udp_send: sendto(sock,0x7f41632f4840,459,0,94.197.120.191:8225,16): Operation not permitted(1) I know that these messages can be produced by iptables blocking the outbound traffic. But our outbound chain looks basically like this: root@ecker:~# iptables-save | grep OUTPUT :OUTPUT DROP [0:0] -A OUTPUT -m state --state RELATED,ESTABLISHED -j ACCEPT -j ACCEPT -A OUTPUT -m state --state INVALID -j DROP -A OUTPUT -o lo -m state --state NEW -j ACCEPT -A OUTPUT -o lo -j ACCEPT -A OUTPUT -m state --state NEW -j ACCEPT We don't have the nf_ct_sip module loaded, syslog doesn't say anything, and even clearing all iptables rules doesn't eliminate those errors. Has anyone ever seen this? It looks like a load thing, because at weekends there are significantly less errors. Thanks, Sebastian

5 7

ACK validation
by jaflong jaflong 11 Mar '16

11 Mar '16

Hi When I receive a ACK how do I check that it belongs to a current dialog. Is there a function that will do this

2 1

Development open again for master branch
by Daniel-Constantin Mierla 11 Mar '16

11 Mar '16

Hello, as branch 4.4 was created to host the upcoming release series 4.4, new features can be now pushed to master branch. They will be part of the next future release, likely to be numbered 5.0. Any fixes that affect existing code in branches 4.4 or older version have to be backported - push first to master and then cherry pick -- some guidelines exemplified with branch 3.2 are available at: - https://www.kamailio.org/wiki/devel/backporting-to-3.2.x As usual, the next future release will be out as stable version in like 8-10 months, with the additional note that 5.0 is planned to include some major refactoring and enhancements. See: - https://www.kamailio.org/wiki/devel/kamailio-5.0-design Many thanks to all contributors and enjoy Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 18-20, 2016 Berlin, Germany - http://www.kamailioworld.com

1 0

Branch 4.4 created
by Daniel-Constantin Mierla 11 Mar '16

11 Mar '16

Hello, the GIT branch 4.4 was created, it will host the release series 4.4.x. To get this branch from GIT, you can use: git clone https://github.com/kamailio/kamailio.git kamailio cd kamailio git checkout -b 4.4 origin/4.4 Notes about installing Kamailio from this branch are available at: - https://www.kamailio.org/wiki/install/4.4.x/git Hopefully in about two weeks or so the full release of 4.4.0 will be out. >From now on, any corresponding fix has to be pushed first to master branch and then cherry-picked to branch 4.4. No new features can get in branch 4.4. Enhancements to documentation or helping tools are still allowed. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 18-20, 2016 Berlin, Germany - http://www.kamailioworld.com

1 0

kamailio print log message
by vinod patidar 11 Mar '16

11 Mar '16

i send SIP reg request to kamailio and i got SIP request in wireshark and also when i dump tcp then i got sip request but in my kamailio.log i am not got SIP request please help me my kamailio.log print this in my terminal /usr/local/sbin/kamailio[25086]:INFO:<core>[main.c:798]:sig_usr():Signal 15 received /usr/local/sbin/kamailio[15085]:INFO:<core>[main.c:798]:sig_usr():Signal 15 received /usr/local/sbin/kamailio[15079]:INFO:<core>[sctp_core.c:53]:sctp_core_destroy():SCTP API not initialized /usr/local/sbin/kamailio[15079]:INFO:<core>[sctp_core.c:75]:sctp_core_check_support():SCTP API not enabled-if you want to use it load sctp module /usr/local/sbin/kamailio[23190]:INFO:rr[../outbound/api.h:54]:ob_load_api():failed to import bind_ob /usr/local/sbin/kamailio[23190]:INFO:rr[rr-mod.c:174]:mod_init():outbound module not available /usr/local/sbin/kamailio[23190]:INFO:usrloc[hslot.c:51]:ul_init_locks():locks aaray size 1024 /usr/local/sbin/kamailio[23190]:INFO:<core>[udp_server.c:150]:probe_max_receive_buffer():SO_RCVBUF is initially 163840 /usr/local/sbin/kamailio[23190]:INFO:<core>[udp_server.c:150]:probe_max_receive_buffer():SO_RCVBUF is finally 327680 /usr/local/sbin/kamailio[23198]:INFO:ctl[io_listener.c:210]:io_listen_loop():io_listen_loop:using epoll_lt as the io watch method (auto detected) i want to print INVITE TO FROM Via header file in my kamailio.log

1 0

SIP Autentication Failed
by Bruno Henrique Gomes Gouvêa 11 Mar '16

11 Mar '16

Hello, First excuse any mistakes in English. I'm working on a platform that makes use of integrated Kamailio with asterisk (Elastix mt). This platform allows the login of my SIP clients via the extension number and the client's domain, where the kamailio after authentication change the username to sip EXTENSION + _ + DOMAIN (expected by Asterisk) and sends to the asterisk. I need the client can also send the sip user in standard EXTENSION + _ + DOMAIN, for it changed the view subscriber to also return a username record in the format that the softphone is sending, but the platform still seems not authenticate the REGISTER using new way functions in auth_check ("$ FD", "subscriber", "0") and auth_challenge ("$ FD", "0"). Thus works: Login: 101 Domain/server: pabx999999.ligou.me Password: PASSWORD_PLAIN This way does not work: Login: 101_pabx999999.ligou.me Domain/server: pabx999999.ligou.me Password: PASSWORD_PLAIN Already got a good look at the documentation and could not find the problem, could give me some help? Kamailio.cfg: http://pasted.co/db8fb3a2 Execution debug log: http://pasted.co/49bfe5bf Contents view subscriber: +-----------------------------+---------------------+---------------+------+ | username | domain | ha1 | ha1b | +-----------------------------+---------------------+---------------+------+ | admin | pabx999999.ligou.me | PASSWORD_PLAIN| NULL | | adminIM | pabx999999.ligou.me | PASSWORD_PLAIN| NULL | | 101 | pabx999999.ligou.me | PASSWORD_PLAIN| NULL | | 103 | pabx999999.ligou.me | PASSWORD_PLAIN| NULL | | 103IM | pabx999999.ligou.me | PASSWORD_PLAIN| NULL | | admin_pabx999999.ligou.me | pabx999999.ligou.me | PASSWORD_PLAIN| NULL | | adminIM_pabx999999.ligou.me | pabx999999.ligou.me | PASSWORD_PLAIN| NULL | | 101_pabx999999.ligou.me | pabx999999.ligou.me | PASSWORD_PLAIN| NULL | | 103_pabx999999.ligou.me | pabx999999.ligou.me | PASSWORD_PLAIN| NULL | | 103IM_pabx999999.ligou.me | pabx999999.ligou.me | PASSWORD_PLAIN| NULL | -------------------------------------------- Bruno H. G. Gouvêa ---

2 14

how to generate kamailio user response
by vinod patidar 10 Mar '16

10 Mar '16

i am install kamailio in my linux pc and start kamailio successfully and i want to print log messages in kamailio.log file please help me and give me any idea what i need to add in my config file please give me brief idea or reference for follow next step

1 2

how to print log message in kamailio.log
by Shiv Patidar 10 Mar '16

10 Mar '16

my kamailio.log file print this but i want to print user registration please help me what i need do /usr/local/sbin/kamailio[25086]:INFO:<core>[main.c:798]:sig_usr():Signal 15 received /usr/local/sbin/kamailio[15085]:INFO:<core>[main.c:798]:sig_usr():Signal 15 received /usr/local/sbin/kamailio[15079]:INFO:<core>[sctp_core.c:53]:sctp_core_destroy():SCTP API not initialized /usr/local/sbin/kamailio[15079]:INFO:<core>[sctp_core.c:75]:sctp_core_check_support():SCTP API not enabled-if you want to use it load sctp module /usr/local/sbin/kamailio[23190]:INFO:rr[../outbound/api.h:54]:ob_load_api():failed to import bind_ob /usr/local/sbin/kamailio[23190]:INFO:rr[rr-mod.c:174]:mod_init():outbound module not available /usr/local/sbin/kamailio[23190]:INFO:usrloc[hslot.c:51]:ul_init_locks():locks aaray size 1024 /usr/local/sbin/kamailio[23190]:INFO:<core>[udp_server.c:150]:probe_max_receive_buffer():SO_RCVBUF is initially 163840 /usr/local/sbin/kamailio[23190]:INFO:<core>[udp_server.c:150]:probe_max_receive_buffer():SO_RCVBUF is finally 327680 /usr/local/sbin/kamailio[23198]:INFO:ctl[io_listener.c:210]:io_listen_loop():io_listen_loop:using epoll_lt as the io watch method (auto detected)

3 3

In kamailio.log not print log message
by Shiv Patidar 09 Mar '16

09 Mar '16

user registration is done successfully but in /var/log/kamailio.log is not print log messages i follow some link but still occur problem i follow this link http://nil.uniza.sk/sip/kamailio/kamailio-logging-how-debian-lenny https://www.kamailio.org/wiki/install/4.0.x/git https://www.kamailio.org/wiki/tutorials/3.2.x/syslog what i am do please help me

2 5

rare impossibility to send requests to TLS connected peer
by Vasiliy Ganchev 09 Mar '16

09 Mar '16

Hi folk! Have a rare issue with tls connections. Have a server with 100+ active tls connections, that are connected to separate TCP proxy, that listen on port 443. TCP proxy send sip traffic to Kamailio via localhost:5061. In this way all connections in Kamailio are considered from Localhost. Issue scheme: - UA-A (sip-phone, UDP/TCP/TLS) -> call -> sip server (Kamailio+Asterisk based) -> UA-B (sip phone, transport=TLS) - The troubles happened in part of flow to UA-B: --- kamailio script make lookup(location), find AOR for UA-B (here is OK) --- after t_relay, kamailio make branch_route logic and try to send INVITE to tcp socket --- HERE IS the trouble: connection seems to be opened, but Kamailio cannot send INVITE there (say: no open tcp connection found) --- Kamailio try to open tcp connection from itself, TCP proxy refuse it --- as a result - Kamailio answer 477 to Asterisk, and finish processing debug level 3 logs (from the point where Kamailio try to send INVITE in opened connection) Jan 25 14:32:43 /usr/sbin/kamailio[4722]: DEBUG: <core> [tcp_main.c:1463]: _tcpconn_find(): tcpconn_find: 0 port 60286 Jan 25 14:32:43 /usr/sbin/kamailio[4722]: DEBUG: <core> [ip_addr.c:243]: print_ip(): tcpconn_find: ip 127.0.0.1 Jan 25 14:32:43 /usr/sbin/kamailio[4722]: DEBUG: <core> [tcp_main.c:1820]: tcp_send(): tcp_send: no open tcp connection found, opening new one Jan 25 14:32:43 /usr/sbin/kamailio[4722]: DEBUG: <core> [ip_addr.c:243]: print_ip(): tcpconn_new: new tcp connection: 127.0.0.1 Jan 25 14:32:43 /usr/sbin/kamailio[4722]: DEBUG: <core> [tcp_main.c:1073]: tcpconn_new(): tcpconn_new: on port 60286, type 3 Jan 25 14:32:43 /usr/sbin/kamailio[4722]: DEBUG: <core> [tcp_main.c:1626]: _tcpconn_add_alias_unsafe(): _tcpconn_add_alias_unsafe: alias port 60286 for hash 2646, id 404319 Jan 25 14:32:43 /usr/sbin/kamailio[4722]: DEBUG: <core> [tcp_main.c:1626]: _tcpconn_add_alias_unsafe(): _tcpconn_add_alias_unsafe: alias port 60286 for hash 2728, id 404319 Jan 25 14:32:43 /usr/sbin/kamailio[4722]: DEBUG: <core> [tcp_main.c:1624]: _tcpconn_add_alias_unsafe(): _tcpconn_add_alias_unsafe: alias already present Jan 25 14:32:43 /usr/sbin/kamailio[4722]: DEBUG: <core> [tcp_main.c:1382]: tcpconn_add(): tcpconn_add: hashes: 2646:2728:0, 863 Jan 25 14:32:43 /usr/sbin/kamailio[4722]: DEBUG: tls [tls_server.c:184]: tls_complete_init(): Using TLS domain TLSc<default> Jan 25 14:32:43 /usr/sbin/kamailio[4722]: DEBUG: tls [tls_domain.c:700]: sr_ssl_ctx_info_callback(): SSL handshake started Jan 25 14:32:43 /usr/sbin/kamailio[4722]: ERROR: <core> [tcp_main.c:2742]: tcpconn_1st_send(): connect 127.0.0.1:60286 failed (RST) Connection refused Jan 25 14:32:43 /usr/sbin/kamailio[4722]: ERROR: <core> [tcp_main.c:2752]: tcpconn_1st_send(): 127.0.0.1:60286: connect & send for 0xb1d3b7e8 failed: Connection refused (111) Jan 25 14:32:43 /usr/sbin/kamailio[4722]: DEBUG: <core> [tcp_main.c:2947]: tcpconn_chld_put(): tcpconn_chld_put: destroying connection 0xb1d3b7e8 (404319, -1) flags 0060 Jan 25 14:32:43 /usr/sbin/kamailio[4722]: ERROR: tm [../../forward.h:247]: msg_send(): tcp_send failed Jan 25 14:32:43 /usr/sbin/kamailio[4722]: DEBUG: tm [t_fwd.c:1591]: t_send_branch(): t_send_branch: send to 127.0.0.1:60286 (3) failed Jan 25 14:32:43 /usr/sbin/kamailio[4722]: WARNING: tm [t_fwd.c:1610]: t_send_branch(): ERROR: t_send_branch: sending request on branch 0 failed Jan 25 14:32:43 /usr/sbin/kamailio[4722]: DEBUG: tm [t_funcs.c:358]: t_relay_to(): ERROR:tm:t_relay_to: t_forward_nonack returned error Jan 25 14:32:43 /usr/sbin/kamailio[4722]: DEBUG: tm [t_funcs.c:375]: t_relay_to(): -477 error reply generation delayed Jan 25 14:32:43 /usr/sbin/kamailio[4722]: DEBUG: tm [t_lookup.c:1526]: t_unref(): t_unref: delayed error reply generation(-477) Jan 25 14:32:43 /usr/sbin/kamailio[4722]: DEBUG: <core> [parser/msg_parser.c:339]: parse_headers(): parse_headers: flags=ffffffffffffffff Jan 25 14:32:43 /usr/sbin/kamailio[4722]: DEBUG: tm [t_reply.c:1670]: cleanup_uac_timers(): DEBUG: cleanup_uac_timers: RETR/FR timers reset Jan 25 14:32:43 /usr/sbin/kamailio[4722]: DEBUG: tm [t_reply.c:728]: _reply_light(): DEBUG: reply sent out. buf=0xb67c000c: SIP/2.0 477 Unfortun..., shmem=0xb1b676cc: SIP/2.0 477 Unfortun Jan 25 14:32:43 /usr/sbin/kamailio[4722]: DEBUG: tm [t_funcs.c:156]: put_on_wait(): DEBUG: put on WAIT Jan 25 14:32:43 /usr/sbin/kamailio[4722]: DEBUG: tm [t_reply.c:738]: _reply_light(): DEBUG: _reply_light: finished Additionally want to say, that suddenly we don't have clear reproduce, it happened only on 2 production servers, what are common conditions for issue: - transport=TLS - when issued call is done, I checked with netstat, the connection is alive - the issued device is re-REGISTERed as usual, in kamctl ul show I see registration before/during/after issued call - issue appear only when connection through our custom TCP proxy :443 (to Kamailio it is from 127.0.0.1:5061) - if connection switched to Kamailio 5061 directly - issue never (till now) reproduced - in one of the issued server - I noticed big number of tcp id. Doing kamcmd core.tcp_list, I saw e.g. id: 567852 (usually id numbers are not more than 1000). - if kamailio/device reboot - problem disappear Because of not completely understanding how tcp core is working in kamailio, ask any suggestion, why such could happen. Any idea/hints would be greatly appreciated! kamailio -v version: kamailio 4.2.5 (i386/linux) flags: STATS: Off, EXTRA_DEBUG, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, DBG_F_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 on 16:03:38 Jan 12 2016 with gcc 4.3.2 sorry for the long description cheers! -- View this message in context: http://sip-router.1086192.n5.nabble.com/rare-impossibility-to-send-requests… Sent from the Users mailing list archive at Nabble.com.

2 5

Branching v4.4
by Daniel-Constantin Mierla 09 Mar '16

09 Mar '16

Hello, I am considering to branch v4.4 latest on March 10, after that the master branch can be used again for adding new features and fixes will be backported to branch 4.4. If it proves to worth branching earlier, it can be done. As an estimate date for the release of v4.4.0, a good candidate would be one of the days between March 22, 23 or 24. Any other suggestions or comments? Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, March 7-9, 2016 - http://www.asipto.com

1 1

Kamailio 5.0 - Scripted DB Migrations
by Tim Chubb 08 Mar '16

08 Mar '16

As a suggestion for 5.0 having a scripted DB migration system would make upgrading much easier. Something like what Asterisk now uses (Alembic https://pypi.python.org/pypi/alembic) would be ideal, as it works in a DB agnostic manner and you would just have to ensure that there is a driver for the DB you wish to use. That way you would just update the definition script and select the target when you run the updater, and would allow kamailio a greater choice in backing DB's as once the script is written assuming a driver is available for you RDBMS the schema would be successfully ported. This would do away with maintaining separate MySQL, Postgres, Oracle, MSSQL scripts as well. The only issue I can think with a system like this is in the situation where hardcoded SQL strings are in the source and use a RDBMS specific function as that could cause some "fun" in debugging...

3 2

question about dlg_var
by 张顺通 08 Mar '16

08 Mar '16

I want to distingguish 200 OK is for INVITE or reINVITE。 so I do this: request_route { if (is_method("INVITE") && !has_totag()) { $dlg_var(is_init_invite) = "1"; } } onreply_route { xlog("L_INFO","[$rs-$rr][is_init_invite:$dlg_var(is_init_invite)]"); if (is_method("INVITE") && t_check_status("200") && $dlg_var(is_init_invite) == "1") { # do somting $dlg_var(is_init_invite) = "0"; } xlog("L_INFO","[$rs-$rr][is_init_invite:$dlg_var(is_init_invite)]"); } What I confuse is： When I recive 100 Trying LOG is [100 - Trying][is_init_invite:0] [100 - Trying][is_init_invite:1] why is_init_invite is 0?

4 5

Kamailio design that needs one b2bua
by anthony gonzalez 08 Mar '16

08 Mar '16

Hello there, I'm designing and implementing a kamailio solution for sip routing between several softswitch based on prefixes. IIn this setup i will need an b2bua in order to do topology hiding and also to make transparent the call transfer between phones registered in kamailio and softswitches(that doesn't support very well REFER messages). I was searching what solution should I use and I found 3 opensource solutions that for me seems to be acceptable: Sems. Freeswitch. Opensips with module b2bua. Based on your experience and contemplating the stability, performance and support, what is the solution that you advice me ? Thank you for you time. Anthony

2 1

is 't_reply_with_body' currently available in kamailio
by jaflong jaflong 07 Mar '16

07 Mar '16

Hi Is the function 't_reply_with_body' currently available in Kamailio. I am using 4.3.5 I get the error0(7176) ERROR: <core> [cfg.y:3295]: yyparse(): cfg. parser: failed to find command t_reply_with_body (params 3) thanks

3 5

Kamailio 5.0 - pseudo variable name cleanup
by Sven Neuhaus 07 Mar '16

07 Mar '16

Hi, a suggestion for Kamailio 5.0 is to get rid of the cryptic, case sensitive pseudo-variable names. Instead we replace them with descriptive names like the ones already provided by the TLS module. The current pseudo variables are a constant nuisance unless you deal with Kamailio every day and know them all by heart. They are an unnecessary barrier for new users learning Kamailio and force you to refer to the documentation more often than necessary. Ideally there would be a script to replace old variable names in configuration files with the new ones to make the transition less painful. -Sven

3 3

How to get Authorization Attributes
by Matthew Harrold 07 Mar '16

07 Mar '16

Hi All, I am trying to get attributes from the Proxy-Authenticate WWW-Authenticate and Authorization headers, namely username, realm, uri, nonce, and response, I can see some of these are already presented as psuedo-variables, however nonce and response are not. What's the most efficient way to obtain these headers? I'm currently using a regex, but there must be a more efficient method. Thanks for your help, Matthew

3 2

How to comunicate two different server voip
by Raihan Satya 06 Mar '16

06 Mar '16

Halo, my name dadang setiawan sample case VOIP Server A IP Address : 192.168.10.57 voip number Jon 100 Voip server B IP Address : 192.168.15.30 voip number Brian 200 i am using linphone video for comunication jon & brian the question is How to solution comunication jon with brian withhout 200(a)192.168.15.30 but direct 200 without using @

2 1

Re: [SR-Users] KAMAILIO Installation Problem
by Ed Todd 05 Mar '16

05 Mar '16

KAMAILIO Installation Problem Fred: re confusion. A few more details. Site 1 is a development site. The project in question is VoIP, using ANTISIP. A Windows server application is monitoring activities and carrying out local functions via IPhones. It has a LAN with static IPs for: · Windows 10 PC used for development · Windows 10 laptop, used for testing · MACBOOK, used for IPhone development · ‘Other PC’, recently converted to Linux + KAMailIO, using the binaries installation route rather than the source code route. It has a Comcast Modem/Router, Cisco DPC3941B Site 2 is a business site, which demonstrates the application using its own Windows PC and IPhones. It does not have any static IPs. Each site uses IPhones as the user interface of this application, via cell or WIFI. Both sites can communicate via SIP.ANTISIP.COM Site 1 can communicate with site 1 KAMAILIO but site 2 cannot: timeouts occur. I just installed what seems like a standard package: · Debian 32 or 64 bit graphical · Apt-get install git gcc make flex bison mysql-server kamailio-mysql-modules kamailio-tls-modules kamailio Should be OK so far apart from the .COM name of the Linux box. I then updated files as described earlier and the problems start. Apart from changing a few constants, I have not altered any script, I am just using the standard one shipped in the package above. I’ve had to guess how to tell KAMailIO about the static IP.

1 0

KAMAILIO Installation Problem
by Ed Todd 05 Mar '16

05 Mar '16

I have a local area network with: · A Windows PC with a VOIP application · A Linux PC with KAMailIO There is also a remote Windows PC with the same VOIP application. There is also an IPhone VOIP application that communicates with the Windows PC application. All communications use the same KAMailIO system. The IPhone application communicates with the PC on the local network. It times out trying to communicate. I have tried a URL and a static IP address for SIP_DOMAIN, neither succeeds. Here are some extracts from the log (ignore timestamp gap, several calls were attempted). I think the problem is that the reply to the PC is being sent to 192.168.1.12 instead of 92.13.147.87. Mar 4 00:28:51 SIS-Server kamailio[18136]: DEBUG: <core> [ip_addr.c:243]: print_ip(): tcpconn_new: new tcp connection: 92.13.147.87 Mar 4 00:28:52 SIS-Server kamailio[18130]: DEBUG: tls [tls_server.c:354]: tls_accept(): TLS accept successful Mar 4 00:28:52 SIS-Server kamailio[18130]: DEBUG: tls [tls_server.c:361]: tls_accept(): tls_accept: new connection from 92.13.147.87:49302 using TLSv1/SSLv3 AES256-SHA 256 Mar 4 00:28:52 SIS-Server kamailio[18130]: DEBUG: tls [tls_server.c:364]: tls_accept(): tls_accept: local socket: 96.81.57.12:5061 Mar 4 00:28:52 SIS-Server kamailio[18130]: DEBUG: auth [api.c:86]: pre_auth(): auth:pre_auth: Credentials with realm '96.81.57.12' not found Mar 4 00:28:55 SIS-Server kamailio[18130]: DEBUG: registrar [reply.c:374]: build_contact(): created Contact HF: Contact: <sip:hvs_uk_pc@192.168.1.12:49302;transport=TLS;line=b3d07204a26e36e>;expires=120 Mar 4 00:37:09 SIS-Server kamailio[18135]: DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 11==9 && [96.81.57.12] == [127.0.0.1] Mar 4 00:37:09 SIS-Server kamailio[18135]: DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 5060 (advertise 0) matches port 5060 Mar 4 00:37:09 SIS-Server kamailio[18135]: DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 11==13 && [96.81.57.12] == [192.168.1.181] Mar 4 00:37:09 SIS-Server kamailio[18135]: DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 5060 (advertise 0) matches port 5060 Mar 4 00:37:09 SIS-Server kamailio[18135]: DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 11==11 && [96.81.57.12] == [96.81.57.12] Mar 4 00:37:09 SIS-Server kamailio[18135]: DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 5060 (advertise 0) matches port 5060 Mar 4 00:37:09 SIS-Server kamailio[18135]: DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 11==9 && [96.81.57.12] == [127.0.0.1] Mar 4 00:37:09 SIS-Server kamailio[18135]: DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 5060 (advertise 0) matches port 5060 Mar 4 00:37:09 SIS-Server kamailio[18135]: DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 11==13 && [96.81.57.12] == [192.168.1.181] Mar 4 00:37:09 SIS-Server kamailio[18135]: DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 5060 (advertise 0) matches port 5060 Mar 4 00:37:09 SIS-Server kamailio[18135]: DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 11==11 && [96.81.57.12] == [96.81.57.12] Mar 4 00:37:09 SIS-Server kamailio[18135]: DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 5060 (advertise 0) matches port 5060Mar 4 00:37:12 SIS-Server kamailio[18136]: ERROR: <core> [tcp_main.c:4159]: handle_tcpconn_ev(): connect 192.168.1.12:49302 failed Changes to KAMCTLRC are: SIP_DOMAIN=ed-todd-randd.com DBENGINE=MYSQL DBNAME=kamailio DBRWUSER="kamailio" DBRWPW="kamailiorw" Changes to KAMAILIO.CFG are: #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_TLS Plus I set debug=4 for logging. Whatsmyip.org shows the right IP address (static) and hostname www.ed-todd-randd.com So, any clues?

2 1

Installation problem: timeouts
by Ed Todd 05 Mar '16

05 Mar '16

I have a local area network with: · A Windows PC with a VOIP application · A Linux PC with KAMailIO There is also a remote Windows PC with the same VOIP application. There is also an IPhone VOIP application that communicates with the Windows PC application. All communications use the same KAMailIO system. The IPhone application communicates with the PC on the local network. It times out trying to communicate. The application works fine with either PC on another server (sip.antisip.com) I have tried a URL and a static IP address for SIP_DOMAIN, neither succeeds. Here are some extracts from the log (ignore timestamp gap, several calls were attempted). I think the problem is that the reply to the PC is being sent to 192.168.1.12 instead of 92.13.147.87. Mar 4 00:28:51 SIS-Server kamailio[18136]: DEBUG: <core> [ip_addr.c:243]: print_ip(): tcpconn_new: new tcp connection: 92.13.147.87 Mar 4 00:28:52 SIS-Server kamailio[18130]: DEBUG: tls [tls_server.c:354]: tls_accept(): TLS accept successful Mar 4 00:28:52 SIS-Server kamailio[18130]: DEBUG: tls [tls_server.c:361]: tls_accept(): tls_accept: new connection from 92.13.147.87:49302 using TLSv1/SSLv3 AES256-SHA 256 Mar 4 00:28:52 SIS-Server kamailio[18130]: DEBUG: tls [tls_server.c:364]: tls_accept(): tls_accept: local socket: 96.81.57.12:5061 Mar 4 00:28:52 SIS-Server kamailio[18130]: DEBUG: auth [api.c:86]: pre_auth(): auth:pre_auth: Credentials with realm '96.81.57.12' not found Mar 4 00:28:55 SIS-Server kamailio[18130]: DEBUG: registrar [reply.c:374]: build_contact(): created Contact HF: Contact: <sip:hvs_uk_pc@192.168.1.12:49302;transport=TLS;line=b3d07204a26e36e>;expires=120 Mar 4 00:37:09 SIS-Server kamailio[18135]: DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 11==9 && [96.81.57.12] == [127.0.0.1] Mar 4 00:37:09 SIS-Server kamailio[18135]: DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 5060 (advertise 0) matches port 5060 Mar 4 00:37:09 SIS-Server kamailio[18135]: DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 11==13 && [96.81.57.12] == [192.168.1.181] Mar 4 00:37:09 SIS-Server kamailio[18135]: DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 5060 (advertise 0) matches port 5060 Mar 4 00:37:09 SIS-Server kamailio[18135]: DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 11==11 && [96.81.57.12] == [96.81.57.12] Mar 4 00:37:09 SIS-Server kamailio[18135]: DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 5060 (advertise 0) matches port 5060 Mar 4 00:37:09 SIS-Server kamailio[18135]: DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 11==9 && [96.81.57.12] == [127.0.0.1] Mar 4 00:37:09 SIS-Server kamailio[18135]: DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 5060 (advertise 0) matches port 5060 Mar 4 00:37:09 SIS-Server kamailio[18135]: DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 11==13 && [96.81.57.12] == [192.168.1.181] Mar 4 00:37:09 SIS-Server kamailio[18135]: DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 5060 (advertise 0) matches port 5060 Mar 4 00:37:09 SIS-Server kamailio[18135]: DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 11==11 && [96.81.57.12] == [96.81.57.12] Mar 4 00:37:09 SIS-Server kamailio[18135]: DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 5060 (advertise 0) matches port 5060Mar 4 00:37:12 SIS-Server kamailio[18136]: ERROR: <core> [tcp_main.c:4159]: handle_tcpconn_ev(): connect 192.168.1.12:49302 failed Changes to KAMCTLRC are: SIP_DOMAIN=ed-todd-randd.com DBENGINE=MYSQL DBNAME=kamailio DBRWUSER="kamailio" DBRWPW="kamailiorw" Changes to KAMAILIO.CFG are: #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_TLS Plus I set debug=4 for logging. Whatsmyip.org shows the right IP address (static) and hostname www.ed-todd-randd.com So, any clues?

1 0

DMQ question
by Alex Balashov 04 Mar '16

04 Mar '16

Hi, If I understand the DMQ documentation correctly, when a SIP request is distributed to other Kamailio instances via dmq_t_replicate(), it is not merely sent as a payload in a novel way, but actually "rehydrated" by the receiver into a mock request on the other end and processed as through it were a "real" message that came into the request_route[] in the natural way. Is this an accurate assumption? -- Alex -- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 9

options_reply() - ruri contains username
by Alex Balashov 04 Mar '16

04 Mar '16

Hello, This may have been discussed before, but when options_reply() from the siputils module is used to respond to OPTIONS requests and those OPTIONS requests come in with an RURI containing a user part, the following error is emitted to the logs: LM_ERR("ruri contains username\n"); No reply for the OPTIONS ping is given in this scenario. OPTIONS pings against a username are very common, and I can find no justification for this check in my interpretation of RFC 3261 scripture. According to Section 11.1 ("Construction of an OPTIONS request"): An OPTIONS request is constructed using the standard rules for a SIP request as discussed in Section 8.1.1. Moreover, the example request line given in that very section is: OPTIONS sip:carol@chicago.com SIP/2.0 Might it perhaps be agreeable to remove this constraint? Or is there some rationale for its presence that I have simply failed to grasp? Thanks! -- Alex -- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 1

Reg: Facing issue in Registration of SIP Client
by sainath.ellendula＠wipro.com 04 Mar '16

04 Mar '16

Hi Franz, On Successful completion of Registration of IMS client. Now, I was trying to REGISTER SIP Client, But REGISTER request failing with 403 forbidden- User unknown. Please find attachment of X-lite configuration and Tcpdump from #422. Could you please suggest me any changes has to be done. Thanks in advance!! Regards, Sainath The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com

3 9

Hello I am having problems pointing to a custom dispatcher table
by Humberto Rodriguez 03 Mar '16

03 Mar '16

Hello I am having problems pointing to a custom dispatcher table, when I use "dispatcher" as table, the kamailio service start running normal, but when I try to use for example "my_dispatcher" table the error shown bellow appears. I do not want to rebuilt the database, *my purpose is to have multiples dispatcher tables on the same server*, "dispatcher, my_dispatcher", "my_dispatcher_1", for example. The connection to the custom database is OK Thanks in advance for your help. ---------------------------------------------------------------------------------------- /var/log/messages Mar 3 14:03:00 kama-balance4 /usr/local/kamailio/sbin/kamailio[19153]: ERROR: dispatcher [dispatch.c:676]: init_ds_db(): invalid table version (found 0 , required 1, 2, 3 or 4)#012(use kamdbctl reinit) Mar 3 14:03:00 kama-balance4 /usr/local/kamailio/sbin/kamailio[19153]: ERROR: dispatcher [dispatcher.c:335]: mod_init(): could not initiate a connect to the database Mar 3 14:03:00 kama-balance4 /usr/local/kamailio/sbin/kamailio[19153]: ERROR: <core> [sr_module.c:970]: init_mod(): init_mod(): Error while initializing module dispatcher (/usr/local/kamailio/lib64/kamailio/modules/dispatcher.so)

2 1

mariadb and sql_query_async: the db driver module doesn't support async query
by Daniel Tryba 03 Mar '16

03 Mar '16

So trying to push some inserts/updates to the background I added: async_workers=1 and replaced sql_xquery("ca", "$var(query)", "ra"); sql_result_free("ra"); with sql_query_async("ca", "$var(query)"); where: $var(query)="insert into CDR.callogs (accountcode,number) values ('"+$var(accountcode)+"',substring('"+$rU+"',1,4))"; Only to find out that the queries aren't inserted and the log contains: /usr/sbin/kamailio[24754]: ERROR: sqlops [sql_api.c:429]: sql_do_query_async(): the db driver module doesn't support async query The table is an InnoDB some insert delayed should give these kind of errors, but isn't the point of sql_query_async that it does the plain normal insert in a background task? Test machine: kamailio 4.2.5

1 1

Kamailio 5.0 - db_cluster module sql_query_async support
by Daniel Tryba 03 Mar '16

03 Mar '16

If the backends for a dbcluster supports sql_query_async, sql_query_async should be available. At this moment (only tested with 4.3.x) async isn't supported.

1 0

Kamailio v4.3.5 Released
by Daniel-Constantin Mierla 03 Mar '16

03 Mar '16

Hello, Kamailio SIP Server v4.3.5 stable release is out. This is a maintenance release of the latest stable branch, 4.3, that includes fixes since release of v4.3.4. There is no change to database schema or configuration language structure that you have to do on previous installations of v4.3.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.3.5. For more details about version 4.3.5 (including links and guidelines to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2016/03/kamailio-v4-3-5-released/ RPM, Debian/Ubuntu packages will be available soon as well. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Kamailio 5.0 - HTTP loading of config files
by Tim Chubb 03 Mar '16

03 Mar '16

As a suggestion for 5.0 being able to load config files from a URL, with any includes being taken relative to the URL would be a really handy feature. It would allow a central source for config files in a cluster and make it a much simpler process to push out changes to routing to all nodes.

6 11

Releasing Kamailio v4.3.5
by Daniel-Constantin Mierla 03 Mar '16

03 Mar '16

Hello, it's been a while since the release of v4.3.4, therefore I am considering to release a new minor version from branch 4.3 next week, most probably on Thursday, March 3, 2016. As usual, if there you are aware of issues not reported to the tracker [1], do it as soon as possible to get a chance to be fixed before the release date. Cheers, Daniel [1] https://github.com/kamailio/kamailio/issues -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, March 7-9, 2016 - http://www.asipto.com

1 2

change the to_tag on send_reply() or sl_send_reply()
by Uri Shacked 02 Mar '16

02 Mar '16

Hi, Is i possible to control the to_tag when i send my own stateless reply from kamailio. Thanks, Uri

3 4

Handling of messages with manipulated Event header.
by Christopher Vella 02 Mar '16

02 Mar '16

Hi, I apologise if this is not the correct venue for such a question... I am running kamailio v4.3.4 (x86_64/linux) on debian Jessie, sourced from; deb http://deb.kamailio.org/kamailio jessie main In my routing logic I am handling SUBSCRIBE requests and striping extra Event package parameters appended to "ua-profile" event types. i.e.; Event: ua-profile;model=somemodel;vendor=somevendor;profile-type=user becomes; Event: ua-profile This is done with this code; } else if( is_method("SUBSCRIBE")) { subst('/ua-profile.*/ua-profile/') ; handle_subscribe(); t_release(); However, in such conditions "search_event()" doesn't correctly find the event. Even though it looks like it is searching for the right event type; start event= [ua-profile/7] handle_subscribe(): unsupported presence event ua-profile --- Begin debug ---- ]3(7664) DEBUG: <core> [re.c:457]: subst_run(): matched (332, 79): [Event: ua-profile;model=somemodel;vendor=somevendor;profile-type=user] with [Event: ua-profile]extops.c:680]: subst_f(): textops: replacing at offset 376[Event: ua-profile;model=somemodel;vendor=somevendor;profile-type=user] 13(7664) debug: textops [textops.c:696]: subst_f(): lst was 0x7fb391fd0da8 13(7664) error: *** cfgtrace:request_route=[presence] c=[/etc/kamailio/kamailio.cfg] l=365 a=24 n=handle_subscribe 13(7664) debug: presence [event_list.c:348]: search_event(): start event= [ua-profile/7] 13(7664) notice: presence [subscribe.c:1092]: handle_subscribe(): unsupported presence event ua-profile 13(7664) debug: tm [t_lookup.c:1011]: t_check_msg(): debug: t_check_msg: msg id=1 global id=1 t start=0x7fb388763290 13(7664) debug: tm [t_lookup.c:1083]: t_check_msg(): debug: t_check_msg: t already found! 13(7664) debug: <core> [mem/shm_mem.c:101]: _shm_resize(): warning:vqm_resize: resize(0) called 13(7664) debug: tm [t_reply.c:1591]: cleanup_uac_timers(): debug: cleanup_uac_timers: retr/fr timers reset 13(7664) debug: <core> [tcp_main.c:2196]: tcpconn_send_put(): send from reader (7664 (13)), reusing fd 13(7664) debug: <core> [tcp_main.c:2430]: tcpconn_do_send(): sending... 13(7664) debug: <core> [tcp_main.c:2464]: tcpconn_do_send(): after real write: c= 0x7fb38875ecf8 n=368 fd=9 13(7664) debug: <core> [tcp_main.c:2465]: tcpconn_do_send(): buf= sip/2.0 489 bad event via: sip/2.0/tcp 127.0.1.1:5060;branch=z9hg4bk-2999-1-0;rport=34679;received=192.168.88.88 cseq: 1 subscribe call-id: 1-2999(a)127.0.1.1 from: <sip:c.vella@albourne.com>;tag=1 to: <sip:provisioning@albourne.com>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-811a --- End debug --- As a test, I send another message with clean "Event: ua-profile" and this works , event= [ua-profile/7] is found. --- Begin debug --- ]4(7665) DEBUG: <core> [re.c:457]: subst_run(): matched (332, 18): [Event: ua-profile ] with [Event: ua-profile]extops.c:680]: subst_f(): textops: replacing at offset 376 [Event: ua-profile 14(7665) DEBUG: textops [textops.c:696]: subst_f(): lst was 0x7fb391fd0da8 14(7665) ERROR: *** cfgtrace:request_route=[PRESENCE] c=[/etc/kamailio/kamailio.cfg] l=365 a=24 n=handle_subscribe 14(7665) DEBUG: presence [event_list.c:348]: search_event(): start event= [ua-profile/7] 14(7665) DEBUG: presence [subscribe.c:1161]: extract_sdialog_info_ex(): 'Expires' header found, value= 3600 14(7665) DEBUG: presence [subscribe.c:1195]: extract_sdialog_info_ex(): 'To' header ALREADY PARSED: <sip:provisioning@albourne.com> 14(7665) DEBUG: presence [subscribe.c:1266]: extract_sdialog_info_ex(): generating to_tag 14(7665) DEBUG: sl [sl.c:329]: get_reply_totag(): totag stateful mode (tm) 14(7665) DEBUG: presence [subscribe.c:1326]: extract_sdialog_info_ex(): subs->contact= sip:c.vella@192.168.88.179:59206;transport=tcp - len = 46 14(7665) DEBUG: presence [subscribe.c:1043]: handle_subscribe(): subscription status= active - inserted 14(7665) DEBUG: presence [subscribe.c:554]: update_subscription(): update subscription 14(7665) DEBUG: presence [notify.c:115]: printf_subs(): pres_uri: sip:c.vella@albourne.com 14(7665) DEBUG: presence [notify.c:116]: printf_subs(): watcher_user@watcher_domain: c.vella(a)albourne.com 14(7665) DEBUG: presence [notify.c:117]: printf_subs(): to_user@to_domain: provisioning(a)albourne.com 14(7665) DEBUG: presence [notify.c:118]: printf_subs(): from_user@from_domain: c.vella(a)albourne.com 14(7665) DEBUG: presence [notify.c:119]: printf_subs(): callid/from_tag/to_tag: 1-3002(a)127.0.1.1/1/a6a1c5f60faecf035a1ae5b6e96e979a-1030 14(7665) DEBUG: presence [notify.c:120]: printf_subs(): local_cseq/remote_cseq: 0/1 14(7665) DEBUG: presence [notify.c:121]: printf_subs(): local_contact/contact: sip:192.168.88.85:5060;transport=tcp/sip:c.vella@192.168.88.179:59206;transport=tcp 14(7665) DEBUG: presence [notify.c:122]: printf_subs(): record_route: 14(7665) DEBUG: presence [notify.c:123]: printf_subs(): sockinfo_str: tcp:192.168.88.85:5060 14(7665) DEBUG: presence [notify.c:125]: printf_subs(): event: ua-profile 14(7665) DEBUG: presence [notify.c:126]: printf_subs(): status: active 14(7665) DEBUG: presence [notify.c:127]: printf_subs(): reason: 14(7665) DEBUG: presence [notify.c:128]: printf_subs(): version: 1 14(7665) DEBUG: presence [notify.c:129]: printf_subs(): expires: 3600 14(7665) DEBUG: presence [notify.c:131]: printf_subs(): updated/updated_winfo: -1/-1 14(7665) DEBUG: presence [subscribe.c:629]: update_subscription(): subscription not in dialog 14(7665) DEBUG: presence [subscribe.c:634]: update_subscription(): inserting in shtable 14(7665) DEBUG: tm [t_lookup.c:1011]: t_check_msg(): DEBUG: t_check_msg: msg id=1 global id=1 T start=0x7fb388763290 14(7665) DEBUG: tm [t_lookup.c:1083]: t_check_msg(): DEBUG: t_check_msg: T already found! 14(7665) DEBUG: <core> [mem/shm_mem.c:101]: _shm_resize(): WARNING:vqm_resize: resize(0) called 14(7665) DEBUG: tm [t_reply.c:1591]: cleanup_uac_timers(): DEBUG: cleanup_uac_timers: RETR/FR timers reset 14(7665) DEBUG: <core> [tcp_main.c:2196]: tcpconn_send_put(): send from reader (7665 (14)), reusing fd 14(7665) DEBUG: <core> [tcp_main.c:2430]: tcpconn_do_send(): sending... 14(7665) DEBUG: <core> [tcp_main.c:2464]: tcpconn_do_send(): after real write: c= 0x7fb38875ecf8 n=399 fd=9 14(7665) DEBUG: <core> [tcp_main.c:2465]: tcpconn_do_send(): buf= SIP/2.0 202 OK Via: SIP/2.0/TCP 127.0.1.1:5060;branch=z9hG4bK-3002-1-0;rport=34680;received=192.168.88.88 CSeq: 1 SUBSCRIBE Call-ID: 1-3002(a)127.0.1.1 From: <sip:c.vella@albourne.com>;tag=1 To: <sip:provisioning@albourne.com>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-1030 Expires: 3600 Contact: <sip:192.168.88.85:5060;transport=tcp> --- End debug --- I suspect I am not understanding this correctly. Is there a proper way around this? I wonder if I am wrong to think that textops Module manipulates the actual request message? Thank you! Regards, -- Chris Vella Albourne Partners (Canada) Limited

2 2

kamailio 4.3.4 crash
by Rene Montilva 02 Mar '16

02 Mar '16

Hi list i've updated to kamailio 4.3.4 , i work with push wake up config, but kamailio now crash and display this messages ar 1 19:38:43 proxy6-2 /usr/sbin/kamailio[21630]: WARNING: tm [t_lookup.c:1476]: t_unref(): WARNING: script writer didn't release transaction Mar 1 19:38:43 proxy6-2 /usr/sbin/kamailio[21630]: WARNING: tm [h_table.c:130]: free_cell_helper(): removed cell 0x7ff059882448 is still linked in hash table (t_lookup.c:1486) Mar 1 19:38:43 proxy6-2 snmpd[2204]: Connection from UDP: [10.107.148.69]:52614->[10.107.148.66] Mar 1 19:38:44 proxy6-2 kernel: [6727397.951325] kamailio[21649]: segfault at 0 ip 00007ff0d92fbfb2 sp 00007fff15f54c90 error 6 in tm.so[7ff0d92a3000+eb000] Mar 1 19:38:44 proxy6-2 /usr/sbin/kamailio[21599]: WARNING: tm [h_table.c:130]: free_cell_helper(): removed cell 0x7ff0598719e8 is still linked in hash table (h_table.c:440) Mar 1 19:38:44 proxy6-2 /usr/sbin/kamailio[21599]: WARNING: tm [h_table.c:130]: free_cell_helper(): removed cell 0x7ff059877eb0 is still linked in hash table (h_table.c:440) i don't know what happend, but i guess is my push configuration, like the message display at "t_lookup.c". i got this in the configure file: route[PUSHJOIN] { if (!is_method("REGISTER")) return; $var(hjoin) = 0; lock("$tU"); $var(hjoin) = $sht(vtp=>join::$tU); $sht(vtp=>join::$tU) = $null; unlock("$tU"); if($var(hjoin)==0) return; $var(id_index) = $(var(hjoin){s.select,0,:}{s.int}); $var(id_label) = $(var(hjoin){s.select,1,:}{s.int}); xdbg("resuming trasaction [$var(id_index):$var(id_label)] $tU ($var(hjoin))\n"); xlog("L_INFO","resuming trasaction [$var(id_index):$var(id_label)] $tU ($var(hjoin))\n"); t_continue("$var(id_index)", "$var(id_label)","LL"); } route[LL] { lookup("location"); route(RELAY); }

2 2

Kamailio as a front-end proxy
by Anton Tonev 02 Mar '16

02 Mar '16

Hello everybody, I am a new user of Kamailio (4.3.1), I am working with it since 1-2 months. The thing that I'm trying to do is to build the following system: same LAN 192.168.0.1 Alice proprietary SIP Server [Public_IP_X] ------------ [Public_IP_Y] Kamailio [172.26.0.1] ---------- [172.26.0.1] with 192.168.0.1 registrar Bob Obviously Kamailio has to translate the local addresses of Alice and Bob, e.g. to use the Nathelper module. The module is doing well its job because the Contact headers are replaced with the Public_IP_X when a REGISTER message is sent by Alice's or Bob's sip phones (I am using Linphone and Zoiper as clients). Once the incoming sip register was treated by Kamailio it is sent to the proprietary SIP Server. The server sends 200 OK to Kamailio and the proxy relays the message to the clients. So the sip registration for me it is OK. But when it comes to initiate a call from Alice to Bob the things are not as I expect it. The initial request INVITE sent from Alice goes to the sip server but then the server instead of sending the INVITE for Bob through Kamailio, it sends the message directly to Bob's device. Does anyone knows how to "tell" to the sip server, using the SIP protocol, that it must use the proxy? The only thing I have in mind is to force Kamailio to replace the contact of Alice and more precisely the host/ip address by the proxy's host/ip address. I tested this idea and the sip server did what I was expecting but for me this is not a proper solution. To do that I used this discussion - http://opensips.org/pipermail/users/2010-October/014873.html Thank you in advance for your attention ! Best regards, Anton

6 7

LCR Commands Kamctl
by Alberto Sagredo 02 Mar '16

02 Mar '16

I have just edited kamctl in order to make lcr commands work. Hope that helps someone. Using Kamailio 4.3.0 Please find kamctl attached also ### LCR management # lcr() *{* *case* $1 *in* show_gws) # merr "command disabled" # exit 1 require_dbengine mecho "lcr gateways" QUERY="select * FROM lcr_gw ORDER BY id; " $DBROCMD "$QUERY" ;; show_routes) # merr "command disabled" # exit 1 require_dbengine mecho "lcr routes" QUERY="select * FROM lcr_rule ORDER BY id;" $DBROCMD "$QUERY" ;; reload) # merr "command disabled" # exit 1 kamcmd lcr.reload ;; dump_gws) # merr "command disabled" # exit 1 kamcmd lcr.dump_gws ;; dump_routes) # merr "command disabled" # exit 1 kamcmd lcr.dump_rules *exit* 1 ;; eval_weights) *shift* $AWK 'BEGIN { if (ARGC < 2) { printf("Usage: lcr eval_weights <list of weights (integers 1-254)>\n"); exit; } iters = 100000; for (i = 1; i < ARGC; i++) { counts[i] = 0; } for (i = 1; i <= iters; i++) { for (j = 1; j < ARGC; j++) { elem[j] = ARGV[j] * rshift(int(2147483647 * rand()), 8); } at = 1; max = elem[at]; for (j = 2; j < ARGC; j++) { if (elem[j] > max) { max = elem[j]; at = j; } } counts[at] = counts[at] + 1; } for (i = 1; i < ARGC; i++) { printf("weight %d probability %.4f\n", ARGV[i], counts[i]/iters); } }' $@ ;; *) usage_lcr *exit* 1 *esac* *}*

3 2

Fwd: [kamailio]: Querie on ratelimit module
by Prashant Desai 02 Mar '16

02 Mar '16

Hi Kamailio, We are using Kamailio V4.2.3 for our project and we are trying to address the performance requirement. We have following queries and need some inputs from Kamailio. *Requirement :* CPU Utilization Limit to 70%, If the limit is crossed Kamailio has to reject the Request. And If it is within the limit Kamailio has to Process the Requests. (i.e. Kamailio has to dynamically set and reset the policy based on CPU Load). *Our Findings* : We have come across ratelimit module in Kamailio, we tried using the same to address the requirement but we see it is not dynamically setting and resetting the policy (once if it starts rejecting then every time it rejects it, only if we restart the kamailo it works) *What we Need*: We need to know 1. Are we using the right module to address the above requirement ? 2. We are only using ratelimit, module, do we have to include any other module along with ratelimit? 3. We are using FEEDBACK algorithm, is that OK ? 4. Setting “ modparam("ratelimit", "pipe", "3:FEEDBACK:70") “ , is this correct ? 5. In ratelimit.c , function static int pipe_push(struct sip_msg * msg, int id) { *case* *PIPE_ALGO_FEEDBACK*: *LM_DBG(**"drop_rate [%d],hash[*pipes[id].counter][%d]\n"**,*drop_rate, hash[*pipes[id].* *counter**])*; ret = (hash[*pipes[id].counter % 100] < *drop_rate) ? -1 : 1; } We are not able to understand the above line ( hash[*pipes[id].counter % 100]). Could you please elaborate. 6. Where are we lagging ? Regards, Prashanth ------------------------------ ============================================================================================================================ Disclaimer: This message and the information contained herein is proprietary and confidential and subject to the Tech Mahindra policy statement, you may review the policy at http://www.techmahindra.com/Disclaimer.html externally http://tim.techmahindra.com/tim/disclaimer.html internally within TechMahindra. ============================================================================================================================

2 2

RFC: Kamailio 5.0
by Daniel-Constantin Mierla 01 Mar '16

01 Mar '16

Hello, at Fosdem and other events this year, while meeting many of Kamailio devs, the topic of what's next for Kamailio came into discussions, with suggestions that after 4.4 it may be time for some major leap to 5.0. I created a page where I collected the main ideas put on table so far - rather unpolished draft, but eventually a good starting point: * https://www.kamailio.org/wiki/devel/kamailio-5.0-design Everyone is invited to add suggestions to the wiki page and/or discuss here on mailing lists, what should be there in 5.0, what is good (to keep) or bad (to remove) with current version, what is missing and it is "must have", ... Not to have a single long tread, for each new idea, start a new email thread with subject prefixed by Kamailio 5.0. I think this impacts users, not only devs, and their contribution is useful as well, so it should be fine to discuss the ideas/concepts on sr-users. When it gets to real implementation discussions, then it can be on sr-dev. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, March 7-9, 2016 - http://www.asipto.com

1 0

catching lost websocket ua
by Juha Heinanen 01 Mar '16

01 Mar '16

I have done some more digging on websocket:closed event route. In DB_ONLY mode, easy deregistration of a websocket UA at connection close would require access in the event route to connection_id of the closed connection. The function that is run at the close has ws_connection_t argument: static void wsconn_run_route(ws_connection_t *wsc) ws_connection_t record has these two fields: int id; /* id and id_hash are identical to the values */ unsigned id_hash; /* for the corresponding TCP/TLS connection */ Is id the same thing as what is stored in connection_id field of location table? If it is, may be its value could be made available to the event route in a pseudo variable. -- Juha

2 8

(no subject)
by Loïc Boissy 01 Mar '16

01 Mar '16

Hello, We have detected a problem on the increment of CSeq in the SIP header from kamailio. In a session SIP subscribe / notify, we notice that there is a notify that has the same CSeq than the last. As an attachment capture wireshard to clarify our remarks. Is that you can check where is the problem? Best regards. Loïc -- Visit Digigram Booth N3534 <http://www.mapyourshow.com/shows/index.cfm?Show_ID=nab16&exhid=1436&booth=N…> at the NAB Show in Las Vegas April 18-21! Check out our showroom click here <http://nab16.mapyourshow.com/7_0/exhibitor_details.cfm?exhid=1436>! Use our Guest Pass code LV6965 <https://registration.experientevent.com/showNAB161/default.aspx?Tracking=WB…> for your free pass www.nabshow.com

3 5

Domain Reload issue
by Errol Samuels 29 Feb '16

29 Feb '16

Hi All, I seem to have an issue when I run the command "kamctl domain reload" where kamailio stops running. Feb 27 07:40:16 il-kam01 kernel: [878538.747898] traps: kamailio[11903] general protection ip:7f19ca760a20 sp:7ffe5a1e8890 error:0 in domain.so[7f19ca754000+16000] Feb 27 07:40:16 il-kam01 /usr/sbin/kamailio[11911]: CRITICAL: <core> [pass_fd.c:293]: receive_fd(): EOF on 16 Feb 27 07:40:16 il-kam01 /usr/sbin/kamailio[11896]: ALERT: <core> [main.c:783]: handle_sigs(): child process 11903 exited by a signal 11 Feb 27 07:40:16 il-kam01 /usr/sbin/kamailio[11896]: ALERT: <core> [main.c:786]: handle_sigs(): core was not generated Feb 27 07:40:16 il-kam01 /usr/sbin/kamailio[11896]: INFO: <core> [main.c:798]: handle_sigs(): terminating due to SIGCHLD Feb 27 07:40:16 il-kam01 /usr/sbin/kamailio[11908]: INFO: <core> [main.c:849]: sig_usr(): signal 15 received Feb 27 07:40:16 il-kam01 /usr/sbin/kamailio[11909]: INFO: <core> [main.c:849]: sig_usr(): signal 15 received Feb 27 07:40:16 il-kam01 /usr/sbin/kamailio[11907]: INFO: <core> [main.c:849]: sig_usr(): signal 15 received Feb 27 07:40:16 il-kam01 /usr/sbin/kamailio[11910]: INFO: <core> [main.c:849]: sig_usr(): signal 15 received Feb 27 07:40:16 il-kam01 /usr/sbin/kamailio[11905]: INFO: <core> [main.c:849]: sig_usr(): signal 15 received Feb 27 07:40:16 il-kam01 /usr/sbin/kamailio[11902]: INFO: <core> [main.c:849]: sig_usr(): signal 15 received Feb 27 07:40:16 il-kam01 /usr/sbin/kamailio[11900]: INFO: <core> [main.c:849]: sig_usr(): signal 15 received Feb 27 07:40:16 il-kam01 /usr/sbin/kamailio[11897]: INFO: <core> [main.c:849]: sig_usr(): signal 15 received Feb 27 07:40:16 il-kam01 /usr/sbin/kamailio[11904]: INFO: <core> [main.c:849]: sig_usr(): signal 15 received Feb 27 07:40:16 il-kam01 /usr/sbin/kamailio[11901]: INFO: <core> [main.c:849]: sig_usr(): signal 15 received Feb 27 07:40:16 il-kam01 /usr/sbin/kamailio[11906]: INFO: <core> [main.c:849]: sig_usr(): signal 15 received Feb 27 07:40:16 il-kam01 /usr/sbin/kamailio[11898]: INFO: <core> [main.c:849]: sig_usr(): signal 15 received Feb 27 07:40:16 il-kam01 /usr/sbin/kamailio[11899]: INFO: <core> [main.c:849]: sig_usr(): signal 15 received Feb 27 07:40:16 il-kam01 /usr/sbin/kamailio[11911]: INFO: <core> [main.c:849]: sig_usr(): signal 15 received Feb 27 07:40:16 il-kam01 kernel: [878538.757055] traps: kamailio[11896] general protection ip:7f19ca760a20 sp:7ffe5a1e9000 error:0 in domain.so[7f19ca754000+16000] root@il-kam01:/usr/lib64/kamailio# kamailio -V version: kamailio 4.2.0 (x86_64/linux) 36e695 flags: STATS: Off, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, DBG_F_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: 36e695 compiled on 19:52:29 Jan 28 2015 with gcc 4.9.2 Any ideas? Thanks in advance.

2 3

Help
by Kevin Farrell-Pelletier 29 Feb '16

29 Feb '16

Hi, I work for a VOIP service provider, and have been tasked with optimizing our infrastructure. We have been providing VOIP services to our clients via Asterisk VMs (PIAF) in an ESXi environment, hosted in a datacenter. We are looking for some kind of SIP Router, which would authenticate clients and route their SIP traffic to the appropriate server. By doing so, we are hoping to further secure our infrastructure and to possibly have only one Public IP (which would resolve to the Private IP of the SIP router). The Asterisk servers serve IVR/RINGGROUPS/OUTBOUNDTRUNKS/INBOUNDROUTES/OUTBOUNDROUTES. The Sip Router would therefore route all SIP traffic between the phones and the Asterisk servers, ad the phones would register to the SIP Router. I have tried many solutions (Kamailio, OpenSER, siproxd, Brekeke), but have not been able to configure these services to work the way we want them to. I am including a chart along with this email to outline what we would like to accomplish. Any suggestions or guides would be immensely appreciated. Thank you all for your time. Kevin Farrell Pelletier - Technicien informatique TI // Réseautique // Téléphonie IP // Programmation IT // Networking // VoIP // Application development 9060, boul Parkway, Anjou, Québec, Canada H1J1N5 Téléphone / Phone : 514.907.2000 Ext.203 F : 1.888.582.4001 - SF/TF : 1.855.907.2001 Web : <http://www.connextek.ca/> www.connextek.ca Pensez ENVIRONNEMENT, c'est important: n'imprimez que si nécessaire Consider the ENVIRONMENT before printing

4 3

What is the meaning of "shmized"
by sgy 28 Feb '16

28 Feb '16

Hi, Sorry, but I just want to know the meaning of "shmized", I can't find it in the dictionary, thank you very much. typedef struct _cfg_group { int num; /*!< number of variables within the group */ cfg_mapping_t *mapping; /*!< describes the mapping betweeen the cfg variable definition and the memory block */ char *vars; /*!< pointer to the memory block where the values are stored -- used only before the config is shmized. */ -- Best Regards, Shengy

4 5

want to go mobile, common issues ?
by Sébastien Brice 27 Feb '16

27 Feb '16

Hi everyone. I have enjoyed talk about realtime of last FOSDEM, talk about K. was good. I want to know how do you solve sip dialogs with mobile? actually I know REGISTER is important to authenticate and this request generally last 1hour until then you have to register again. How to solve that for mobile telephony (and what about idle) ? do cellular phone use this schema ? you have to go UDP ? does TCP comes better to handle mobile telephony? I want to ask about IMS as well :-) I heard many good things about Image Multimedia Systems, but in concrete terms what is it? Does IMS target mobile only ? i heard about Boghe, it has many more parameters regarding DNS SRV and extra options. What's so different from my traditionnal Linphone ? thx you

2 1

ims_qos module support IPv6 address?
by Ayon Pal 26 Feb '16

26 Feb '16

Hi, I have been recently configuring kamailio IMS (v 4.3.4) for IPv6 user registration. I have my PCSCF Gm interface, ICSCF Mw interface and SCSF Mw interface all listening on IPv6 address on port 5060. PCSCF Rx interface is configured with IPv4 address to communicate with PCRF. Similarly ICSCF and SCSCF are also configured with IPv4 address to communicate with HSS (not Fraunhofer HSS) . My Linphone client configured with IPv6 address is sending register message to PCSCF. PCSCF throws the following error on while sending the AAR to PCRF *ubuntu kamailio[22627]: INFO: <script>: REGISTER (sip:001010000000004@TestNetwork (5555:0:0:1:ABCD:0:0:1:9060) to sip:001010000000004@TestNetwork, KFEtBsO-dm)ubuntu kamailio[22627]: ERROR: ims_qos [rx_aar.c:878]: rx_send_aar_register(): Unable to add framed IP AVPubuntu kamailio[22627]: ERROR: ims_qos [rx_aar.c:904]: rx_send_aar_register(): unexpected errorubuntu kamailio[22627]: ERROR: ims_qos [mod.c:1145]: w_rx_aar_register(): Failed to send AARubuntu kamailio[22627]: ERROR: ims_qos [mod.c:1189]: w_rx_aar_register(): Error trying to send AARubuntu kamailio[22627]: WARNING: tm [t_lookup.c:1476]: t_unref(): WARNING: script writer didn't release transaction* On debugging, I found that the ip version is hard coded to AF_INET.( a comment is also written TODO: IPV6 support) in ims_qos/mod.c file. On changing it to AF_INET6 ( I did this 'cos I saw IPv6 based checking on other files like "ims_qos/rx_avp.c") I can see AAR from PCSCF to PCRF, but Framed IPv6 Prefix AVP is incorrect. So, I would like to know if ims_qos module supports IPv6 address ? Regards, Ayon

1 0

Kamailio: Internal call
by Vicente Falco 26 Feb '16

26 Feb '16

I have a kamailo with 2 ethernet cards, the first with ip public and the other for asterisk virtual machines. When I call to a phone number (not intenal) it works, but whe I call to a asterisk extension... it does'nt work. (message: service unavailable 500) I have the kamailio with multidomain. What can I do? -- Un Saludo, Vicente Falcó

2 1

Media server can't return ACK correctly
by Alexandru Covalschi 26 Feb '16

26 Feb '16

Hello everyone. The setup is: Carrier ip is CARRIER_IP Public network Kamailio IP will be PUBLIC_IP Private network Kamailio IP will be KAMAILIO_PRIVATE_IP Private network Freeswitch IP is FREESWITCH_PRIVATE_IP ACK CARRIER_IP -> PUBLIC_IP->FREESWITCH_PRIVATE_IP And freeswitch tries to actually send ACK back to PUBLIC_IP which he can't access. Kamailio trace: http://pastebin.com/raw/1W1sXuUa Freeswitch trace: http://pastebin.com/raw/KkZCwTTJ request_route: http://pastebin.com/raw/Y17pXUGY NATMANAGE route: http://pastebin.com/raw/0BpPDjN0 WITHINDLG route: http://pastebin.com/raw/5LpwSigF I'm seeking help with that - what parameter I need to change/add to solve that? Maybe it's a networking problem - but why then ACK reaches Freeswitch and all other requests flow OK? Thanks in advance, Alex -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/

2 9

[4.2.6] Keep-alive sent by dialog module seems does not refresh the dialog...
by Jack Wang 26 Feb '16

26 Feb '16

Hello, I used the below settings to test the keep-alive function within a dialog: modparam("dialog", "default_timeout", 60) modparam("dialog", "ka_timer", 10) modparam("dialog", "ka_interval", 30) and use dlg_set_property("ka-src") or dlg_set_property("ka-dst") to send OPTIONS to caller or callee. A few questions to ask: (1) What's the difference between ka_timer and ka_interval? I read the http://www.kamailio.org/docs/modules/4.2.x/modules/dialog.html , but still can not understand very clearly. I know the ka_interval means OPTIONS will be sent after 30 seconds when dialog started or last keep-alive has been done. but I don't know how ka_timer works. (2) From the packets captured I can see server sent OPTIONS and the client (caller or callee) sent back 200 OK. According to the document, "The dialog timeout is reset each time a sequential request is processed" However, the dialog was still timeout, it seems that the timeout value was not reset. And I make the clients send the OPTIONS and frequently , the dialog never expires now. So the refresh only works on the keep-alive sent by clients not the server here? Thanks :)

1 0

Binary interface: usrloc replications - restore registrations after node restart
by evoip＠ukr.net 25 Feb '16

25 Feb '16

Hi ! It's really nice option - that now Opensips nodes can replicate "SIP registrations" and "Dialogs" between each other w *"bin_listen" options. http://www.opensips.org/Documentation/Interface-Binary-2-1 * I have a few question related to cluster architecture with fail over: 1). - If one of the "Opensips phone edges" was restarted - can it catch all USRLOC registrations throught binary interface from alive nodes ? - if it can't do restoring of the database through binary interface - maybe you have an idea how to do that ? 2). - it's share dialogs information - if one node is suddenly down, can call dialog be proceeded on other node ? With kind regards, Ewgeny Berladin Voip Engineer

2 1

how to drop 200ok and survive?
by Uri Shacked 25 Feb '16

25 Feb '16

Hi, for some strange reason, ask my regulator.... i need to manipulate certain calls. the scenario goes like this: 1. caller sends invite to kamailio. 2. kamailio transfer the call to asterisk. 3. asterisk send progress and play "hello". 4. asterisk creates a new call (dial) to the same kamailio with destination callee. 5. the callee answers the call. here, i need to block the 200ok. so that the caller does not receive it. i managed to block it with t_suspend(). but, there is no bidirectional media. the 183 progress was sent with sendreceive. it seems the asterisk is waiting for the ACK in order to open both ways for media. i tried to use uac_send_req() but it is being sent with no to tag. and when i try manipulating the uac_req(turi) it does not help because it takes all the string i entered and wraps it with <>. any ideas? BR, Uri

3 5

Credential server
by Marek Moravčík 25 Feb '16

25 Feb '16

Hi guys, I am currently doing research in SIP security. As described in RFC 6072, there can be credential server storing public and private keys (certificates). I have question, if there is any support from Kamailio for this, or if somebody has tried similar things with Kamailio. Thanks! Best regards Marek Moravcik

2 2

Can not start kamailio 4.3 and 4.4. Fork errors with working configuration.
by Yuriy Gorlichenko 25 Feb '16

25 Feb '16

Checked config at another servers instanses. I tried to start 2 workable configs. All starts ends with 3 type of errors error.log error2.log error3.log Debug outputs of 3 times started one config on one same server. I think error outside kamailio because 4.3 and 4.4 also compiled successfully. But at this time I can not understand where is trouble. In first file I see that kamailio can not connect to database but user/pass is correct, and I also see that kamailio can not make fork of process. 2 and 3 process have different troubles but base is one - can not fork.

2 1

ovh trunk
by Sébastien Brice 24 Feb '16

24 Feb '16

Hi everyone I followed this guide http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb and got it working (101, 102 and 103) can call eachother. But now i am trying to figure Asterisk's role out. I am more an ipbx person and i am used to register providers trunk in asterisk/sip.conf file, like this: register => [peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension] doing that i got plenty of OPTIONS request and 200 OK reply between my Kamailio and my provider (and it is a bit noisy) doing that i feel missing kamailio's logic and power to deal with externals trunk provider The thing is i need my authenticated users (101,102,103) be capable dialing my trunk and requesting INVITE for non-local request. What is the best way to achieve that? My DID provider gave me user/passwd/realm. I heard about avp special variables (auth_XXXX_avp and uac) and some snippets config that could help me to go there. Is that efficient to place the routing's logic to Kamailio and how to do that with my ovh trunk? thx you

2 1

Re: [SR-Users] Cancel remove or drop transaction
by Uri Shacked 24 Feb '16

24 Feb '16

On 24 Feb 2016 22:08, "Uri Shacked" <uri.shacked(a)gmail.com> wrote: > Because the t_continue will send the 200 that i want not to send (but not > to drop in on_reply). > The thing is that acording to the situation, i need to decide after a few > seconds, if i drop or allow. > On 24 Feb 2016 21:41, "Uri Shacked" <ushacked(a)gmail.com> wrote: > >> Hi, >> When using t_suspend. Is it possible to cancel, drop or remove >> transaction from memory without using t_continue ? >> Uri >> >

1 0

Cancel remove or drop transaction
by Uri Shacked 24 Feb '16

24 Feb '16

Hi, When using t_suspend. Is it possible to cancel, drop or remove transaction from memory without using t_continue ? Uri

2 1

DNSSRV to resolve to multiple Kamailio servers
by Derek Bolichowski 24 Feb '16

24 Feb '16

Hello, I'm trying to create a DNSSRV record that resolves to multiple Kamailio servers for my clients to authenticate against. Before using Kamailio, we had a DNSSRV record setup that resolved to our multiple Asterisk servers via their WAN Address. I've tried to do the same with Kamailio, but not having much luck. The DNSSRV record is setup as follows, as you can see from the output of a `dig` query: ;_sip._udp.kam.some-domain.com. IN SRV ;; ANSWER SECTION: _sip._udp.kam.some-domain.com. 14399 IN SRV 20 100 5060 kam1.some-domain.com. _sip._udp.kam.some-domain.com. 14399 IN SRV 10 100 5060 kam2.some-domain.com. I am not able to register any devices to this SRV record. However I /CAN/ register directly to kam1.some-domain.com and kam2.some-domain.com Registration attempts to the DNSSRV Record (kam.some-domain.com) do not even appear in /var/log/messages. We have the domain name for kam (SRV Record), kam1 and kam2 setup in kamailio.domain and kamailio.dispatcher. What am I missing here? Thank you for taking the time to read this! Derek Bolichowski

3 2

getting error on kamailio with homer configurations
by Achintha 24 Feb '16

24 Feb '16

hi , we configured homer with kamailio sip capture (server-A) then i try configured another kamailio with siptrace (server-B) mysql database configured with "homer_data" DB i try to call to the second kamailio server using sip phone but didn't get data on homer ui then I find an error on kamailio server configured with homer (server-B) as follows, 5(18128) ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on query: Table 'homer_data.sip_capture_call_20160223' doesn't exist (1146) 5(18128) ERROR: <core> [db_query.c:235]: db_do_insert_cmd(): error while submitting query kindly help me to solve this -- Thanking you Achintha

2 2

Issue with Double Record Route
by Alberto Sagredo 24 Feb '16

24 Feb '16

Im fancing some issues due to kamailio behind nat and thought double record route could help me . But enabling that : modparam("rr", "enable_double_rr", 1) And later using record_route_advertised_address("X.X.X.X:5060"); is giving me this error: w_record_route_advertised_address(): Double attempt to record-route Off course, only a single Record Route is present.. What am i doing wrong? Using Kamailio 4.3.4

2 5

catching lost websocket ua
by Juha Heinanen 23 Feb '16

23 Feb '16

when a ua that has registered over tcp has lost its connection to sip proxy, t_relay() succeeds and branch failure route is executed: Feb 21 05:07:26 lohi /usr/bin/sip-proxy[20528]: INFO: ********** entering branch_route [CONTACT_BRANCH] Feb 21 05:07:26 lohi /usr/bin/sip-proxy[20528]: INFO: ********** activating t_on_branch_failure("contact)" Feb 21 05:07:26 lohi /usr/bin/sip-proxy[20528]: INFO: Routing INVITE <sip:jh-0x16737a0@192.98.102.10:44186;transport=tcp> to contact via <sip:192.98.102.10:51297;transport=tcp> Feb 21 05:07:26 lohi /usr/bin/sip-proxy[20528]: WARNING: tm [t_fwd.c:1543]: t_send_branch(): ERROR: t_send_branch: sending request on branch 0 failed Feb 21 05:07:26 lohi /usr/bin/sip-proxy[20528]: INFO: ********* t_relay success Feb 21 05:07:31 lohi /usr/bin/sip-proxy[20480]: INFO: ********** entering tm:branch-failure:contact but when a ua that has registered over websocket over tcp has lost is connection to sip proxy, t_relay() fails and branch failure route is not executed: Feb 21 05:08:26 lohi /usr/bin/sip-proxy[20523]: INFO: ********** entering branch_route [CONTACT_BRANCH] Feb 21 05:08:26 lohi /usr/bin/sip-proxy[20523]: INFO: ********** activating t_on_branch_failure("contact)" Feb 21 05:08:26 lohi /usr/bin/sip-proxy[20523]: INFO: Routing INVITE <sip:an41pa6p@ggjdptrk9jm0.invalid;transport=ws> to contact via <sip:192.98.102.10:39057;transport=ws> Feb 21 05:08:26 lohi /usr/bin/sip-proxy[20523]: WARNING: <core> [msg_translator.c:2756]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found Feb 21 05:08:26 lohi /usr/bin/sip-proxy[20523]: ERROR: <core> [msg_translator.c:1974]: build_req_buf_from_sip_req(): could not create Via header Feb 21 05:08:26 lohi /usr/bin/sip-proxy[20523]: ERROR: tm [t_fwd.c:462]: prepare_new_uac(): could not build request Feb 21 05:08:26 lohi /usr/bin/sip-proxy[20523]: ERROR: tm [t_fwd.c:1712]: t_forward_nonack(): ERROR: t_forward_nonack: failure to add branches Feb 21 05:08:26 lohi /usr/bin/sip-proxy[20523]: INFO: ********* t_relay failure in the latter case, is there some means to get access to the failing branch in order to be able to unregister it, delete its rtpengine session, etc? why can't kamailio behave the same way no matter if pure tcp connection or one using websocket protocol over tcp connection fails? -- juha

2 2

PUBLISH request receives 404 response, and loops on loopback interface
by Matthew Harrold 23 Feb '16

23 Feb '16

Hi all, I am new to Kamailio but have previous experience with Asterisk, Freeswitch and SIP. I am using a fairly default config file with some sections removed (E.G NAT) to simplify things, and some extra logging so I can see the way packets traverse the configuration. Config here - http://pastebin.com/raw/CCgT0C78 I have noticed that when a UAC sends a PUBLISH request, Kamailio immediately responds '404 Not Found' and more worryingly the logs and a packet capture running on the lo interface indicate it's sending the PUBLISH request to itself in a loop until it hits the Max-Forwards limit. Publish URI is PUBLISH sip:201@kamailio.marrold.co.uk;transport=UDP SIP/2.0 I have two questions, 1) Why is this request being sent a 404, when the UAC is in the USRLOC table? 2) Why is the request looping until Max Forwards is exhausted? Any help appreciated, Marrold

3 5

Re: [SR-Users] Kamailio Cluster
by Arsen Hovhanissian 23 Feb '16

23 Feb '16

Sorry forgot to specify (in-memory) not using the DB, I feel like using the DB for such a task would be such a drag on it

2 1

Installation issues
by Ed Todd 23 Feb '16

23 Feb '16

I tried installing the pre-compiled version of 4.3 under Debian and had all sorts of problems. I found there were 2 sets of paths involved in documentation/procedures plus a reference to a 3rd which caused a complete failure when the directory referred to was missing. I ran into more problems when I tried to access my server from outside my network (static IP). So I thought I’d bite the bullet and install from source. I get to the bit where I am creating the database and even more errors: a.. cannot connect to local mysql server b.. kamdbctl.mysql 112: unary operator expected In one part of the procedure it refers to db_mysql and another to mysql. Is there a simple installation guide around anywhere which will help me get around all these problems? All I want is a Debian with KAMAILIO running on it!

2 1

Re: [SR-Users] Kamailio Cluster
by Arsen Hovhanissian 23 Feb '16

23 Feb '16

Since this thread is open, I wanted to ask, is it possible to replicate dialog data over to multiple nodes as well?

1 0

avp's are not available on default onreply_route ?
by Uri Shacked 22 Feb '16

22 Feb '16

before route[relay] i set $avp(x)="test" on the managed onreply_route i can use $avp(x). on the default onreply_route it is not available. meaning <null>. is that how it suppose to work? it makes it difficult to block a certain 200ok, like i meant to, on the issue - "how to drop 200ok and survive?" thanks, uri

2 2

sip_trace vs uac_reg
by James Cloos 22 Feb '16

22 Feb '16

I've started adding sip_trace to my kamailios, and mostly have it working, but cannot figure out how to capture the outgoing calls from uac_reg. I had to add: reply_route { sip_trace(); } onsend_route { sip_trace(); } to capture those packets. What *_route {} does uac_reg target for its outgoing packets? I read uac_reg.[ch], and see that it uses the t_request method from tm's struct tm_binds. And that the methods are defined via a call to load_tm_api(), and that t_request defaults to the request() function in tm/uac.c. But I don't see from that which route chunk it uses. -JimC -- James Cloos <cloos(a)jhcloos.com> OpenPGP: 0x997A9F17ED7DAEA6

3 3

Upgrade 4.0 to 4.3
by Daniel W. Graham 22 Feb '16

22 Feb '16

I'm looking to upgrade a system from 4.0 to 4.3 I see the various upgrade docs https://www.kamailio.org/wiki/install/upgrade/4.0.x-to-4.1.0 https://www.kamailio.org/wiki/install/upgrade/4.1.x-to-4.2.0 https://www.kamailio.org/wiki/install/upgrade/4.2.x-to-4.3.0 If the updates are completed as shown in the docs, can I jump directly to 4.3? Anything to look out for? Thanks! -dan

3 4

Unable to use json module
by Sunil More 22 Feb '16

22 Feb '16

Hello All, I am getting the below error , when ever i try to use json.so module for kamailio. I have compiled and installed with json.so however this error occurs. 0(20153) ERROR: <core> [sr_module.c:576]: load_module(): could not open module </usr/local/kamailio//lib64/kamailio/modules/json.so>: /usr/local/kamailio//lib64/kamailio/modules/json.so: undefined symbol: is_error 0(20153) : <core> [cfg.y:3365]: yyerror_at(): parse error in config file /usr/local/kamailio/etc/kamailio/cc_kamailio/basic.cfg, line 69, column 12-20: failed to load module ERROR: bad config file (1 errors) The kamailio version I am using is : version: kamailio 4.4.0-pre0 (x86_64/linux) 4e7573 Please let me know, how can i proceed here. Thanking You, Sunil More Ph : 9503338275

2 3

db_cluster.so module and kamctlrc DBHOST= field
by Derek Bolichowski 22 Feb '16

22 Feb '16

Hi there, New user to Kamailio here. We currently have it up and running in a virtualized environment with 1 Kamailio sever, 1 Asterisk server and 1 MySQL server. I'm currently writing install scripts to make deploying new nodes/servers easy and to keep settings the same across the board. I've chosed to load the db_cluster.so module in kamailio.cfg, as we will have 2x MySQL servers in master-master replication which will contain the 'kamailio' and 'asterisk' tables. I've just hit a stumbling block - in `kamctlrc`, there is a field called `DBHOST=`. How can I reference my cluster here? In kamailio.cfg, I simply define DBURL as "cluster//<clustername>". What is the syntax for 'DBHOST=' in 'kamctlrc'? Can I reference the cluster? Can I have 2 separate DBHOST= lines? Looking for some guidance on this one. Thanks, Derek B.

2 2

Unable to start kamailio
by sgy 21 Feb '16

21 Feb '16

Hi, Today I download the source codes in version 4.3 by git, and compile and install it. I configure the kamctlrc and kamailio.cfg, but I can't start it, Please refer to following error. Feb 21 21:47:36 ubuntu /usr/local/sbin/kamailio[13672]: ERROR: db_mysql [km_dbase.c:124]: db_mysql_submit_query(): driver error on query: Table 'kamailio.version' doesn't exist (1146) Feb 21 21:47:36 ubuntu /usr/local/sbin/kamailio[13672]: ERROR: <core> [db_query.c:132]: db_do_query_internal(): error while submitting query Feb 21 21:47:36 ubuntu /usr/local/sbin/kamailio[13672]: ERROR: <core> [db.c:397]: db_table_version(): error in db_query Feb 21 21:47:36 ubuntu /usr/local/sbin/kamailio[13672]: ERROR: <core> [db.c:436]: db_check_table_version(): querying version for table location Feb 21 21:47:36 ubuntu /usr/local/sbin/kamailio[13672]: ERROR: usrloc [dlist.c:642]: register_udomain(): error during table version check. Feb 21 21:47:36 ubuntu /usr/local/sbin/kamailio[13672]: ERROR: registrar [reg_mod.c:561]: domain_fixup(): failed to register domain Feb 21 21:47:36 ubuntu /usr/local/sbin/kamailio[13672]: ERROR: <core> [route.c:1154]: fix_actions(): fixing failed (code=-1) at cfg:/usr/local/etc/kamailio//kamailio.cfg:647 Feb 21 21:47:36 ubuntu /usr/local/sbin/kamailio[13672]: ERROR: <core> [route.c:1154]: fix_actions(): fixing failed (code=-1) at cfg:/usr/local/etc/kamailio//kamailio.cfg:649 Feb 21 21:47:36 ubuntu /usr/local/sbin/kamailio[13672]: INFO: <core> [sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized -- Best Regards, Shengy

2 2

Re: [SR-Users] [SPAM]Re: Registration Failure - IMS
by Franz Edler 21 Feb '16

21 Feb '16

Hi Cristina, > But I have another problem related to registration process. After REGISTER- > 401 Unauthorized-REGISTER-200 OK, the IMS client sends the SUBSCRIBE > message (for the "reg" event package subscription) to the S-CSCF, the latter > replies with NOTIFY and the client correctly responds with 200 OK. In order to > be notified on any change of registration state for the client, also the P-CSCF > sends the SUBSCRIBE message to the S-CSCF, the S-CSCF sends a NOTIFY to > the proxy but, instead of replying with 200 OK, the P-CSCF replies with 404- > Not Here (like it doesn't recognize that the recipient of the NOTIFY in the > Req-URI is the P-CSCF itself). > > Have you ever seen similar issue? I have now spent some time to reproduce the issue. First of all: I had to adapt module reg_mod.c in ims_registrar_pcscf, because it containes a hardcoded P-CSCF address: str pcscf_uri = str_init("sip:pcscf.ims.smilecoms.com:4060"); which is only valid for smilecoms. Did you also change that? Then I got exactly the same problem. I found that the P-CSCF config causes in the part "# Check for Subsequent requests:" a reject " sl_send_reply("404","Not here");" I have doubts that this is correctly designed and have to dig deeper into the logic of the config-file. Fact is: the NOTIFY request does not contain a Route header and as it not an ACK it goes straight to the reject. Maybe any IMS expert can shed some light on this code as shown below: # 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 { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; BR Franz

1 0

Make SIP session between two kamailio servers
by Cockhootec Yahrabee 21 Feb '16

21 Feb '16

Hi everyone, I am curious if there is a way to make SIP session between two kamailio server instances eg. if there is way to use kamailio as an SIP client and SIP server at the same time. I am thinking about the following configuration: kamailio main server ----- kamailio server B ----- SIP client B1 | | | kamailio server A ----- SIP client A2 | | | SIP client A1 I want to be able to make SIP session between: SIP client A1 : SIP client A2 kamailio server A : kamailio server B SIP client A1 : SIP client B1 (this would be great) If there is currently no way to do this do you think that it would be possible to create the module which will support this kind of functionality? Thanks in advance for the replies

2 4

How long will be client online in kamailio server since client is forcely closed
by sgy 21 Feb '16

21 Feb '16

Hi All, If the client is forcely closed, the client will be shown as online state in kamailio when I run the command: "kamctl online". How long will be the client online since client is closed? Thank you. -- Best Regards, Shengy

2 2

Re: [SR-Users] [SPAM]Re: Registration Failure - IMS
by Franz Edler 20 Feb '16

20 Feb '16

Hello Cristina, > In the attach you can find the trace of the registration. I have checked your tracefile and confirm: that is a problem which needs further debugging. Up to now I did not activate "WITH_REGINFO" in my installation, therefore I cannot compare. If I find time (this weekend) I will do that and come back with the findings. BR Franz

1 0

rfc: kamailio.cfg polishing for v4.4
by Daniel-Constantin Mierla 20 Feb '16

20 Feb '16

Hello, wondering if people feel it is need to polish the current default kamailio.cfg, to prepare the version for next major release. It can be browsed online at: - https://github.com/kamailio/kamailio/blob/master/etc/kamailio.cfg It is mainly about very common scenarios that are encountered out there and we should deal with by default. Have in mind that there are the basic and advanced versions of kamailio.cfg in the etc folder. Also, I am looking for tunings for core parameters and module parameters that make the config more suitable for common use case. Speaking of those extra config files, I think the kamailio-advanced.cfg and kamailio-basic.cfg should be deployed in shared doc folder, not being active configs upon default installation. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com http://miconda.eu

3 3

BUG: qm_free: freeing already freed pointer
by Rene Montilva 19 Feb '16

19 Feb '16

Hi list i've got this bug report in kamailio 4.3 at syslog, then it frozen, i checked my config file looking for htable parameter, and i notice i have htable param without autexpire is this reason why kamailio crash? "/usr/sbin/kamailio[24113]: : <core> [mem/q_malloc.c:453]: qm_free(): BUG: qm_free: freeing already freed pointer (0x7f673dead5d8), called from tm: h_table.c: free_cell(133), first free tm: h_table.c: free_cell(133) - aborting " -- Ing. Rene Montilva *FOSS Developer and VoIP Engineer.*

2 5

Re: [SR-Users] Error or Bug: [t_msgbuilder.c:490]: build_local_reparse(): ERROR: build_local_reparse: HDR_EOH_T was not found
by Sid Mason 19 Feb '16

19 Feb '16

Yes, there is a append_hf("Remote-Party-ID: below is the complete section: # Send to Asterisk route[TOASTERISK] { ds_mark_dst("P"); if(!ds_select_dst("1", "4")) { sl_send_reply("500", "Service Unavailable"); xlog("L_INFO","[$fU@$si:$sp]{$rm} No destinations available for $rd \n"); exit; } # Replace FROM header to Kamailio's address # http://www.kamailio.org/docs/modules/4.3.x/modules/uac.html#uac.f.uac_repla… uac_replace_from("","sip:$sel(cfg_get.kamailio.bind_external_domain):5060"); # rewrite Destinatin URI #$du= $au + "@10.1.4.22:5060"; # Add Remote-Party-ID to header append_hf("Remote-Party-ID: \"$au\" <sip:$au@ $sel(cfg_get.kamailio.bindip_external)>;party=calling;privacy=off;screen=no"); xlog("L_INFO","route[TOASTERISK]: [$fU@$si:$sp]{$rm} From Outside World to Asterisk Box $du\n"); rtpproxy_manage("cawei"); route(RELAY); exit; } Regards Subject: Re: [SR-Users] Error or Bug: [t_msgbuilder.c:490]: build_local_reparse(): ERROR: build_local_reparse: HDR_EOH_T was not found Message-ID: <56B79AFD.8010505(a)gmail.com> Content-Type: text/plain; charset="windows-1252" Hello, do you have an action like: append_hf("Remote-Party-ID: ..."); in kamailio.cfg? Can you paste all that action here? Cheers, Daniel

2 2

Registration Failure
by Cristina Caridi 19 Feb '16

19 Feb '16

Hello, I'm a new starter of Kamailio (SER). I am using the version 4.4.0-dev6 in a Debian machine. I've configured all the IMS network elements, the FHoSS and also Boghe IMS/RCS client in a Windows machine. When an IMS user belonging to my network tries to register and IP SEC mechanism is enabled, the registration flow stops at 401 Unauthorized from S-CSCF -> I-CSCF -> P-CSCF -> client. The problem is that the HSS in MAA message does not send the SIP-Auth-Data-item that should be cointain the Confidentiality Key and the Integrity Key; so the client is not able to challenge the S-CSCF sending another REGISTER message. Do you have any suggestions about this? When IP SEC is disabled (by managing client options), the registration works well. Maybe is there any configuration at HSS level that i'm not considering? Thank you in advance for your help, Cristina

4 7

ACC Module
by Alberto Sagredo 19 Feb '16

19 Feb '16

I have a strange behaviour when configuring ACC module to put CDRs It must be as simple as this: modparam("acc", "cdr_enable", 1) But got this error: feb 19 09:26:14 kamailio /usr/local/sbin/kamailio[23712]: ERROR: acc [acc_mod.c:604]: mod_init(): failed to init cdr generation feb 19 09:26:14 kamailio /usr/local/sbin/kamailio[23712]: ERROR: <core> [sr_module.c:962]: init_mod(): Error while initializing module acc (/usr/local/lib64/kamailio/modules/acc.so) I have read documentation and seems ok. Tables in databases are created and ALTER commands applied. IM getting actually ACC registers on acc table, but could not active cdrs for acc. What am i missing?

1 2

dispatcher flag
by alexandre.anconetti＠orange.com 19 Feb '16

19 Feb '16

Hi, I'm using the dispatcher module with Kamailio 4.3.4, the keep alive mechanism and the routing are OK, but I would like to get in my request_route the flag(s) (active, inactive, probing ...) that the dispatcher give to each destination. I know that in command line I can run kamcmd or kamctl to get : NRSETS: 1 RECORDS: { SET: { ID: 1 TARGETS: { DEST: { URI: sip: xxx.xxx.xxx.xxx:xxxx FLAGS: IP PRIORITY: 0 } DEST: { URI: sip:xxx.xxx.xxx.xxx:xxxx FLAGS: AP PRIORITY: 0 How can I get that in the request_route process ? Any suggestion would be appreciated ;) Best Regards, Alexandre. _________________________________________________________________________________________________________________________ Ce message et ses pieces jointes peuvent contenir des informations confidentielles ou privilegiees et ne doivent donc pas etre diffuses, exploites ou copies sans autorisation. Si vous avez recu ce message par erreur, veuillez le signaler a l'expediteur et le detruire ainsi que les pieces jointes. Les messages electroniques etant susceptibles d'alteration, Orange decline toute responsabilite si ce message a ete altere, deforme ou falsifie. Merci. This message and its attachments may contain confidential or privileged information that may be protected by law; they should not be distributed, used or copied without authorisation. If you have received this email in error, please notify the sender and delete this message and its attachments. As emails may be altered, Orange is not liable for messages that have been modified, changed or falsified. Thank you.

2 3

LCR Module issue
by Alberto Sagredo 18 Feb '16

18 Feb '16

Hi I have LCR working but not able to add more gateways? It seems only first one is accepted What could be wrong? I show you database data and kamcmd commands output. mysql> select * from lcr_rule_target; +----+--------+---------+-------+----------+--------+ | id | lcr_id | rule_id | gw_id | priority | weight | +----+--------+---------+-------+----------+--------+ | 1 | 1 | 1 | 1 | 1 | 1 | | 2 | 1 | 2 | 1 | 1 | 1 | | 3 | 2 | 2 | 2 | 1 | 1 | +----+--------+---------+-------+----------+--------+ 3 rows in set (0.00 sec) mysql> select * from lcr_rule; +----+--------+-----------+----------+-------------+---------+---------+ | id | lcr_id | prefix | from_uri | request_uri | stopper | enabled | +----+--------+-----------+----------+-------------+---------+---------+ | 1 | 1 | 6 | NULL | NULL | 1 | 1 | | 2 | 1 | DIDNUMBER | NULL | NULL | 1 | 1 | | 3 | 2 | DIDNUMBER | NULL | NULL | 1 | 1 | +----+--------+-----------+----------+-------------+---------+---------+ 3 rows in set (0.00 sec) mysql> select * from lcr_gw; +----+--------+----------------+--------------+----------+------+--------+------------+-----------+-------+--------+------+-------+---------+ | id | lcr_id | gw_name | ip_addr | hostname | port | params | uri_scheme | transport | strip | prefix | tag | flags | defunct | +----+--------+----------------+--------------+----------+------+--------+------------+-----------+-------+--------+------+-------+---------+ | 1 | 1 | Carrier_Prueba | 172.x.x.x | NULL | 5060 | NULL | 1 | 1 | 0 | NULL | NULL | 1 | NULL | | 5 | 2 | Proveedor_1 | 172.x.x.x | NULL | 5060 | NULL | 1 | 1 | 0 | NULL | NULL | 0 | NULL | +----+--------+----------------+--------------+----------+------+--------+------------+-----------+-------+--------+------+-------+---------+ 2 rows in set (0.00 sec) mysql> root@kamailio:~# kamcmd lcr.reload root@kamailio:~# kamcmd lcr.dump_rules { lcr_id: 1 rule_id: 2 prefix: DIDNUMBER from_uri: request_uri: stopper: 1 } { gw_index: 1 priority: 1 weight: 1 } { lcr_id: 1 rule_id: 1 prefix: 6 from_uri: request_uri: stopper: 1 } { gw_index: 1 priority: 1 weight: 1 } 9 1 root@kamailio:~# kamcmd lcr.dump_gws { lcr_id: 1 gw_id: 1 gw_index: 1 gw_name: Carrier_Prueba scheme: sip: ip_addr: 172.x.x.x hostname: port: 5060 params: transport: ;transport=udp strip: 0 prefix: tag: flags: 1 state: 0 defunct_until: 0

2 2

Questions
by francisco rodriguez 18 Feb '16

18 Feb '16

Hi; Today I'm trying to use my users and include unregistered, there is a problem, this was solved soon or do. I appreciate your attention and cooperation Francisco Rodriguez

2 1

Re: [SR-Users] Kamailio IMS deployment
by sainath.ellendula＠wipro.com 17 Feb '16

17 Feb '16

Hi Franz, Thanks for your response. Even we observed in Wireshark traces, no username is included in REGISTER request. Any changes required? No, we are not using NATURAL in our platform. Observed there is no communication between PCSCF – ICSCF while registration and also attaching the tracefile(tcpdump) using command “tcpdump -i any -s 3000 -w /var/tmp/trace_OpenIMS.pcap not port 22 and not port 3306”. Please suggest me if I have missed anything. Regards, Sainath From: Franz Edler [mailto:franz.edler@technikum-wien.at] Sent: Tuesday, February 9, 2016 1:01 PM To: Sainath Ellendula (NEP) <sainath.ellendula(a)wipro.com> Cc: basitstar(a)hotmail.com; sr-users(a)lists.sip-router.org Subject: AW: RE: Re:[SR-Users] Kamailio IMS deployment The REGISTER request is OK. There is no username included in REGISTER request. Regarding the mismatch: do you use NATURAL on your platform? Next step should be a tracefile (tcpdump) and later a debug-log from the node which shows the problem. Von Samsung-Tablet gesendet -------- Ursprüngliche Nachricht -------- Von: sainath.ellendula(a)wipro.com<mailto:sainath.ellendula@wipro.com> Datum: 2016.02.09 08:05 (GMT+01:00) An: franz.edler(a)technikum-wien.at<mailto:franz.edler@technikum-wien.at> Cc: basitstar(a)hotmail.com<mailto:basitstar@hotmail.com>, sr-users(a)lists.sip-router.org<mailto:sr-users@lists.sip-router.org> Betreff: RE: Re:[SR-Users] Kamailio IMS deployment Hi Franz, Please find ping status and seems good. root@server:~# ping -c 1 hss.net1.test PING hss.net1.test (10.0.0.9) 56(84) bytes of data. 64 bytes from server.net1.test (10.0.0.9): icmp_seq=1 ttl=64 time=0.017 ms --- hss.net1.test ping statistics --- 1 packets transmitted, 1 received, 0% packet loss, time 0ms rtt min/avg/max/mdev = 0.017/0.017/0.017/0.000 ms root@server:~# ping -c 1 pcscf.net1.test PING pcscf.net1.test (10.0.0.10) 56(84) bytes of data. 64 bytes from server.net1.test (10.0.0.10): icmp_seq=1 ttl=64 time=0.020 ms --- pcscf.net1.test ping statistics --- 1 packets transmitted, 1 received, 0% packet loss, time 0ms rtt min/avg/max/mdev = 0.020/0.020/0.020/0.000 ms root@server:~# ping -c 1 icscf.net1.test PING icscf.net1.test (10.0.0.11) 56(84) bytes of data. 64 bytes from server.net1.test (10.0.0.11): icmp_seq=1 ttl=64 time=0.018 ms --- icscf.net1.test ping statistics --- 1 packets transmitted, 1 received, 0% packet loss, time 0ms rtt min/avg/max/mdev = 0.018/0.018/0.018/0.000 ms root@server:~# ping -c 1 scscf.net1.test PING scscf.net1.test (10.0.0.12) 56(84) bytes of data. 64 bytes from server.net1.test (10.0.0.12): icmp_seq=1 ttl=64 time=0.017 ms --- scscf.net1.test ping statistics --- 1 packets transmitted, 1 received, 0% packet loss, time 0ms rtt min/avg/max/mdev = 0.017/0.017/0.017/0.000 ms root@server:~# Seems Some progress happened: Tried register using Boghe IMS client and observed few errors: 0(1445) INFO: <script>: [REGISTER] from [sip:alice@net1.test] to [sip:alice@net1.test] 0(1445) ERROR: <script>: REGISTER (sip:alice@net1.test (10.0.0.8:55280) to sip:alice@net1.test, 2006cdca-0076-bacb-5010-bffa714b3cdb) 0(1445) DEBUG: <core> [parser/msg_parser.c:153]: get_hdr_field(): get_hdr_field: cseq <CSeq>: <7994> <REGISTER> 0(1445) DEBUG: <core> [parser/msg_parser.c:187]: get_hdr_field(): DEBUG: get_hdr_body : content_length=0 0(1445) DEBUG: maxfwd [mf_funcs.c:85]: is_maxfwd_present(): value = 70 0(1445) DEBUG: <core> [parser/msg_parser.c:89]: get_hdr_field(): found end of header 0(1445) DEBUG: sanity [mod_sanity.c:248]: w_sanity_check(): sanity checks result: 1 0(1445) DEBUG: siputils [checks.c:97]: has_totag(): no totag 0(1445) DEBUG: <core> [select.c:412]: run_select(): Calling SELECT 0x7fb548dd2400 0(1445) DEBUG: tm [t_lookup.c:1011]: t_check_msg(): DEBUG: t_check_msg: msg id=2 global id=1 T start=0xffffffffffffffff 0(1445) DEBUG: tm [t_lookup.c:466]: t_lookup_request(): t_lookup_request: start searching: hash=65064, isACK=0 0(1445) DEBUG: tm [t_lookup.c:424]: matching_3261(): DEBUG: RFC3261 transaction matching failed 0(1445) DEBUG: tm [t_lookup.c:648]: t_lookup_request(): DEBUG: t_lookup_request: no transaction found 0(1445) DEBUG: tm [t_lookup.c:1080]: t_check_msg(): DEBUG: t_check_msg: msg id=2 global id=2 T end=(nil) 0(1445) DEBUG: sl [sl.c:280]: send_reply(): reply in stateless mode (sl) 0(1445) DEBUG: siputils [checks.c:97]: has_totag(): no totag 0(1445) DEBUG: tm [t_lookup.c:1011]: t_check_msg(): DEBUG: t_check_msg: msg id=2 global id=2 T start=(nil) 0(1445) DEBUG: tm [t_lookup.c:1086]: t_check_msg(): DEBUG: t_check_msg: T previously sought and not found 0(1445) DEBUG: ims_registrar_pcscf [save.c:273]: save_pending(): contact requesting to expire in 600000 seconds 0(1445) DEBUG: ims_usrloc_pcscf [usrloc.c:106]: get_alias_host_from_contact(): no alias param 0(1445) DEBUG: ims_usrloc_pcscf [usrloc.c:155]: get_hash_slot(): using host for hash [10.0.0.8] 0(1445) DEBUG: ims_usrloc_pcscf [usrloc.c:159]: get_hash_slot(): Returning hash slot: [167] 0(1445) DEBUG: ims_usrloc_pcscf [udomain.c:465]: get_pcontact(): Searching for contact in P-CSCF usrloc [sip:alice@10.0.0.8:55280;transport=udp] 0(1445) DEBUG: ims_usrloc_pcscf [usrloc.c:106]: get_alias_host_from_contact(): no alias param 0(1445) DEBUG: ims_usrloc_pcscf [usrloc.c:183]: get_aor_hash(): using host for hash [10.0.0.8] 0(1445) DEBUG: ims_usrloc_pcscf [usrloc.c:187]: get_aor_hash(): Returning hash slot: [1662767783] 0(1445) DEBUG: ims_usrloc_pcscf [udomain.c:483]: get_pcontact(): Comparing needle user@host:port [alice@10.0.0.8:55280] and contact_user@contact_host:port [alice@10.0.0.8:51219] 0(1445) DEBUG: ims_usrloc_pcscf [udomain.c:494]: get_pcontact(): Searching for [sip:alice@10.0.0.8:55280;transport=udp] and comparing to [sip:alice@10.0.0.8:51219;transport=udp] 0(1445) DEBUG: ims_usrloc_pcscf [udomain.c:553]: get_pcontact(): Port don't match: 51219 (contact) 51219 (received) != 55280! 0(1445) DEBUG: ims_registrar_pcscf [save.c:310]: save_pending(): Adding pending pcontact: <sip:alice@10.0.0.8:55280;transport=udp> 0(1445) DEBUG: ims_usrloc_pcscf [pcontact.c:132]: new_pcontact(): New contact [<sip:alice@10.0.0.8:55280;transport=udp>] with 0 associated IMPUs in state: [registration pending] 0(1445) DEBUG: ims_registrar_pcscf [save.c:314]: save_pending(): registering for UL callback 0(1445) DEBUG: ims_usrloc_pcscf [usrloc.c:106]: get_alias_host_from_contact(): no alias param Please guide me for any more changes has to be done, As observed in Wireshark(Register request from 10.0.0.8(host)-PCSCF as REGISTER sip:net1.test SIP/2.0). I think the register request should sent as REGISTER sip:alice@net1.test SIP/2.0 Thanks in advance!! Regards, Sainath From: Franz Edler [mailto:franz.edler@technikum-wien.at] Sent: Tuesday, February 9, 2016 3:00 AM To: Sainath Ellendula (NEP) <sainath.ellendula(a)wipro.com<mailto:sainath.ellendula@wipro.com>> Cc: basitstar(a)hotmail.com<mailto:basitstar@hotmail.com>; sr-users(a)lists.sip-router.org<mailto:sr-users@lists.sip-router.org> Subject: RE: Re:[SR-Users] Kamailio IMS deployment Sainath, what was the result for ping on - icsct.net1.test - scscf.net1.test - hss.net1.test to increase debug level you have to modify the corresponding cfg-file, e.g. kamailio-icscf.cfg in line 35: debug=2 Just increase to debug=3 and then start the process again. BR Franz The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com<http://www.wipro.com> The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com

3 8

[TOPOH] Contact header for 302 Moved Temporarily
by SamyGo 17 Feb '16

17 Feb '16

Hi All, I've recently stumbled upon this little hitch while using kamailio with topoh module that the Contact header do not contains the User part for the 302 Moved temporarily packet. *My topology:* UserA<====\ ->Kamailio<===>FreeSwitch UserB<====/ The B party has set Call forwarding on their phone hence phone sends a 302 Moved Temporarily to Kamailio. Via: SIP/2.0/UDP TOPOH.KAMAILIO.IP:5060;branch=z9hG4bK0767.4e4e57deca89da5c5f5ae97228325f85.0 Via: SIP/2.0/UDP TOPOH.KAMAILIO.IP;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.y7MGZLMGZwM.1LgRWIC9gIgx4fMGZAOBVAOBNfzuaVHRaYpB1LNSQLpx4uMx3Az6eL3RsBCxu-zRrUWSeOgjeBk.IVm4ds34aONc** From: "+4319714111" <sip:+4319714111@FREESWITCH.IP.HERE>;tag=yUr5UZ7eF794K To: <sip:502@USER.B.IP.HERE:5060>;tag=105223296 Call-ID: cd811276-4b4d-1234-66ae-005056867dbc CSeq: 87257355 INVITE Contact: <sip:06606017597@FREESWITCH.IP.HERE:5060> User-Agent: Yealink SIP-T46G 28.80.0.70 Diversion: <sip:502@USER.B.IP.HERE:5060>;reason=unconditional Content-Length: 0 This is modified in Kamailio TOPOH and sent to FreeSwitch as following SIP/2.0 302 Moved Temporarily Via: SIP/2.0/UDP 10.0.20.71;received=10.0.20.71;rport=5060;branch=z9hG4bKeBcy9HKK9cDKr From: "+4319714111" <sip:+4319714111@FREESWITCH.IP.HERE>;tag=yUr5UZ7eF794K To: <sip:502@USER.B.IP.HERE:5060>;tag=105223296 Call-ID: cd811276-4b4d-1234-66ae-005056867dbc CSeq: 87257355 INVITE Contact: <*sip:TOPOH.KAMAILIO.IP* ;line=sr-N6IAzBy6WBy6MxFwW.qwPSW5ohWINhaYNLu4g9W4OhWI3wKLgRsIpUg5kGs7g9P-W.y6My**> User-Agent: Yealink SIP-T46G 28.80.0.70 Diversion: <sip:502@USER.B.IP.HERE:5060>;reason=unconditional Content-Length: 0 Which results in a Call Originate from FreeSwitch with RURI as this: INVITE *sip:TOPOH.KAMAILIO.IP*;line=sr-N6IAzBy6WBy6MxFwW.qwPSW5ohWINhaYNLu4g9W4OhWI3wKLgRsIpUg5kGs7g9P-W.y6My** SIP/2.0 I am thinking that in TOPOH Module some patch is required to atleast retain the $rU for 3XX replies , not sure if this will break some RFC or Kamailio stability etc ! Thanks, Sammy.

3 4

siptrace module plain sip mode
by Grant Bagdasarian 17 Feb '16

17 Feb '16

Hello, The documentation of the siptrace module specifies duplication of sip messages to a capture server using the HEP protocol or plain SIP mode. Which params to I need to set (and with which values) in the siptrace and sipcapture to enable plain SIP mode? I tried several configurations but none of the worked. Seems like only HEP works. My test kamailio (sip capture node with sipcapture module) doesn't receive the duplicated SIP messages. Any ideas? Regards, Grant

2 2

Installatioin issues
by Ed Todd 17 Feb '16

17 Feb '16

I am installing KAMAILIO under DEBIAN. I have got it to work fine (TLS) until I upgraded my Comcast to business service to get static IP’s. Now it no longer works. I am using the Cisco DPC3941 modem/router and am tring to connect an IPhone on WIFI to the PC SIP server on the LAN. Here are the log entries: Feb 16 19:14:59 SIP-Server kamailio[2010]: NOTICE: <core> [main.c:738]: handle_sigs(): Thank you for flying kamailio!!! Feb 16 19:14:59 SIP-Server kamailio[2019]: INFO: <core> [main.c:849]: sig_usr(): signal 15 received Feb 16 20:01:25 SIP-Server kamailio: INFO: tls [tls_init.c:385]: init_tls_compression(): tls: init_tls: disabling compression... Feb 16 20:01:26 SIP-Server kamailio: INFO: <core> [tcp_main.c:4745]: init_tcp(): using epoll_lt as the io watch method (auto detected) Feb 16 20:01:26 SIP-Server kamailio[3876]: INFO: rr [../outbound/api.h:54]: ob_load_api(): Failed to import bind_ob Feb 16 20:01:26 SIP-Server kamailio[3876]: INFO: rr [rr_mod.c:160]: mod_init(): outbound module not available Feb 16 20:01:26 SIP-Server kamailio[3876]: INFO: usrloc [hslot.c:53]: ul_init_locks(): locks array size 1024 Feb 16 20:01:26 SIP-Server kamailio[3876]: ERROR: ctl [init_socks.c:126]: init_unix_sock(): ERROR: init_unix_sock: bind: No such file or directory [2] Feb 16 20:01:26 SIP-Server kamailio[3876]: ERROR: ctl [ctl.c:283]: mod_init(): ERROR: ctl: mod_init: init ctrl. sockets failed Feb 16 20:01:26 SIP-Server kamailio[3876]: ERROR: <core> [sr_module.c:968]: init_mod(): Error while initializing module ctl (/usr/lib/x86_64-linux-gnu/kamailio/modules/ctl.so) I am informed that SIP-ALG is disabled. Any ideas?

2 1

Remove_hf_is not working
by aman jeff 17 Feb '16

17 Feb '16

Hi , I am working kamailio 4.4.0 ,i want to remove my Contact header completely.So I used remove_hf..but its now working ..Let me know the solution Am very thankful to you Regards Jefff

2 1

db_cluster.so module and kamctlrc DBHOST= field
by Hosted Services 16 Feb '16

16 Feb '16

Hi there, New user to Kamailio here. We currently have it up and running in a virtualized environment with 1 Kamailio sever, 1 Asterisk server and 1 MySQL server. I'm currently writing install scripts to make deploying new nodes/servers easy and to keep settings the same across the board. I've chosed to load the db_cluster.so module in kamailio.cfg, as we will have 2x MySQL servers in master-master replication which will contain the 'kamailio' and 'asterisk' tables. I've just hit a stumbling block - in `kamctlrc`, there is a field called `DBHOST=`. How can I reference my cluster here? In kamailio.cfg, I simply define DBURL as "cluster//<clustername>". What is the syntax for 'DBHOST=' in 'kamctlrc'? Can I reference the cluster? Can I have 2 separate DBHOST= lines? Looking for some guidance on this one. Thanks, Derek B.

2 1

Database Insert
by Daniel W. Graham 16 Feb '16

16 Feb '16

I see there is a new SQLops function sql_query_async that has been added in 4.2 (I believe?). Is there a good way to perform database insert using version 4.0? Using mysql Thanks! -dan

2 4

Need to validate a IP address with subnet mask
by Sunil More 16 Feb '16

16 Feb '16

Hi All, I want to validate a IP address in kamailio for requests coming from different IPs. I have browsed and found permissions module, however is there a way where i can directly compare a source IP address to an address stored in an avp and get a true or false value. something like this $si==192.168.2.0/24 Thanking You, Sunil More

3 7

Textops and Multipart Body - adding ISUP
by Sergey Basov 16 Feb '16

16 Feb '16

Hi All. I need to use SIP-I with my upstream to set A number as unknown. I was read about adding incapsulation ISUP using if(has_body("application/sdp")) { set_body_multipart(); msg_apply_changes() $var(acm) = "7e Od 04 55 75 69 20 4d 61 6b 65 43 61 6c 6c"; append_body_part("$var(acm)","application/isup;version=itu-t92+","signal; handling=optional"); msg_apply_changes() xlog("L_INFO", "ISUP Changes Applied Succesfully"); } It is converts body to multipart and inserts ISUP But it inserts is as text. Is there any possibility to insert basic ISUP message to update it after using sipt_destination($rU, 31, 4); and sipt_set_calling($fU, 4, 0, 3); Thank you. -- Best regards, Sergey Basov e-mail: sergey.v.basov(a)gmail.com

3 10

kamailio 4 with asterisk
by Sébastien Brice 15 Feb '16

15 Feb '16

this one (written by Daniel) http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb maybe a bit outdated but still consistent ? the thing i am really stuck with (and concerning real-time) is that none of my extensions (from asterisk CLI) are online: ns3325046*CLI> sip show peers Name/username Host Dyn Forcerport Comedia ACL Port Status Description Realtime 102/102 (Unspecified) D Auto (No) No 0 Unmonitored Cached RT 103/103 (Unspecified) D Auto (No) No 0 Unmonitored Cached RT 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 2 offline] that drives asterisk crazy ! and logger reports: app_dial.c:2411 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) every time i place a call. The tutorial written by daniel mention a channel configuration pretty minimal: INSERT INTO sipusers (name, defaultuser, host, sippasswd, fromuser, fromdomain, mailbox) VALUES ('102', '102', 'dynamic', '102', '102', 'yoursip.com', '102'); and since there's no context associated to the 102 extension i cant figure out where that channel enter the dialplan ? [public] [LocalSet] [default] ???? and a dialplan exten => _1XX,1,Dial(SIP/${EXTEN}) exten => _1XX,n,Voicemail(${EXTEN},u) exten => _1XX,n,Hangup exten => _1XX,101,Voicemail(${EXTEN},b) exten => _1XX,102,Hangup i am sorry to bother you with issues more asterisk oriented than kamailio. By the way i took a good start with kamailio as it seems to work flawlessly on my system. thx you. On Mon, Feb 15, 2016 at 12:26:06PM +0100, Sébastien Brice wrote: > Hi Everyone, i like the way this tutorial explain asterisk and kamailio integration. Which tutorial? > the only thing i missed is asterisk behaviors'r regarding sip registration ? That was a part of a tutorial I once saw. In essence asterisk uses the kamailio database, UA registers on kamailio and is stored there, asterisk sees the same data (realtime). Sébastien BRICE VoIP, Support et Intégration

2 1

no pkg memory left
by Gholamreza Sabery 15 Feb '16

15 Feb '16

Hello. I am seeing the following error repeating again and again in the log file: /sbin/kamailio[14510]: ERROR: registrar [reply.c:199]: build_contact(): no pkg memory left I am using 64MB of SHM memory and 8MB of PKG memory. What is the reason for this error? How to solve it?

2 2

kamailio 4 with asterisk
by Sébastien Brice 15 Feb '16

15 Feb '16

Hi Everyone, i like the way this tutorial explain asterisk and kamailio integration. the only thing i missed is asterisk behaviors'r regarding sip registration ? I tryed to place a call between 102 and 103 extensions and experimenting an issue Asterisk tells me that the subscriber is absent and I'm sent directly to voicemail ! -- Executing [103@public:1] Dial("SIP/102-00000001", "SIP/103") in new stack [Feb 14 21:00:15] WARNING[19444][C-00000001]: app_dial.c:2411 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) ns3325046*CLI> sip show peers Name/username Host Dyn Forcerport Comedia ACL Port Status Description Realtime 102/102 (Unspecified) D Auto (No) No 0 Unmonitored Cached RT 103/103 (Unspecified) D Auto (No) No 0 Unmonitored Cached RT 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 2 offline] apart that my sip.conf and extensions.conf are very minimal: [LocalSets] exten => _1XX,1,Dial(SIP/${EXTEN}) exten => _1XX,n,Voicemail(${EXTEN},u) exten => _1XX,n,Hangup exten => _1XX,102,Voicemail(${EXTEN},b) exten => _1XX,103,Hangup [general] context=LocalSets ; Default context for incoming calls. Defaults to 'default' rtcachefriends=yes ; Cache realtime friends by adding them to the internal list i did INSERT TO the users in mysql tables (sipusers, sipregs and voicemail) and registering extensions from UA works ok (i am using jitsi) Whats wrong with my setup ? thank you

2 1

kamctl add VS kamctl ul add under SIPML
by kssyrr 15 Feb '16

15 Feb '16

2 3

uac module configuration
by Andrew Tan 15 Feb '16

15 Feb '16

Hello, Submit a ticket in GitHub apologize about that I should have sent my question here. I would like some help in configuring the uac module to route to an gateway without specifically setting the URI-domain from the UAC. I would like to set change it in the Kamailio side. Currently I use uac_auth on failure route. All is working if I set the URI from on the client to xxx(a)gatewayaddress.com. But if I just dial just the number as is which translate to xxx(a)kamailioserver.com the invite doesnt go anywhere. My failure route is like so failure_route[MANAGE_FAILURE] { xlog("DEBUG: MANAGE FAILURE ROUTE: "); if ($T_reply_code == "407") # there's also 401 { $avp(arealm) = GATEWAY; $avp(auser) = USERNAME; $avp(apass) = PASSWORD; #$ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); #rewritehost("sip.ca.didlogic.net"); uac_auth(); rtpengine_offer("force trust-address replace-origin replace-session-connection ICE=remove RTP/AVP"); route(NATMANAGE); route(RELAY); exit; } } Thank you

2 1

LCR
by Alexey V. Panfilov 15 Feb '16

15 Feb '16

Hello, I'm using LCR module in my kamailio setup. I can access to the gw_uri_avp's value of matched gateway, but I'd like to get value of selected id of lcr_rule_target. Is there any way to get it? Thanks. P.S. Access to value of custom table's field (for example, named as 'flags') of cr_rule_target will be better. I'd like to store at this field the price of selected direction. -- Best regards, Alexey V. Panfilov

2 4

Kamailio 4 && Asterisk. Getting started
by Sébastien Brice 14 Feb '16

14 Feb '16

Hi everyone I did follow the tutorial Kamailio 4.0.x and asterisk written by Daniel-C and I need a hand :-) after a successful install (kamailio and asterisk on the same server), i did setup 2 sip extensions (102 and 103) and tryed to place a call between them and experimenting an issue Asterisk tells me that the subscriber is absent and I'm sent directly to voicemail ! -- Executing [103@public:1] Dial("SIP/101-00000001", "SIP/103") in new stack [Feb 14 21:00:15] WARNING[19444][C-00000001]: app_dial.c:2411 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) ns3325046*CLI> sip show peers Name/username Host Dyn Forcerport Comedia ACL Port Status Description Realtime 102/102 (Unspecified) D Auto (No) No 0 Unmonitored Cached RT 103/103 (Unspecified) D Auto (No) No 0 Unmonitored Cached RT 2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 2 offline] Actually none of my extensions are online and i am wondering if Kamailio forward register with asterisk ? transport between K and Asterisk seems ok (at least i did setup bindport in both sip.conf and kamailio.cfg) apart that my sip.conf and extensions.conf are very minimal: exten => _1XX,1,Dial(SIP/${EXTEN}) exten => _1XX,n,Voicemail(${EXTEN},u) exten => _1XX,n,Hangup exten => _1XX,102,Voicemail(${EXTEN},b) exten => _1XX,103,Hangup [general] context=LocalSets ; Default context for incoming calls. Defaults to 'default' rtcachefriends=yes ; Cache realtime friends by adding them to the internal list I am a bit new with Kamailio and I don't know if this behavior is normal since i have a sipregs mysql table that could do the job on purpose? Could someone point me to the right direction and enlight my knowing, thanks you thx you

1 0

Kamailio and freeswitch integration for SBC
by malik sherif 12 Feb '16

12 Feb '16

Hello, I was able to have successful call with kamailio and after integrating kamailio with freeswitch using the following link, the invite timeout and as a result the call failed.The status on kamailio show these info. Do I need to download outbound model? I am running kamailio ver 4.1.1 and freeswitch 1.4.18. kamailio[19153]: 0(19162) INFO: rr [../outbound/api.h:54]: ob_load_api(): Failed to import bind_ob kamailio[19153]: 0(19162) INFO: rr [rr_mod.c:159]: mod_init(): outbound module not available http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc Thank you very much and your help is appreciated. Thanks Abdul <http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc> [http://kb.asipto.com/_media/wiki:logo.png]<http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc> freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc [Asipto ...<http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc> kb.asipto.com The scope of this tutorial is to show how you can use Kamailio (former OpenSER) and FreeSWITCH to build a complete SIP/VoIP platform for large number of subscribers.

4 45

BLF on different IPs/domains
by Phil Lavin 12 Feb '16

12 Feb '16

Hi all, A different BLF query to my other one so creating another thread. We've tried to have a BLF client SUBSCRIBE using a secondary IP on Kamailio. It is not sent a NOTIFY, but a client which has done a SUBSCRIBE using the primary IP is. If we route the problematic client such that it connects to the primary IP, it too is sent a NOTIFY. Looking at the query that is done on the database when a call is made to 441164969988 it is as follows: select `callid`,`to_tag`,`from_tag` from `active_watchers` where `presentity_uri`='sip:441164969988@185.28.212.118' AND `event`='dialog' Where 185.28.212.118 is the primary IP. In reality, it should be doing an additional query for anything considered `myself`. i.e. select `callid`,`to_tag`,`from_tag` from `active_watchers` where `presentity_uri`='sip:441164969988@10.20.11.118' AND `event`='dialog' I've been scouring the docs for a way to add additional IPs and/or domains for the purpose of querying active_watchers but I can't see anything relevant. What's the process here? Thanks Phil

2 3

kamailio xmpp gateway configuration issue
by kevin hifx 12 Feb '16

12 Feb '16

Dear friends, I am facing an issue, when trying to configure xmpp gateway between kamailio and ejabbred xmpp server. I have followed the steps mention in this web site: http://www.kamailio.org/docs/modules/4.1.x/modules/xmpp.html The kamailio service is running but in the log file it show that xmpp connection failed. "*Feb 5 11:32:33 sip-xmpp-server /usr/sbin/kamailio[646]: ERROR: xmpp [network.c:122]: net_connect(): connect() failed: Connection refused* " (Both kamailio and ejabbred installed on the same server in local network and have local DNS server) I have added following entries in the kamailio configuration file. I am only mentioning the xmpp section. " ####### Global Parameters ######### alias=sip-xmpp.kevin.org ####### Modules Section ######## #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "mi_rpc.so" loadmodule "acc.so" # XMPP loadmodule "xmpp.so" # ----------------- setting module-specific parameters --------------- modparam("mi_fifo", "fifo_name", "/var/run/kamailio/kamailio_fifo") modparam("rr", "enable_full_lr", 1) # XMPP modparam("xmpp", "domain_separator", "*") modparam("xmpp", "gateway_domain", "sip-xmpp.kevin.org") modparam("xmpp", "xmpp_domain", "xmpp-sip.kevin.org") modparam("xmpp", "xmpp_host", "xmpp.kevin.org") ###############XMPP # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; ### absorb retransmissions ### if (!t_newtran()) { sl_reply_error(); return; } if (method == "MESSAGE") { log("*** xmpp-handled MESSAGE message.\n"); if (xmpp_send_message()) { t_reply("200", "Accepted"); } else { t_reply("404", "Not found"); } return; } log("*** xmpp: unhandled message type\n"); t_reply("503", "Service unavailable"); return; ############### " could you please help me solve this issue. Thank you kevin

2 5

Only Cassandra, Possible?
by Barış Şekerciler 12 Feb '16

12 Feb '16

Hello, I want to use only Cassandra DB for Kamailio. Is this possible? If it yes, how can I do that? For example, if I replace the DBENGINE as DB_CASSANDRA can I use it like this? Thank you. Regards.

2 3

Kamailio TLS
by Kethzer Docteur 12 Feb '16

12 Feb '16

Im trying to enable Kamailio TLS but the configuration is not running at all when I cahnge it liike this as follow. Kamailio 3.3 ubuntu 12.04 LTS #!define WITH_TLS #!ifdef WITH_TLS enable_tls=yes #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #!endif Please help -- Kethzer Docteur

4 3

Negative return codes from functions
by Phil Lavin 11 Feb '16

11 Feb '16

Hi all, Just a sanity check, really. Does Kamailio consider negative response codes to be false? For example, should the following log execute? if (!foo()) { # Returns -2 x_log("Foo is false"); } The reason for asking here is that I'm implementing flood protection using pike, based off the kamailio.cfg that ships with v4.3. The logic is thus: if (!pike_check_req()) { # Do blocking } However pike_check_req only returns -1 or -2 in the case of failure, never 0. The blocking code is, thus, never executed. Changing to explicitly check for != 1 works correctly. Am I missing something here or is the example kamailio.cfg incorrect? Cheers

2 3

Kamailio 4.2.6 crash
by Igor Potjevlesch 11 Feb '16

11 Feb '16

Hello, I have seen couple of crashes since yesterday. I copy/paste the last "bt full": Program terminated with signal 11, Segmentation fault. #0 0x0000000000619694 in fm_extract_free (qm=0x7fcdf8780000, frag=0x7fcdf8a62c78) at mem/f_malloc.c:206 206 *pf= (gdb) bt full #0 0x0000000000619694 in fm_extract_free (qm=0x7fcdf8780000, frag=0x7fcdf8a62c78) at mem/f_malloc.c:206 pf = 0x6576203b50555349 hash = 2097 #1 0x000000000061ad68 in fm_malloc (qm=0x7fcdf8780000, size=1104, file=0x7fce0f081a7b "tm: t_msgbuilder.c", func=0x7fce0f0860a0 "build_local_reparse", line=369) at mem/f_malloc.c:490 f = 0x7fcdf8780b08 frag = 0x7fcdf8a62c78 hash = 156 __FUNCTION__ = "fm_malloc" #2 0x00007fce0f011fa9 in _shm_malloc (size=1099, file=0x7fce0f081a7b "tm: t_msgbuilder.c", function=0x7fce0f0860a0 "build_local_reparse", line=369) at ../../mem/shm_mem.h:208 p = 0x6 #3 0x00007fce0f01920e in build_local_reparse (Trans=0x7fcdf8a5e2f0, branch=0, len=0x7fff00c32fbc, method=0x7fce0f087a25 "ACK", method_len=3, to=0x7fff00c32d90, reason=0x0) at t_msgbuilder.c:369 invite_buf = 0x7fcdf8a62868 "INVITE sip:<CALLED>@goren SIP/2.0\r\nRecord-Route: <sip:A.B.C5.60;lr;did=5b1.3112>\r\nCSeq: 1 INVITE\r\nCall-ID: 3da8-469-1111201513435-JANGO-1-A.B.C_leg2\r\nFrom: <sip:<CALLING>@<HOST_FROM>>;"... invite_buf_end = 0x7fcdf8a62c70 "" cancel_buf = 0x415440 "1\355I\211\321^H\211\342H\203\344\360PTI\307\300\300\363m" s = 0x7fcdf8a62868 "INVITE sip:<CALLED>@goren SIP/2.0\r\nRecord-Route: <sip:A.B.C5.60;lr;did=5b1.3112>\r\nCSeq: 1 INVITE\r\nCall-ID: 3da8-469-1111201513435-JANGO-1-A.B.C_leg2\r\nFrom: <sip:<CALLING>@<HOST_FROM>>;"... s1 = 0x0 d = 0x10703ab0 <Address 0x10703ab0 out of bounds> invite_len = 1032 hf_type = 2540 first_via = 0 to_len = 67 cancel_buf_len = 1099 reason_len = 0 code_len = 0 reas1 = 0x0 reas_last = 0x0 hdr = 0xa7294e __FUNCTION__ = "build_local_reparse" #4 0x00007fce0f02e58b in build_ack (rpl=0x7fce107038b8, trans=0x7fcdf8a5e2f0, branch=0, ret_len=0x7fff00c32fbc) at t_reply.c:439 to = {s = 0xa728f6 "To: <sip:+<CALLED>@A.B.C5.60:5060>;tag=3658827875-928685\r\nContent-Length: 0\r\n\r\n", len = 67} __FUNCTION__ = "build_ack" #5 0x00007fce0f03b3ca in reply_received (p_msg=0x7fce107038b8) at t_reply.c:2253 msg_status = 488 last_uac_status = 100 ack = 0x7fce10588010 "\001" ack_len = 0 branch = 0 reply_status = 274235864 onreply_route = 1 cancel_data = {cancel_bitmap = 0, reason = {cause = 0, u = {text = {s = 0x0, len = 10955086}, e2e_cancel = 0x0, packed_hdrs = {s = 0x0, len = 10955086}}}} uac = 0x7fcdf8a5e458 t = 0x7fcdf8a5e2f0 lack_dst = {send_sock = 0x13a948, to = {s = {sa_family = 10306, sa_data = "\247\000\000\000\000\000(\r\030\000\000\000\000"}, sin = {sin_family = 10306, sin_port = 167, sin_addr = {s_addr = 0}, sin_zero = "(\r\030\000\000\000\000"}, sin6 = {sin6_family = 10306, sin6_port = 167, sin6_flowinfo = 0, sin6_addr = { __in6_u = {__u6_addr8 = "(\r\030\000\000\000\000\000\270|^\020\316\177\000", __u6_addr16 = {3368, 24, 0, 0, 31928, 4190, 32718, 0}, __u6_addr32 = {1576232, 0, 274627768, 32718}}}, sin6_scope_id = 93}}, id = 0, proto = -72 '\270', send_flags = {f = 106 'j', blst_imask = 94 '^'}} backup_user_from = 0x758a50 backup_user_to = 0x415440 backup_domain_from = 0x300c337e0 backup_domain_to = 0x0 backup_uri_from = 0x7fce10703ab0 backup_uri_to = 0x7fff00c330d0 backup_xavps = 0x628478 replies_locked = 0 branch_ret = 0 prev_branch = 275790040 blst_503_timeout = 32718 hf = 0x18e00c33060 onsend_params = {req = 0x7fff00c32f50, rpl = 0x47deb8, param = 0x0, code = 274623280, flags = 32718, branch = 0, t_rbuf = 0xa7294e, dst = 0xa727e1, send_buf = { s = 0x7fff00c33010 "\320\060", <incomplete sequence \303>, len = 6402299}} ctx = {rec_lev = 7715456, run_flags = 0, last_retcode = 7704068, jmp_env = {{__jmpbuf = {232, 140523014225936, 140523014226728, 140523014618240, 29, 0, 140523015781040, 4281408}, __mask_was_saved = 12793824, __saved_mask = {__val = {0, 140733206179856, 6439748, 140733206179616, 140523001001690, 140733206179584, 0, 67108864, 65537912, 1288288, 1570952, 1576232, 8, 94, 0, 1473240891392}}}}} __FUNCTION__ = "reply_received" #6 0x000000000048cc3a in do_forward_reply (msg=0x7fce107038b8, mode=0) at forward.c:783 new_buf = 0x0 dst = {send_sock = 0x0, to = {s = {sa_family = 0, sa_data = '\000' <repeats 13 times>}, sin = {sin_family = 0, sin_port = 0, sin_addr = {s_addr = 0}, sin_zero = "\000\000\000\000\000\000\000"}, sin6 = {sin6_family = 0, sin6_port = 0, sin6_flowinfo = 0, sin6_addr = {__in6_u = { __u6_addr8 = '\000' <repeats 15 times>, __u6_addr16 = {0, 0, 0, 0, 0, 0, 0, 0}, __u6_addr32 = {0, 0, 0, 0}}}, sin6_scope_id = 0}}, id = 0, proto = 0 '\000', send_flags = {f = 0 '\000', blst_imask = 0 '\000'}} new_len = 1 r = 0 ip = {af = 12792240, len = 32767, u = {addrl = {6466393, 280}, addr32 = {6466393, 0, 280, 0}, addr16 = {43865, 98, 0, 0, 280, 0, 0, 0}, addr = "Y\253b\000\000\000\000\000\030\001\000\000\000\000\000"}} s = 0x410004 "" len = 0 __FUNCTION__ = "do_forward_reply" #7 0x000000000048e27d in forward_reply (msg=0x7fce107038b8) at forward.c:885 ---Type <return> to continue, or q <return> to quit--- No locals. #8 0x0000000000509c9c in receive_msg ( buf=0xa727c0 "SIP/2.0 488 Not Acceptable Here\r\nVia: SIP/2.0/UDP A.B.C5.60;rport=5060;branch=z9hG4bKce12.98a05ce27e0243c4d775920d81e7ce82.0\r \nVia: SIP/2.0/UDP <HOST_FROM>;branch=z9hG4bK.5667f6e0\r\nFrom: <sip:332"..., len=398, rcv_info=0x7fff00c333b0) at receive.c:275 msg = 0x7fce107038b8 ctx = {rec_lev = 10237056, run_flags = 0, last_retcode = 0, jmp_env = {{__jmpbuf = {0, 0, 0, 272136986608, 1812476198913, 0, 272145363728, 272145384176}, __mask_was_saved = 0, __saved_mask = {__val = {140523014366192, 140733206180688, 1, 140522613671152, 272137013029, 50195, 1024, 4307759872, 140522613671152, 140733206180608, 6299381, 140733206180896, 140522613671152, 81, 6299509, 140733206180976}}}}} ret = 12792656 inb = { s = 0xa727c0 "SIP/2.0 488 Not Acceptable Here\r\nVia: SIP/2.0/UDP A.B.C5.60;rport=5060;branch=z9hG4bKce12.98a05ce27e0243c4d775920d81e7ce82.0\r \nVia: SIP/2.0/UDP <HOST_FROM>;branch=z9hG4bK.5667f6e0\r\nFrom: <sip:332"..., len = 398} __FUNCTION__ = "receive_msg" #9 0x0000000000608f02 in udp_rcv_loop () at udp_server.c:521 len = 398 buf = "SIP/2.0 488 Not Acceptable Here\r\nVia: SIP/2.0/UDP A.B.C5.60;rport=5060;branch=z9hG4bKce12.98a05ce27e0243c4d775920d81e7ce82.0\r \nVia: SIP/2.0/UDP <HOST_FROM>;branch=z9hG4bK.5667f6e0\r\nFrom: <sip:332"... tmp = 0x3f60c4067a <Address 0x3f60c4067a out of bounds> from = 0x7fce105e7920 fromlen = 16 ri = {src_ip = {af = 2, len = 4, u = {addrl = {151524537, 0}, addr32 = {151524537, 0, 0, 0}, addr16 = {5305, 2312, 0, 0, 0, 0, 0, 0}, addr = "\271\024\b\t", '\000' <repeats 11 times>}}, dst_ip = {af = 2, len = 4, u = {addrl = {1016190299, 0}, addr32 = {1016190299, 0, 0, 0}, addr16 = {54619, 15505, 0, 0, 0, 0, 0, 0}, addr = "[Õ<", '\000' <repeats 11 times>}}, src_port = 5060, dst_port = 5060, proto_reserved1 = 0, proto_reserved2 = 0, src_su = { s = {sa_family = 2, sa_data = "\023Ä¹\024\b\t\000\000\000\000\000\000\000"}, sin = {sin_family = 2, sin_port = 50195, sin_addr = {s_addr = 151524537}, sin_zero = "\000\000\000\000\000\000\000"}, sin6 = {sin6_family = 2, sin6_port = 50195, sin6_flowinfo = 151524537, sin6_addr = {__in6_u = { __u6_addr8 = '\000' <repeats 15 times>, __u6_addr16 = {0, 0, 0, 0, 0, 0, 0, 0}, __u6_addr32 = {0, 0, 0, 0}}}, sin6_scope_id = 0}}, bind_address = 0x7fce105aa2b0, proto = 1 '\001'} __FUNCTION__ = "udp_rcv_loop" #10 0x00000000004a6d9b in main_loop () at main.c:1629 i = 0 pid = 0 si = 0x7fce105aa2b0 si_desc = "udp receiver child=0 sock=A.B.C5.60:5060\000\177\000\000 5\303\000\377\177\000\000\003zN\000\000\000\000\000\016\b\000\000\377\177\00 0\000\260\204x\370\315\177\000\000\000\000\000\020\004\000\000\000\260\204x\ 370\315\177\000\000@TA\000\000\000\000\000\340\067\303\000\001\000\000\000p5 \303\000\377\177\000\000\246zN\000\000\000\000" nrprocs = 8 __FUNCTION__ = "main_loop" #11 0x00000000004acedf in main (argc=7, argv=0x7fff00c337e8) at main.c:2581 cfg_stream = 0xac7010 c = -1 r = 0 tmp = 0x7fff00c34f70 "" tmp_len = 32767 port = 12793534 proto = 0 options = 0x6ff8f8 ":f:cm:M:dVIhEeb:l:L:n:vKrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:" ret = -1 seed = 3913881212 rfd = 4 debug_save = 0 debug_flag = 0 dont_fork_cnt = 0 n_lst = 0x40d11c p = 0xc2 <Address 0xc2 out of bounds> __FUNCTION__ = "main" Regards, Igor.

2 15

Reg: REGISTER request don't have username in request line
by sainath.ellendula＠wipro.com 11 Feb '16

11 Feb '16

Hi Team, Working on Kamailio setup with IMS components(P-CSCF, I-CSCF, S-CSCF). Assigned IP's: hss/server-10.0.0.9 P-CSCF-10.0.0.10 I-CSCF-10.0.0.11 S-CSCF-10.0.0.12 Here tried to send an REGISTER request from Boghe IMS client, the register request should go from P-CSCF to I-CSCF, here P-CSCF - I-CSCF communication is not happening but I could able ping. One more thing observed from logs here is REGISTER request don't have username (username is not there in request line), Execution log: 0(1338) DEBUG: ims_usrloc_pcscf [usrloc.c:106]: get_alias_host_from_contact(): no alias param Please find attachment of Zone file, SCSCF.xml, Diameter.xml, ICSCF.xml and tcpdump. Any changes/suggestions would be very helpful. Regards, Sainath The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com

2 1

Terminate dialogs on [websocket:closed]
by mayamatakeshi 11 Feb '16

11 Feb '16

Hello, I am using module websocket and it works very well. However I would like to send BYE to the other end if event [websocket:closed] happens. >From the docs I can see websocket module itself doesn't provide for this. I was considering doing something like this: - use module htable to match $si:$sp to dialogs - use event_route[dialog:start] to insert dialog info to my htable under $si:$sp of Websocket side of the call - use event_route[dialog:end] to remove dialog info from htable - use event_route[websocket:closed] to iterate over entries in the htable under key $si:$sp and call dlg_get() and dlg_bye(). Obs: in the above, there is a risk of losing some dialogs as insertion in htable cannot be done atomically, but I am fine with it as it it not expected to happen as WebSocket users would only infrequently generate simultaneous calls. However before going with this, I would like to ask for other possible approaches. Regards, Takeshi

2 6

Route for BYE requests
by alexandre.anconetti＠orange.com 11 Feb '16

11 Feb '16

Hello, I would like some advice about a situation i have encountered recently. I'm using Kamailio 4.3.4 as a "proxy" : I have a media server, and to reach this server all the requests have to pass through my Kamailio server. The Kamailio server does not have to handle authentification or registration, all of that stuff is done on the media server. To do that I'm using the following route : route[REDIRECT] { if($rU == "" ) { $ru = ("sip:1.2.3.4:xxxx"); } else { $ru = ("sip:" + $rU + "@1.2.3.4:xxxx"); } } 1.2.3.4 is my media server's address. I have 2 users : an "internal" one registered on the media server, and an "external" one which is a random caller. It works fine for EXT -> Kamailio -> Media server requests but when I try to hang up with my internal user, the "BYE" request uses the following path : Internal user -> Media server -> Kamailio -> Media server instead of : Internal user -> Media server -> Kamailio -> External user. Obviously it is because of the REDIRECT route which send back the "BYE" request to the media server, when it should sends that request to the external user. So, how should I modify that route to do what I want ? Maybe should I create a second route ? Any answer will be appreciated :) Best regards, Alexandre _________________________________________________________________________________________________________________________ Ce message et ses pieces jointes peuvent contenir des informations confidentielles ou privilegiees et ne doivent donc pas etre diffuses, exploites ou copies sans autorisation. Si vous avez recu ce message par erreur, veuillez le signaler a l'expediteur et le detruire ainsi que les pieces jointes. Les messages electroniques etant susceptibles d'alteration, Orange decline toute responsabilite si ce message a ete altere, deforme ou falsifie. Merci. This message and its attachments may contain confidential or privileged information that may be protected by law; they should not be distributed, used or copied without authorisation. If you have received this email in error, please notify the sender and delete this message and its attachments. As emails may be altered, Orange is not liable for messages that have been modified, changed or falsified. Thank you.

2 1

kamctl ul add...<path>
by kssyrr 10 Feb '16

10 Feb '16

1 0

RTCWeb Breaker question
by SamyGo 10 Feb '16

10 Feb '16

Hi All, reference to this link: https://www.kamailio.org/wiki/devel/rtcweb_breaker#scenarios I want to know if the module to communicate with RTCWeb Breaker is available or it was just a proposal and no more under consideration. I have webrtc clients registered to Kamailio but due to lack of (scalable/efficient) transcoding capabilities they can not make video calls to Video IP-Phones. I tried using webrtc2sip from doubango telecom and it actually enabled me to achieve the goal, the problem with that case is webrtc2sip is working with sipml5 client and there is not a big list of WebRTC clients that work with it. If I can achieve the referred rtc_web_breaker architecture then I believe a lot of webRTC clients will be able to integrate with my setup. Thanks, Regards, Sammy

2 4

Kamailio SST and Dialog module workings
by Jason Penton 10 Feb '16

10 Feb '16

Hi Kamailio users and Devs. I am interested to know if any of you are using SST module with DIALOG module and sharing the timeout avp between the two - expecting the call to be torn down if no in-dialog requests are processed. If you are using it, is it working as you expect? To me it seems that the timers in the dialog module will never be updated by any in-dialog requests. I just wanted to check before I start looking at fixing... perhaps I am missing something. Cheers Jason

2 2

Permission module about from_pattern
by Yasin CANER 10 Feb '16

10 Feb '16

2 3

regular expression for fromuser
by Yasin CANER 10 Feb '16

10 Feb '16

1 0

regular expression for from_pattern
by Yasin CANER 10 Feb '16

10 Feb '16

1 0

info: dialog_ng renamed to ims_dialog
by Daniel-Constantin Mierla 09 Feb '16

09 Feb '16

Hello, based on a previous discussion on mailing lists, I just renamed the dialog_ng to ims_dialog in order to reflect better its purpose nowadays (use together with the other ims modules) and avoid confusion that it is something that includes the features of dialog module and some new stuff. Functions and params are not renamed, being anyhow overlapping a lot with dialog module, but they are not intended to be used at the same time. Perhaps some examples still need to be updated (contributions are welcome here if you identify such places). If any other issue is discovered to the ims modules, write here to get it fixed. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com http://miconda.eu

1 0

Re: [SR-Users] Kamailio IMS deployment
by Franz Edler 09 Feb '16

09 Feb '16

The REGISTER request is OK. There is no username included in REGISTER request.Regarding the mismatch: do you use NATURAL on your platform? Next step should be a tracefile (tcpdump) and later a debug-log from the node which shows the problem. Von Samsung-Tablet gesendet-------- Ursprüngliche Nachricht --------Von: sainath.ellendula(a)wipro.com Datum: 2016.02.09 08:05 (GMT+01:00) An: franz.edler(a)technikum-wien.at Cc: basitstar(a)hotmail.com, sr-users(a)lists.sip-router.org Betreff: RE: Re:[SR-Users] Kamailio IMS deployment Hi Franz, Please find ping status and seems good. root@server:~# ping -c 1 hss.net1.test PING hss.net1.test (10.0.0.9) 56(84) bytes of data. 64 bytes from server.net1.test (10.0.0.9): icmp_seq=1 ttl=64 time=0.017 ms --- hss.net1.test ping statistics --- 1 packets transmitted, 1 received, 0% packet loss, time 0ms rtt min/avg/max/mdev = 0.017/0.017/0.017/0.000 ms root@server:~# ping -c 1 pcscf.net1.test PING pcscf.net1.test (10.0.0.10) 56(84) bytes of data. 64 bytes from server.net1.test (10.0.0.10): icmp_seq=1 ttl=64 time=0.020 ms --- pcscf.net1.test ping statistics --- 1 packets transmitted, 1 received, 0% packet loss, time 0ms rtt min/avg/max/mdev = 0.020/0.020/0.020/0.000 ms root@server:~# ping -c 1 icscf.net1.test PING icscf.net1.test (10.0.0.11) 56(84) bytes of data. 64 bytes from server.net1.test (10.0.0.11): icmp_seq=1 ttl=64 time=0.018 ms --- icscf.net1.test ping statistics --- 1 packets transmitted, 1 received, 0% packet loss, time 0ms rtt min/avg/max/mdev = 0.018/0.018/0.018/0.000 ms root@server:~# ping -c 1 scscf.net1.test PING scscf.net1.test (10.0.0.12) 56(84) bytes of data. 64 bytes from server.net1.test (10.0.0.12): icmp_seq=1 ttl=64 time=0.017 ms --- scscf.net1.test ping statistics --- 1 packets transmitted, 1 received, 0% packet loss, time 0ms rtt min/avg/max/mdev = 0.017/0.017/0.017/0.000 ms root@server:~# Seems Some progress happened: Tried register using Boghe IMS client and observed few errors: 0(1445) INFO: <script>: [REGISTER] from [sip:alice@net1.test] to [sip:alice@net1.test] 0(1445) ERROR: <script>: REGISTER (sip:alice@net1.test (10.0.0.8:55280) to sip:alice@net1.test, 2006cdca-0076-bacb-5010-bffa714b3cdb) 0(1445) DEBUG: <core> [parser/msg_parser.c:153]: get_hdr_field(): get_hdr_field: cseq <CSeq>: <7994> <REGISTER> 0(1445) DEBUG: <core> [parser/msg_parser.c:187]: get_hdr_field(): DEBUG: get_hdr_body : content_length=0 0(1445) DEBUG: maxfwd [mf_funcs.c:85]: is_maxfwd_present(): value = 70 0(1445) DEBUG: <core> [parser/msg_parser.c:89]: get_hdr_field(): found end of header 0(1445) DEBUG: sanity [mod_sanity.c:248]: w_sanity_check(): sanity checks result: 1 0(1445) DEBUG: siputils [checks.c:97]: has_totag(): no totag 0(1445) DEBUG: <core> [select.c:412]: run_select(): Calling SELECT 0x7fb548dd2400 0(1445) DEBUG: tm [t_lookup.c:1011]: t_check_msg(): DEBUG: t_check_msg: msg id=2 global id=1 T start=0xffffffffffffffff 0(1445) DEBUG: tm [t_lookup.c:466]: t_lookup_request(): t_lookup_request: start searching: hash=65064, isACK=0 0(1445) DEBUG: tm [t_lookup.c:424]: matching_3261(): DEBUG: RFC3261 transaction matching failed 0(1445) DEBUG: tm [t_lookup.c:648]: t_lookup_request(): DEBUG: t_lookup_request: no transaction found 0(1445) DEBUG: tm [t_lookup.c:1080]: t_check_msg(): DEBUG: t_check_msg: msg id=2 global id=2 T end=(nil) 0(1445) DEBUG: sl [sl.c:280]: send_reply(): reply in stateless mode (sl) 0(1445) DEBUG: siputils [checks.c:97]: has_totag(): no totag 0(1445) DEBUG: tm [t_lookup.c:1011]: t_check_msg(): DEBUG: t_check_msg: msg id=2 global id=2 T start=(nil) 0(1445) DEBUG: tm [t_lookup.c:1086]: t_check_msg(): DEBUG: t_check_msg: T previously sought and not found 0(1445) DEBUG: ims_registrar_pcscf [save.c:273]: save_pending(): contact requesting to expire in 600000 seconds 0(1445) DEBUG: ims_usrloc_pcscf [usrloc.c:106]: get_alias_host_from_contact(): no alias param 0(1445) DEBUG: ims_usrloc_pcscf [usrloc.c:155]: get_hash_slot(): using host for hash [10.0.0.8] 0(1445) DEBUG: ims_usrloc_pcscf [usrloc.c:159]: get_hash_slot(): Returning hash slot: [167] 0(1445) DEBUG: ims_usrloc_pcscf [udomain.c:465]: get_pcontact(): Searching for contact in P-CSCF usrloc [sip:alice@10.0.0.8:55280;transport=udp] 0(1445) DEBUG: ims_usrloc_pcscf [usrloc.c:106]: get_alias_host_from_contact(): no alias param 0(1445) DEBUG: ims_usrloc_pcscf [usrloc.c:183]: get_aor_hash(): using host for hash [10.0.0.8] 0(1445) DEBUG: ims_usrloc_pcscf [usrloc.c:187]: get_aor_hash(): Returning hash slot: [1662767783] 0(1445) DEBUG: ims_usrloc_pcscf [udomain.c:483]: get_pcontact(): Comparing needle user@host:port [alice@10.0.0.8:55280] and contact_user@contact_host:port [alice@10.0.0.8:51219] 0(1445) DEBUG: ims_usrloc_pcscf [udomain.c:494]: get_pcontact(): Searching for [sip:alice@10.0.0.8:55280;transport=udp] and comparing to [sip:alice@10.0.0.8:51219;transport=udp] 0(1445) DEBUG: ims_usrloc_pcscf [udomain.c:553]: get_pcontact(): Port don't match: 51219 (contact) 51219 (received) != 55280! 0(1445) DEBUG: ims_registrar_pcscf [save.c:310]: save_pending(): Adding pending pcontact: <sip:alice@10.0.0.8:55280;transport=udp> 0(1445) DEBUG: ims_usrloc_pcscf [pcontact.c:132]: new_pcontact(): New contact [<sip:alice@10.0.0.8:55280;transport=udp>] with 0 associated IMPUs in state: [registration pending] 0(1445) DEBUG: ims_registrar_pcscf [save.c:314]: save_pending(): registering for UL callback 0(1445) DEBUG: ims_usrloc_pcscf [usrloc.c:106]: get_alias_host_from_contact(): no alias param Please guide me for any more changes has to be done, As observed in Wireshark(Register request from 10.0.0.8(host)-PCSCF as REGISTER sip:net1.test SIP/2.0). I think the register request should sent as REGISTER sip:alice@net1.test SIP/2.0 Thanks in advance!! Regards, Sainath From: Franz Edler [mailto:franz.edler@technikum-wien.at] Sent: Tuesday, February 9, 2016 3:00 AM To: Sainath Ellendula (NEP) <sainath.ellendula(a)wipro.com> Cc: basitstar(a)hotmail.com; sr-users(a)lists.sip-router.org Subject: RE: Re:[SR-Users] Kamailio IMS deployment Sainath, what was the result for ping on - icsct.net1.test - scscf.net1.test - hss.net1.test to increase debug level you have to modify the corresponding cfg-file, e.g. kamailio-icscf.cfg in line 35: debug=2 Just increase to debug=3 and then start the process again. BR Franz The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com

1 0

GCC version in use
by Daniel-Constantin Mierla 09 Feb '16

09 Feb '16

Hello, looking a bit over our makefiles, I am thinking of simplifying some parts related to compiler optimizations, as I noticed some of them are for gcc 2.x. So I would like to see if people are still using old OSes and/or old GCC versions. In other words, if you use old or some dedicated systems and plan to deploy most recent Kamailio versions, reply here with the gcc version in use, particularly interested to see the use of gcc 2.x or 3.x. The idea is not to remove the support for it completely, but to simply and don't try to find all tunings for old versions, but go more with the default gcc options. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com http://miconda.eu

3 7

Re: [SR-Users] Kamailio IMS deployment
by sainath.ellendula＠wipro.com 09 Feb '16

09 Feb '16

Hi Franz, Thank you so much for your response. As suggested modified changes and don't observe connection refused error. After modification few more errors observed. Detailed errors as below. 10.0.0.9 /catalina-optional.jar:lib/catalina.jar:lib/c3p0-0.9.1.jar:lib/base64.jar:lib/asm.jar:lib/asm-attrs.jar:lib/antlr-2.7.6.jar::log4j.properties:.. 2016-02-01 12:09:28,371 INFO de.fhg.fokus.hss.main.TomcatServer - startTomcat Tomcat-Server is started. 2016-02-01 12:09:29,301 WARN org.apache.catalina.connector.MapperListener - registerEngine Unknown default host: 10.0.0.9 2016-02-01 12:09:30,294 INFO de.fhg.fokus.hss.web.servlet.ResponseFilter - init Response Filter Initialisation! 2016-02-01 12:09:31,251 INFO de.fhg.fokus.hss.main.TomcatServer - startTomcat WebConsole of FHoSS was started ! 2016-02-01 12:09:34,813 WARN org.hibernate.impl.SessionFactoryObjectFactory - addInstance InitialContext did not implement EventContext 2016-02-01 12:09:34,847 INFO de.fhg.fokus.diameter.DiameterPeer.DiameterPeer - <init> Bean style constructor called, don't forget to configure! 2016-02-01 12:09:34,851 INFO de.fhg.fokus.diameter.DiameterPeer.DiameterPeer - configure FQDN: hss.net1.test 2016-02-01 12:09:34,852 INFO de.fhg.fokus.diameter.DiameterPeer.DiameterPeer - configure Realm: net1.test 2016-02-01 12:09:34,853 INFO de.fhg.fokus.diameter.DiameterPeer.DiameterPeer - configure Vendor_ID : 10415 2016-02-01 12:09:34,854 INFO de.fhg.fokus.diameter.DiameterPeer.DiameterPeer - configure Product Name: JavaDiameterPeer 2016-02-01 12:09:34,854 INFO de.fhg.fokus.diameter.DiameterPeer.DiameterPeer - configure AcceptUnknwonPeers: true 2016-02-01 12:09:34,855 INFO de.fhg.fokus.diameter.DiameterPeer.DiameterPeer - configure DropUnknownOnDisconnect: true 2016-02-01 12:09:34,864 INFO de.fhg.fokus.hss.main.HSSContainer - waitForExit Type "exit" to stop FHoSS! 2016-02-01 12:09:34,873 ERROR de.fhg.fokus.diameter.DiameterPeer.peer.StateMachine - I_Snd_Conn_Req StateMachine: Peer icscf.net1.test:3869 not responding to connection attempt 2016-02-01 12:09:34,876 ERROR de.fhg.fokus.diameter.DiameterPeer.peer.StateMachine - I_Snd_Conn_Req StateMachine: Peer scscf.net1.test:3870 not responding to connection attempt 10.0.0.10 root@server:~# kamailio -f /usr/local/etc/kamailio/kamailio-pcscf.cfg loading modules under config path: /usr/local/lib64/kamailio/modules/ 0(1856) INFO: <core> [sctp_core.c:75]: sctp_core_check_support(): SCTP API not enabled - if you want to use it, load sctp module Listening on udp: pcscf.net1.test [10.0.0.10]:4060 Aliases: *: pcscf.net1.test:* WARNING: no fork mode 0(1856) INFO: rr [../outbound/api.h:54]: ob_load_api(): Failed to import bind_ob 0(1856) INFO: rr [rr_mod.c:174]: mod_init(): outbound module not available 0(1856) INFO: ims_usrloc_pcscf [hslot.c:62]: ul_init_locks(): locks array size 512 0(1856) INFO: <core> [udp_server.c:150]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 0(1856) INFO: <core> [udp_server.c:200]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 0(1856) WARNING: ims_usrloc_pcscf [usrloc_db.c:59]: connect_db(): DB connection already open... continuing 0(1856) INFO: <script>: [REGISTER] from [sip:alice@net1.test] to [sip:alice@net1.test] 0(1856) ERROR: <script>: REGISTER (sip:alice@net1.test (10.0.0.8:55859) to sip:alice@net1.test, aaeacddd-376e-4a63-5745-2757d1b9a29d) 10.0.0.11 9(1870) WARNING: cdp [tcp_accept.c:123]: create_socket(): create_socket(): Trying to open/bind/listen on 10.0.0.11 port 3869 9(1870) WARNING: cdp [tcp_accept.c:148]: create_socket(): create_socket(): Successful socket open/bind/listen on 10.0.0.11 port 3869 9(1870) INFO: cdp [acceptor.c:96]: acceptor_process(): Acceptor opened sockets. Entering accept loop ... 10(1871) INFO: cdp [peermanager.c:280]: peer_timer(): peer_timer(): Peer hss.net1.test State 0 10(1871) INFO: cdp [peerstatemachine.c:526]: I_Snd_Conn_Req(): I_Snd_Conn_Req(): Peer hss.net1.test 10(1871) WARNING: cdp [receiver.c:864]: peer_connect(): peer_connect(): Trying to connect to 10.0.0.9 port 3868 10(1871) INFO: cdp [receiver.c:932]: peer_connect(): peer_connect(): Peer hss.net1.test:3868 connected 8(1869) ERROR: cdp [peerstatemachine.c:635]: I_Snd_CER(): I_Snd_CER(): Error on finding local host address > Socket operation on non-socket 10(1871) INFO: cdp [peermanager.c:280]: peer_timer(): peer_timer(): Peer hss.net1.test State 6 10(1871) WARNING: cdp [peermanager.c:315]: peer_timer(): Inactivity on peer [hss.net1.test], sending DWR... - if we don't get a reply, the peer will be closed 10.0.0.12 9(1883) WARNING: cdp [tcp_accept.c:123]: create_socket(): create_socket(): Trying to open/bind/listen on 10.0.0.12 port 3870 9(1883) WARNING: cdp [tcp_accept.c:148]: create_socket(): create_socket(): Successful socket open/bind/listen on 10.0.0.12 port 3870 9(1883) INFO: cdp [acceptor.c:96]: acceptor_process(): Acceptor opened sockets. Entering accept loop ... 7(1881) INFO: cdp [receiver.c:445]: receiver_process(): receiver_process(): [] Receiver process doing init on new process... 7(1881) INFO: cdp [receiver.c:450]: receiver_process(): receiver_process(): [] Receiver process starting up... 10(1884) INFO: cdp [peermanager.c:280]: peer_timer(): peer_timer(): Peer hss.net1.test State 0 10(1884) INFO: cdp [peerstatemachine.c:526]: I_Snd_Conn_Req(): I_Snd_Conn_Req(): Peer hss.net1.test 10(1884) WARNING: cdp [receiver.c:864]: peer_connect(): peer_connect(): Trying to connect to 10.0.0.9 port 3868 10(1884) INFO: cdp [receiver.c:932]: peer_connect(): peer_connect(): Peer hss.net1.test:3868 connected 8(1882) ERROR: cdp [peerstatemachine.c:635]: I_Snd_CER(): I_Snd_CER(): Error on finding local host address > Socket operation on non-socket 10(1884) INFO: cdp [peermanager.c:280]: peer_timer(): peer_timer(): Peer hss.net1.test State 6 10(1884) WARNING: cdp [peermanager.c:315]: peer_timer(): Inactivity on peer [hss.net1.test], sending DWR... - if we don't get a reply, the peer will be closed 10(1884) INFO: cdp [peermanager.c:280]: peer_timer(): peer_timer(): Peer hss.net1.test State 6 Could you please guide me here. Any suggestions would be very helpful. Thanks in advance!! Regards, Sainath -----Original Message----- From: Franz Edler [mailto:franz.edler@technikum-wien.at] Sent: Saturday, January 30, 2016 1:27 PM To: Sainath Ellendula (WT01 - NEP) <sainath.ellendula(a)wipro.com>; basitstar(a)hotmail.com; sr-users(a)lists.sip-router.org Subject: RE: [SR-Users] Kamailio IMS deployment Hi, > As suggested, modified the changes in icscf.xml and scscf.xml files, PFA > modified files. > > Still getting few errors .... Now it seems to be clear to me where the problem might come from: I have restricted access to mysql server to address 10.0.0.9 only. You have to change the bind-address statement in /etc/mysql.cnf accordingly. See: http://dev.mysql.com/doc/refman/5.7/en/server-options.html#option_mysqld_bi… Change the address to "*" or to 0.0.0.0. BR Franz _______________________________________________ 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 The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com

2 6

No contact info(0 bindings) in 200-register
by sgy 08 Feb '16

08 Feb '16

Hi Members, I notice an issue: the client register to kamailio server and could receive 200-register message, but no contact infomation(shows 0 bindings) returns in the message(200-register). I checked contact exist in register message and client shows register successfully. When I run command "kamctl online", the client is not online. Could you met with this issue? Expected to your help, thank you very much. Register: REGISTER sip:12.2.1.197 SIP/2.0 Via: SIP/2.0/UDP 12.2.1.180:5060;rport;branch=z9hG4bKPjWScZ6O7MbSbtauTPHcKpjzsHb5U6wxGP Route: <sip:12.2.1.197;lr> Max-Forwards: 70 From: "2016" <sip:2016@12.2.1.197>;tag=jqx1XzTgoS-XKh-S3LyrYmjrCBSqJZ48 To: "2016" <sip:2016@12.2.1.197> Call-ID: AiD6dDi6YOK4.ArBsMpse2MQEoh4dBmP CSeq: 12358 REGISTER User-Agent: A-client Contact: "2016" <sip:2016@12.2.1.180:5060;ob>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-000000020805>";reg-id=0 Supported: geolocation, gruu, path, outbound, answermode, tdialog, replaces, timer, precondition, norefersub Expires: 3600 Allow: INVITE, ACK, BYE, CANCEL, UPDATE, INFO, NOTIFY, REFER, MESSAGE, OPTIONS Authorization: Digest username="2016", realm="12.2.1.197", nonce="Vq8AMFau/wTtkI9/v58I86rqhbvPhSTW", uri="sip:12.2.1.197", response="aebb4a141197b2200bdfc5dce008d9ac" Content-Length: 0 200-Register: //No contact infomation rE/(a)SIP/2.0 200 OK Via: SIP/2.0/UDP 12.2.1.180:5060;rport=5060;branch=z9hG4bKPjpBm-dzHb.9vJY19Ccy9Yz8sx7jigkAM1;received=12.2.1.180 From: "2015" <sip:2015@12.2.1.197>;tag=tVTyij9jGH8wprKa2olsevAockWFs8k0 To: "2015" <sip:2015@12.2.1.197>;tag=090f6aa955fd574e08bca845127013d9.2885 Call-ID: fIQWmPXYtdkT94lzPscrMjzj3R5san9g CSeq: 58932 REGISTER Server: kamailio (4.2.0 (x86_64/linux)) Content-Length: 0 Best Regards, Shengy

2 9

Re: [SR-Users] Replace User Agent
by Daniel-Constantin Mierla 08 Feb '16

08 Feb '16

Hello, for changing the user-agent header you can use: remove_hf("User-Agent"); append_hf("User-Agent: abc\r\n"); Cheers, Daniel On 02/02/16 11:20, Steven Maddox wrote: > Hi, > > I've brought up this topic up before (here and on IRC) and I feel like > I'm getting closer but still a bit stuck. Since it takes a while to > explain the idea (and get past any confusion!) I've summarized it in a > diagram for easier understanding. > > At work my employers have decided to use this god awful hosted PBX > which is picky about which phones can register and checks their user > agent, the hosted PBX is marked on the diagram as the purple blob. > > > > The progress I've made since last time is getting Kamailio 4.2.4 on > that little white device, the *only* purpose of that device is to > change the user agent - nothing more... since it has lots of ports it > could have lots of "Unapproved" phones directly connected to it. > > Someone gave me a link to this as a starting point... > https://blog.voipxswitch.com/2015/03/27/kamailio-basic-sip-proxy-all-reques… > > However I'm not sure what portions of the default kamailio.cfg should > be kept to use this, additionally it doesn't include anything about > changing user agents. > > The normal configuration of an "Approved" phone is *like*... > > username="441234111999" > password="rAnD0Mp4s5" > auth_user="jsmith" > domain="company.co.uk" > registrar/proxy="uk.ic.56hnt.hlpcom.co.uk" > > In terms of the configuration of an "Unapproved" phone I can either... > > 1) Put in exactly the same as above (as though it isn't intending to > contact Kamailio)... but since the phone is directly connected to the > Kamailio device - perhaps Kamailio could watch SIP traffic coming in > on eth1 (even if it isn't addressed to it), alter the user agent and > pass it out through eth0 ? I'm not sure even Kamailio is capable of > this - but it'd be cool if possible. > > 2) Put in exactly the same as above however just change the > registrar/proxy to be the IP of the Kamailio device... Kamailio can > then keep all other configuration the same as it passes it through - > just change the user agent. > > Any thoughts? > > n.b. > 1) Yes changing the user agent definitely does allow you connect, it's > not just a guess - tested it with Asterisk. > 2) No I don't think the hosted PBX supports path extensions - I'd > rather not rely on it supporting undocumented stuff anyway. > > -- > Steven Maddox > > > _______________________________________________ > 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 Book: SIP Routing With Kamailio - http://www.asipto.com http://miconda.eu

1 0

Users in different domains and multiple domains
by Barış Şekerciler 08 Feb '16

08 Feb '16

Hello everyone! I'm wondering something about Kamailio. Here is the idea: Such and example, I want to give base system which depends on Kamailio to 3 companies in 1 server. So I have 1 Kamailio Server and 4 or 5 Freeswitch server. Thus, I have to set a domain for each company, so total is I need 3 domains. For example a, b, c are users. But they should in the different domains for know which is which. In a word, it should seems like this: a(a)subdomain.domain.com b(a)subdomain.domain.com c(a)subdomain.domain.com a(a)subdomain2.domain.com b(a)subdomain2.domain.com c(a)subdomain2.domain.com or a(a)subdomain.domain2.com b(a)subdomain.domain2.com c(a)subdomain.domain2.com etc... So where can I do this? Remember, Kamailio should in the 1 server and all of domains and users should in this server (I will use Postgres) For more details, you can write. Thank you. Regards.

2 5

Error or Bug: [t_msgbuilder.c:490]: build_local_reparse(): ERROR: build_local_reparse: HDR_EOH_T was not found
by Sid Mason 07 Feb '16

07 Feb '16

Hi, Below is the SIP record captured using NGREP: http://pastebin.com/hUqP56fw Regards

2 1

Error or Bug: [t_msgbuilder.c:490]: build_local_reparse(): ERROR: build_local_reparse: HDR_EOH_T was not found
by Sid Mason 07 Feb '16

07 Feb '16

Hi, Below is the SIP record captured using NGREP Regards U 2016/02/07 20:20:06.054910 90.X.X.68:25189 -> 10.1.4.23:5060 INVITE sip:0031651022945@voip.system.com SIP/2.0. Via: SIP/2.0/UDP 10.1.10.11:51106 ;branch=z9hG4bK-d8754z-17160a2503261a6b-1---d8754z-;rport. Max-Forwards: 70. Contact: <sip:9005@90.X.X.68:25189>. To: "003165656565"<sip:003165656565@voip.system.com>. From: "+3165656565"<sip:9005@voip.system.com>;tag=d7622f3c. Call-ID: MzEyNTgyOGYwODU0NTEwZjE2OGJmMWVmNzRjMzkxZGI.. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. User-Agent: eyeBeam release 1102q stamp 51814. Content-Length: 230. . v=0. o=- 1 2 IN IP4 10.1.10.11. s=CounterPath eyeBeam 1.5. c=IN IP4 10.1.10.11. t=0 0. m=audio 52282 RTP/AVP 0 8 101. a=alt:1 1 : h0yu+bYK 4sdGkNzr 10.1.10.11 52282. a=fmtp:101 0-15. a=rtpmap:101 telephone-event/8000. a=sendrecv. U 2016/02/07 20:20:06.056642 10.1.4.23:5060 -> 90.X.X.68:25189 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 10.1.10.11:51106 ;branch=z9hG4bK-d8754z-17160a2503261a6b-1---d8754z-;rport=25189;received=90.X.X.68. To: "003165656565"<sip:003165656565@voip.system.com >;tag=2b9ac8ee2b24b631c656d775cbaad9e0.8324. From: "+3165656565"<sip:9005@voip.system.com>;tag=d7622f3c. Call-ID: MzEyNTgyOGYwODU0NTEwZjE2OGJmMWVmNzRjMzkxZGI.. CSeq: 1 INVITE. Proxy-Authenticate: Digest realm="voip.system.com", nonce="Vrc3ola3NnZiKoF8g2hRTk/0p4EeWKme". Content-Length: 0. . U 2016/02/07 20:20:06.059536 90.X.X.68:25189 -> 10.1.4.23:5060 ACK sip:003165656565@voip.system.com SIP/2.0. Via: SIP/2.0/UDP 10.1.10.11:51106 ;branch=z9hG4bK-d8754z-17160a2503261a6b-1---d8754z-;rport. To: "003165656565"<sip:003165656565@voip.system.com >;tag=2b9ac8ee2b24b631c656d775cbaad9e0.8324. From: "+3165656565"<sip:9005@voip.system.com>;tag=d7622f3c. Call-ID: MzEyNTgyOGYwODU0NTEwZjE2OGJmMWVmNzRjMzkxZGI.. CSeq: 1 ACK. Content-Length: 0. . U 2016/02/07 20:20:06.061317 90.X.X.68:25189 -> 10.1.4.23:5060 INVITE sip:003165656565@voip.system.com SIP/2.0. Via: SIP/2.0/UDP 10.1.10.11:51106 ;branch=z9hG4bK-d8754z-06468b691617fc32-1---d8754z-;rport. Max-Forwards: 70. Contact: <sip:9005@90.X.X.68:25189>. To: "003165656565"<sip:003165656565@voip.system.com>. From: "+3165656565"<sip:9005@voip.system.com>;tag=d7622f3c. Call-ID: MzEyNTgyOGYwODU0NTEwZjE2OGJmMWVmNzRjMzkxZGI.. CSeq: 2 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Proxy-Authorization: Digest username="9005",realm="voip.system.com ",nonce="Vrc3ola3NnZiKoF8g2hRTk/0p4EeWKme",uri=" sip:003165656565@voip.system.com ",response="3473bd0cf6c9af95690edc953fd7346d",algorithm=MD5. User-Agent: eyeBeam release 1102q stamp 51814. Content-Length: 230. . v=0. o=- 1 2 IN IP4 10.1.10.11. s=CounterPath eyeBeam 1.5. c=IN IP4 10.1.10.11. t=0 0. m=audio 52282 RTP/AVP 0 8 101. a=alt:1 1 : h0yu+bYK 4sdGkNzr 10.1.10.11 52282. a=fmtp:101 0-15. a=rtpmap:101 telephone-event/8000. a=sendrecv. U 2016/02/07 20:20:06.065526 10.1.4.23:5060 -> 90.X.X.68:25189 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 10.1.10.11:51106 ;branch=z9hG4bK-d8754z-06468b691617fc32-1---d8754z-;rport=25189;received=90.X.X.68. To: "003165656565"<sip:003165656565@voip.system.com>. From: "+3165656565"<sip:9005@voip.system.com>;tag=d7622f3c. Call-ID: MzEyNTgyOGYwODU0NTEwZjE2OGJmMWVmNzRjMzkxZGI.. CSeq: 2 INVITE. Content-Length: 0. . U 2016/02/07 20:20:06.065940 10.1.4.23:5060 -> 103.XXX.XXX.22:5060 INVITE sip:003165656565@voip.system.com SIP/2.0. Record-Route: <sip:voip.system.com ;lr;ftag=d7622f3c;vsf=AAAAAE9fWUVwRQEFXldDAB9XFAQLHlcQGwhXG1NZXQ--;nat=yes>. Via: SIP/2.0/UDP voip.system.com:5060 ;branch=z9hG4bKddff.66b1d70306f62152bc5c1a7a1b68eb82.0. Via: SIP/2.0/UDP 10.1.10.11:51106 ;received=90.X.X.68;branch=z9hG4bK-d8754z-06468b691617fc32-1---d8754z-;rport=25189. Max-Forwards: 69. Contact: <sip:9005@90.X.X.68:25189;alias=90.X.X.68~25189~1>. To: "003165656565"<sip:003165656565@voip.system.com>. From: <sip:voip.system.com:5060>;tag=d7622f3c. Call-ID: MzEyNTgyOGYwODU0NTEwZjE2OGJmMWVmNzRjMzkxZGI.. CSeq: 2 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. User-Agent: eyeBeam release 1102q stamp 51814. Content-Length: 230. Remote-Party-ID: "9005" <sip:9005@103.XXX.XXX.23 >;party=calling;privacy=off;screen=no. v=0. o=- 1 2 IN IP4 10.1.10.11. s=CounterPath eyeBeam 1.5. c=IN IP4 10.1.10.11. t=0 0. m=audio 52282 RTP/AVP 0 8 101. a=alt:1 1 : h0yu+bYK 4sdGkNzr 10.1.10.11 52282. a=fmtp:101 0-15. a=rtpmap:101 telephone-event/8000. a=sendrecv. U 2016/02/07 20:20:06.090448 103.XXX.XXX.22:5060 -> 10.1.4.23:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP voip.system.com:5060 ;branch=z9hG4bKddff.66b1d70306f62152bc5c1a7a1b68eb82.0;received=103.XXX.XXX.23. Via: SIP/2.0/UDP 10.1.10.11:51106 ;received=90.X.X.68;branch=z9hG4bK-d8754z-06468b691617fc32-1---d8754z-;rport=25189. Record-Route: <sip:voip.system.com ;lr;ftag=d7622f3c;vsf=AAAAAE9fWUVwRQEFXldDAB9XFAQLHlcQGwhXG1NZXQ--;nat=yes>. From: <sip:voip.system.com:5060>;tag=d7622f3c. To: "003165656565"<sip:003165656565@voip.system.com>. Call-ID: MzEyNTgyOGYwODU0NTEwZjE2OGJmMWVmNzRjMzkxZGI.. CSeq: 2 INVITE. Server: UltraTone. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Contact: <sip:003165656565@103.XXX.XXX.22:5060>. Content-Length: 0. . U 2016/02/07 20:20:10.883489 103.XXX.XXX.22:5060 -> 10.1.4.23:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP voip.system.com:5060 ;branch=z9hG4bKddff.66b1d70306f62152bc5c1a7a1b68eb82.0;received=103.XXX.XXX.23. Via: SIP/2.0/UDP 10.1.10.11:51106 ;received=90.X.X.68;branch=z9hG4bK-d8754z-06468b691617fc32-1---d8754z-;rport=25189. Record-Route: <sip:voip.system.com ;lr;ftag=d7622f3c;vsf=AAAAAE9fWUVwRQEFXldDAB9XFAQLHlcQGwhXG1NZXQ--;nat=yes>. From: <sip:voip.system.com:5060>;tag=d7622f3c. To: "003165656565"<sip:003165656565@voip.system.com>;tag=as18998769. Call-ID: MzEyNTgyOGYwODU0NTEwZjE2OGJmMWVmNzRjMzkxZGI.. CSeq: 2 INVITE. Server: UltraTone. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Contact: <sip:003165656565@103.XXX.XXX.22:5060>. Content-Length: 0. . U 2016/02/07 20:20:10.883574 103.XXX.XXX.22:5060 -> 10.1.4.23:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP voip.system.com:5060 ;branch=z9hG4bKddff.66b1d70306f62152bc5c1a7a1b68eb82.0;received=103.XXX.XXX.23. Via: SIP/2.0/UDP 10.1.10.11:51106 ;received=90.X.X.68;branch=z9hG4bK-d8754z-06468b691617fc32-1---d8754z-;rport=25189. Record-Route: <sip:voip.system.com ;lr;ftag=d7622f3c;vsf=AAAAAE9fWUVwRQEFXldDAB9XFAQLHlcQGwhXG1NZXQ--;nat=yes>. From: <sip:voip.system.com:5060>;tag=d7622f3c. To: "003165656565"<sip:003165656565@voip.system.com>;tag=as18998769. Call-ID: MzEyNTgyOGYwODU0NTEwZjE2OGJmMWVmNzRjMzkxZGI.. CSeq: 2 INVITE. Server: UltraTone. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Contact: <sip:003165656565@103.XXX.XXX.22:5060>. Content-Type: application/sdp. Content-Length: 327. . v=0. o=root 1558510539 1558510539 IN IP4 103.XXX.XXX.22. s=Useragent. c=IN IP4 103.XXX.XXX.22. t=0 0. m=audio 59130 RTP/AVP 9 18 0 8 101. a=rtpmap:9 G722/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 2016/02/07 20:20:10.884941 10.1.4.23:5060 -> 90.X.X.68:25189 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 10.1.10.11:51106 ;received=90.X.X.68;branch=z9hG4bK-d8754z-06468b691617fc32-1---d8754z-;rport=25189. Record-Route: <sip:voip.system.com ;lr;ftag=d7622f3c;vsf=AAAAAE9fWUVwRQEFXldDAB9XFAQLHlcQGwhXG1NZXQ--;nat=yes>. From: "+3165656565"<sip:9005@voip.system.com>;tag=d7622f3c. To: "003165656565"<sip:003165656565@voip.system.com>;tag=as18998769. Call-ID: MzEyNTgyOGYwODU0NTEwZjE2OGJmMWVmNzRjMzkxZGI.. CSeq: 2 INVITE. Server: UltraTone. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Contact: <sip:003165656565@103.XXX.XXX.22:5060>. Content-Length: 0. . U 2016/02/07 20:20:10.885646 10.1.4.23:5060 -> 90.X.X.68:25189 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 10.1.10.11:51106 ;received=90.X.X.68;branch=z9hG4bK-d8754z-06468b691617fc32-1---d8754z-;rport=25189. Record-Route: <sip:voip.system.com ;lr;ftag=d7622f3c;vsf=AAAAAE9fWUVwRQEFXldDAB9XFAQLHlcQGwhXG1NZXQ--;nat=yes>. From: "+3165656565"<sip:9005@voip.system.com>;tag=d7622f3c. To: "003165656565"<sip:003165656565@voip.system.com>;tag=as18998769. Call-ID: MzEyNTgyOGYwODU0NTEwZjE2OGJmMWVmNzRjMzkxZGI.. CSeq: 2 INVITE. Server: UltraTone. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Contact: <sip:003165656565@103.XXX.XXX.22:5060>. Content-Type: application/sdp. Content-Length: 327. . v=0. o=root 1558510539 1558510539 IN IP4 103.XXX.XXX.22. s=Useragent. c=IN IP4 103.XXX.XXX.22. t=0 0. m=audio 59130 RTP/AVP 9 18 0 8 101. a=rtpmap:9 G722/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 2016/02/07 20:20:20.596774 90.X.X.68:25189 -> 10.1.4.23:5060 CANCEL sip:003165656565@voip.system.com SIP/2.0. Via: SIP/2.0/UDP 10.1.10.11:51106 ;branch=z9hG4bK-d8754z-06468b691617fc32-1---d8754z-;rport. To: "003165656565"<sip:003165656565@voip.system.com>. From: "+3165656565"<sip:9005@voip.system.com>;tag=d7622f3c. Call-ID: MzEyNTgyOGYwODU0NTEwZjE2OGJmMWVmNzRjMzkxZGI.. CSeq: 2 CANCEL. Proxy-Authorization: Digest username="9005",realm="voip.system.com ",nonce="Vrc3ola3NnZiKoF8g2hRTk/0p4EeWKme",uri=" sip:003165656565@voip.system.com ",response="b739a517585b9bdcd997b3e3322c3bad",algorithm=MD5. User-Agent: eyeBeam release 1102q stamp 51814. Content-Length: 0. . U 2016/02/07 20:20:20.599102 10.1.4.23:5060 -> 90.X.X.68:25189 SIP/2.0 500 cancel error. Via: SIP/2.0/UDP 10.1.10.11:51106 ;branch=z9hG4bK-d8754z-06468b691617fc32-1---d8754z-;rport=25189;received=90.X.X.68. To: "003165656565"<sip:003165656565@voip.system.com >;tag=79597765d0ff0775a3cc048295d6500c-3fc1. From: "+3165656565"<sip:9005@voip.system.com>;tag=d7622f3c. Call-ID: MzEyNTgyOGYwODU0NTEwZjE2OGJmMWVmNzRjMzkxZGI.. CSeq: 2 CANCEL. Content-Length: 0. . U 2016/02/07 20:20:23.407575 90.X.X.68:25189 -> 10.1.4.23:5060 REGISTER sip:voip.system.com SIP/2.0. Via: SIP/2.0/UDP 10.1.10.11:51106 ;branch=z9hG4bK-d8754z-7176c672f55aba4c-1---d8754z-;rport. Max-Forwards: 70. Contact: <sip:9005@90.X.X.68:25189;rinstance=5a52b9593b8bd582>. To: "+3165656565"<sip:9005@voip.system.com>. From: "+3165656565"<sip:9005@voip.system.com>;tag=4f0c6043. Call-ID: MzliN2E4ZTRkYjViZWFiN2Y4YjRjZDExMjc4ODg0NWQ.. CSeq: 7 REGISTER. Expires: 3600. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. User-Agent: eyeBeam release 1102q stamp 51814. Authorization: Digest username="9005",realm="voip.system.com ",nonce="Vrc1l1a3NGs8NrsUIZb+Kyau9v9Q9jUn",uri="sip:voip.system.com ",response="602901348ea359e385f99c5d4261ea5c",algorithm=MD5. Content-Length: 0. . U 2016/02/07 20:20:23.409286 10.1.4.23:5060 -> 90.X.X.68:25189 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 10.1.10.11:51106 ;branch=z9hG4bK-d8754z-7176c672f55aba4c-1---d8754z-;rport=25189;received=90.X.X.68. To: "+3165656565"<sip:9005@voip.system.com >;tag=2b9ac8ee2b24b631c656d775cbaad9e0.fab9. From: "+3165656565"<sip:9005@voip.system.com>;tag=4f0c6043. Call-ID: MzliN2E4ZTRkYjViZWFiN2Y4YjRjZDExMjc4ODg0NWQ.. CSeq: 7 REGISTER. WWW-Authenticate: Digest realm="voip.system.com", nonce="Vrc3s1a3Noe7UjBYG6PYJGPaghIpTBqa". Content-Length: 0. . U 2016/02/07 20:20:23.515175 90.X.X.68:25189 -> 10.1.4.23:5060 REGISTER sip:voip.system.com SIP/2.0. Via: SIP/2.0/UDP 10.1.10.11:51106 ;branch=z9hG4bK-d8754z-9a36b0060c000e77-1---d8754z-;rport. Max-Forwards: 70. Contact: <sip:9005@90.X.X.68:25189;rinstance=5a52b9593b8bd582>. To: "+3165656565"<sip:9005@voip.system.com>. From: "+3165656565"<sip:9005@voip.system.com>;tag=4f0c6043. Call-ID: MzliN2E4ZTRkYjViZWFiN2Y4YjRjZDExMjc4ODg0NWQ.. CSeq: 8 REGISTER. Expires: 3600. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. User-Agent: eyeBeam release 1102q stamp 51814. Authorization: Digest username="9005",realm="voip.system.com ",nonce="Vrc3s1a3Noe7UjBYG6PYJGPaghIpTBqa",uri="sip:voip.system.com ",response="e7d27803d159526ee124e64e59bcc7d0",algorithm=MD5. Content-Length: 0. . U 2016/02/07 20:20:23.520290 10.1.4.23:5060 -> 90.X.X.68:25189 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.1.10.11:51106 ;branch=z9hG4bK-d8754z-9a36b0060c000e77-1---d8754z-;rport=25189;received=90.X.X.68. To: "+3165656565"<sip:9005@voip.system.com >;tag=2b9ac8ee2b24b631c656d775cbaad9e0.351d. From: "+3165656565"<sip:9005@voip.system.com>;tag=4f0c6043. Call-ID: MzliN2E4ZTRkYjViZWFiN2Y4YjRjZDExMjc4ODg0NWQ.. CSeq: 8 REGISTER. Contact: <sip:9005@90.X.X.68 :25189;rinstance=5a52b9593b8bd582>;expires=600;received="sip:90.X.X.68:25189". Content-Length: 0. . U 2016/02/07 20:20:23.520546 10.1.4.23:5060 -> 90.X.X.68:25189 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 10.1.10.11:51106 ;branch=z9hG4bK-d8754z-9a36b0060c000e77-1---d8754z-;rport=25189;received=90.X.X.68. To: "+3165656565"<sip:9005@voip.system.com >;tag=2b9ac8ee2b24b631c656d775cbaad9e0.351d. From: "+3165656565"<sip:9005@voip.system.com>;tag=4f0c6043. Call-ID: MzliN2E4ZTRkYjViZWFiN2Y4YjRjZDExMjc4ODg0NWQ.. CSeq: 8 REGISTER. Contact: <sip:9005@90.X.X.68 :25189;rinstance=5a52b9593b8bd582>;expires=600;received="sip:90.X.X.68:25189". WWW-Authenticate: Digest realm="voip.gpnsystem.com", nonce="Vrc3s1a3Noe7UjBYG6PYJGPaghIpTBqa", qop="auth". Content-Length: 0. . U 2016/02/07 20:20:24.426155 90.X.X.68:25189 -> 10.1.4.23:5060 SUBSCRIBE sip:9005@voip.system.com SIP/2.0. Via: SIP/2.0/UDP 10.1.10.11:51106 ;branch=z9hG4bK-d8754z-3f31377b941acf34-1---d8754z-;rport. Max-Forwards: 70. Contact: <sip:9005@90.X.X.68:25189>. To: "+3165656565"<sip:9005@voip.system.com>. From: "+3165656565"<sip:9005@voip.system.com>;tag=f1000570. Call-ID: NzJiNTM5YjlhOTIyYjFhNWRkNTkzOWQ3MzViNGJiYTg.. CSeq: 1 SUBSCRIBE. Expires: 300. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. User-Agent: eyeBeam release 1102q stamp 51814. Event: message-summary. Content-Length: 0. . U 2016/02/07 20:20:24.428409 10.1.4.23:5060 -> 90.X.X.68:25189 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 10.1.10.11:51106 ;branch=z9hG4bK-d8754z-3f31377b941acf34-1---d8754z-;rport=25189;received=90.X.X.68. To: "+3165656565"<sip:9005@voip.system.com >;tag=2b9ac8ee2b24b631c656d775cbaad9e0.351d. From: "+3165656565"<sip:9005@voip.system.com>;tag=f1000570. Call-ID: NzJiNTM5YjlhOTIyYjFhNWRkNTkzOWQ3MzViNGJiYTg.. CSeq: 1 SUBSCRIBE. Proxy-Authenticate: Digest realm="voip.system.com", nonce="Vrc3tFa3NoiYOjmQMojfuh3ttECnmpZW". Content-Length: 0. . U 2016/02/07 20:20:24.534158 90.X.X.68:25189 -> 10.1.4.23:5060 SUBSCRIBE sip:9005@voip.system.com SIP/2.0. Via: SIP/2.0/UDP 10.1.10.11:51106 ;branch=z9hG4bK-d8754z-4b7b5d2b2726d57d-1---d8754z-;rport. Max-Forwards: 70. Contact: <sip:9005@90.X.X.68:25189>. To: "+3165656565"<sip:9005@voip.system.com>. From: "+3165656565"<sip:9005@voip.system.com>;tag=f1000570. Call-ID: NzJiNTM5YjlhOTIyYjFhNWRkNTkzOWQ3MzViNGJiYTg.. CSeq: 2 SUBSCRIBE. Expires: 300. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Proxy-Authorization: Digest username="9005",realm="voip.system.com ",nonce="Vrc3tFa3NoiYOjmQMojfuh3ttECnmpZW",uri="sip:9005@voip.system.com ",response="2d1f53e6973bf2eb878f679add8bebc8",algorithm=MD5. User-Agent: eyeBeam release 1102q stamp 51814. Event: message-summary. Content-Length: 0. . U 2016/02/07 20:20:24.537592 10.1.4.23:5060 -> 90.X.X.68:25189 SIP/2.0 404 No voicemail service. Via: SIP/2.0/UDP 10.1.10.11:51106 ;branch=z9hG4bK-d8754z-4b7b5d2b2726d57d-1---d8754z-;rport=25189;received=90.X.X.68. To: "+3165656565"<sip:9005@voip.system.com >;tag=2b9ac8ee2b24b631c656d775cbaad9e0.2dea. From: "+3165656565"<sip:9005@voip.system.com>;tag=f1000570. Call-ID: NzJiNTM5YjlhOTIyYjFhNWRkNTkzOWQ3MzViNGJiYTg.. CSeq: 2 SUBSCRIBE. Content-Length: 0. . U 2016/02/07 20:20:25.358979 10.1.4.23:5060 -> 103.XXX.XXX.22:5060 OPTIONS sip:103.XXX.XXX.22:5060 SIP/2.0. Via: SIP/2.0/UDP voip.system.com:5060 ;branch=z9hG4bK828d.ce4b9911000000000000000000000000.0. To: <sip:103.XXX.XXX.22:5060>. From: <sip:dispatcher@localhost>;tag=56254dc816783af7c55a1ac0231200e3-6d5c. CSeq: 10 OPTIONS. Call-ID: 4063eaf942a6e19e-12704(a)10.1.4.23. Max-Forwards: 70. Content-Length: 0. . U 2016/02/07 20:20:25.365672 103.XXX.XXX.22:5060 -> 10.1.4.23:5060 SIP/2.0 404 Not Found. Via: SIP/2.0/UDP voip.system.com:5060 ;branch=z9hG4bK828d.ce4b9911000000000000000000000000.0;received=103.XXX.XXX.23. From: <sip:dispatcher@localhost>;tag=56254dc816783af7c55a1ac0231200e3-6d5c. To: <sip:103.XXX.XXX.22:5060>;tag=as00a791b0. Call-ID: 4063eaf942a6e19e-12704(a)10.1.4.23. CSeq: 10 OPTIONS. Server: UltraTone. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Accept: application/sdp. Content-Length: 0. . U 2016/02/07 20:20:26.250911 103.XXX.XXX.22:5060 -> 10.1.4.23:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP voip.system.com:5060 ;branch=z9hG4bKddff.66b1d70306f62152bc5c1a7a1b68eb82.0;received=103.XXX.XXX.23. Via: SIP/2.0/UDP 10.1.10.11:51106 ;received=90.X.X.68;branch=z9hG4bK-d8754z-06468b691617fc32-1---d8754z-;rport=25189. Record-Route: <sip:voip.system.com ;lr;ftag=d7622f3c;vsf=AAAAAE9fWUVwRQEFXldDAB9XFAQLHlcQGwhXG1NZXQ--;nat=yes>. From: <sip:voip.system.com:5060>;tag=d7622f3c. To: "003165656565"<sip:003165656565@voip.system.com>;tag=as18998769. Call-ID: MzEyNTgyOGYwODU0NTEwZjE2OGJmMWVmNzRjMzkxZGI.. CSeq: 2 INVITE. Server: UltraTone. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Contact: <sip:003165656565@103.XXX.XXX.22:5060>. Content-Type: application/sdp. Content-Length: 327. . v=0. o=root 1558510539 1558510539 IN IP4 103.XXX.XXX.22. s=Useragent. c=IN IP4 103.XXX.XXX.22. t=0 0. m=audio 59130 RTP/AVP 9 18 0 8 101. a=rtpmap:9 G722/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 2016/02/07 20:20:26.252410 10.1.4.23:5060 -> 90.X.X.68:25189 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.1.10.11:51106 ;received=90.X.X.68;branch=z9hG4bK-d8754z-06468b691617fc32-1---d8754z-;rport=25189. Record-Route: <sip:voip.system.com ;lr;ftag=d7622f3c;vsf=AAAAAE9fWUVwRQEFXldDAB9XFAQLHlcQGwhXG1NZXQ--;nat=yes>. From: "+3165656565"<sip:9005@voip.system.com>;tag=d7622f3c. To: "003165656565"<sip:003165656565@voip.system.com>;tag=as18998769. Call-ID: MzEyNTgyOGYwODU0NTEwZjE2OGJmMWVmNzRjMzkxZGI.. CSeq: 2 INVITE. Server: UltraTone. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Contact: <sip:003165656565@103.XXX.XXX.22:5060>. Content-Type: application/sdp. Content-Length: 327. . v=0. o=root 1558510539 1558510539 IN IP4 103.XXX.XXX.22. s=Useragent. c=IN IP4 103.XXX.XXX.22. t=0 0. m=audio 59130 RTP/AVP 9 18 0 8 101. a=rtpmap:9 G722/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 2016/02/07 20:20:26.362474 90.X.X.68:25189 -> 10.1.4.23:5060 ACK sip:003165656565@103.XXX.XXX.22:5060 SIP/2.0. Via: SIP/2.0/UDP 10.1.10.11:51106 ;branch=z9hG4bK-d8754z-8c04097075073c09-1---d8754z-;rport. Max-Forwards: 70. Route: <sip:voip.system.com ;lr;ftag=d7622f3c;vsf=AAAAAE9fWUVwRQEFXldDAB9XFAQLHlcQGwhXG1NZXQ--;nat=yes>. Contact: <sip:9005@90.X.X.68:25189>. To: "003165656565"<sip:003165656565@voip.system.com>;tag=as18998769. From: "+3165656565"<sip:9005@voip.system.com>;tag=d7622f3c. Call-ID: MzEyNTgyOGYwODU0NTEwZjE2OGJmMWVmNzRjMzkxZGI.. CSeq: 2 ACK. Proxy-Authorization: Digest username="9005",realm="voip.system.com ",nonce="Vrc3ola3NnZiKoF8g2hRTk/0p4EeWKme",uri=" sip:003165656565@voip.system.com ",response="3473bd0cf6c9af95690edc953fd7346d",algorithm=MD5. User-Agent: eyeBeam release 1102q stamp 51814. Content-Length: 0. . U 2016/02/07 20:20:26.362493 90.X.X.68:25189 -> 10.1.4.23:5060 BYE sip:003165656565@103.XXX.XXX.22:5060 SIP/2.0. Via: SIP/2.0/UDP 10.1.10.11:51106 ;branch=z9hG4bK-d8754z-fd38c235b8219d09-1---d8754z-;rport. Max-Forwards: 70. Route: <sip:voip.system.com ;lr;ftag=d7622f3c;vsf=AAAAAE9fWUVwRQEFXldDAB9XFAQLHlcQGwhXG1NZXQ--;nat=yes>. Contact: <sip:9005@90.X.X.68:25189>. To: "003165656565"<sip:003165656565@voip.system.com>;tag=as18998769. From: "+3165656565"<sip:9005@voip.system.com>;tag=d7622f3c. Call-ID: MzEyNTgyOGYwODU0NTEwZjE2OGJmMWVmNzRjMzkxZGI.. CSeq: 3 BYE. Proxy-Authorization: Digest username="9005",realm="voip.system.com ",nonce="Vrc3ola3NnZiKoF8g2hRTk/0p4EeWKme",uri="sip:003165656565@103.XXX.XXX.22 :5060",response="e3f1d030387f1b10ab21c126b8e8d656",algorithm=MD5. User-Agent: eyeBeam release 1102q stamp 51814. Content-Length: 0. . U 2016/02/07 20:20:26.365975 10.1.4.23:5060 -> 103.XXX.XXX.22:5060 ACK sip:003165656565@103.XXX.XXX.22:5060 SIP/2.0. Via: SIP/2.0/UDP voip.system.com:5060 ;branch=z9hG4bKddff.9327d42091ecb853a5c80ed2dd347040.0. Via: SIP/2.0/UDP 10.1.10.11:51106 ;received=90.X.X.68;branch=z9hG4bK-d8754z-8c04097075073c09-1---d8754z-;rport=25189. Max-Forwards: 69. Contact: <sip:9005@90.X.X.68:25189;alias=90.X.X.68~25189~1>. To: "003165656565"<sip:003165656565@voip.system.com>;tag=as18998769. From: "+3165656565"<sip:voip.system.com:5060>;tag=d7622f3c. Call-ID: MzEyNTgyOGYwODU0NTEwZjE2OGJmMWVmNzRjMzkxZGI.. CSeq: 2 ACK. Proxy-Authorization: Digest username="9005",realm="voip.system.com ",nonce="Vrc3ola3NnZiKoF8g2hRTk/0p4EeWKme",uri=" sip:003165656565@voip.system.com ",response="3473bd0cf6c9af95690edc953fd7346d",algorithm=MD5. User-Agent: eyeBeam release 1102q stamp 51814. Content-Length: 0. . U 2016/02/07 20:20:26.367367 10.1.4.23:5060 -> 103.XXX.XXX.22:5060 BYE sip:003165656565@103.XXX.XXX.22:5060 SIP/2.0. Via: SIP/2.0/UDP voip.system.com:5060 ;branch=z9hG4bKedff.e74ad188da31f15ff2ec2bcb57878ae2.0. Via: SIP/2.0/UDP 10.1.10.11:51106 ;received=90.X.X.68;branch=z9hG4bK-d8754z-fd38c235b8219d09-1---d8754z-;rport=25189. Max-Forwards: 69. Contact: <sip:9005@90.X.X.68:25189;alias=90.X.X.68~25189~1>. To: "003165656565"<sip:003165656565@voip.system.com>;tag=as18998769. From: "+3165656565"<sip:voip.system.com:5060>;tag=d7622f3c. Call-ID: MzEyNTgyOGYwODU0NTEwZjE2OGJmMWVmNzRjMzkxZGI.. CSeq: 3 BYE. Proxy-Authorization: Digest username="9005",realm="voip.system.com ",nonce="Vrc3ola3NnZiKoF8g2hRTk/0p4EeWKme",uri="sip:003165656565@103.XXX.XXX.22 :5060",response="e3f1d030387f1b10ab21c126b8e8d656",algorithm=MD5. User-Agent: eyeBeam release 1102q stamp 51814. Content-Length: 0. . U 2016/02/07 20:20:26.368319 103.XXX.XXX.22:5060 -> 10.1.4.23:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP voip.system.com:5060 ;branch=z9hG4bKedff.e74ad188da31f15ff2ec2bcb57878ae2.0;received=103.XXX.XXX.23. Via: SIP/2.0/UDP 10.1.10.11:51106 ;received=90.X.X.68;branch=z9hG4bK-d8754z-fd38c235b8219d09-1---d8754z-;rport=25189. From: "+3165656565"<sip:voip.system.com:5060>;tag=d7622f3c. To: "003165656565"<sip:003165656565@voip.system.com>;tag=as18998769. Call-ID: MzEyNTgyOGYwODU0NTEwZjE2OGJmMWVmNzRjMzkxZGI.. CSeq: 3 BYE. Server: UltraTone. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Content-Length: 0. . U 2016/02/07 20:20:26.369422 10.1.4.23:5060 -> 90.X.X.68:25189 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.1.10.11:51106 ;received=90.X.X.68;branch=z9hG4bK-d8754z-fd38c235b8219d09-1---d8754z-;rport=25189. From: "+3165656565"<sip:9005@voip.system.com>;tag=d7622f3c. To: "003165656565"<sip:003165656565@voip.system.com>;tag=as18998769. Call-ID: MzEyNTgyOGYwODU0NTEwZjE2OGJmMWVmNzRjMzkxZGI.. CSeq: 3 BYE. Server: UltraTone. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. Content-Length: 0. . <CAEmfXJ4KeqaPD_7k9pgFJNK3Ny4OsQ5X36M+r=Y6LQ1eYwy0pg(a)mail.gmail.com>

1 0

Re: [SR-Users] Error or Bug: [t_msgbuilder.c:490]: build_local_reparse(): ERROR: build_local_reparse: HDR_EOH_T was not found
by Sid Mason 07 Feb '16

07 Feb '16

Hi, Thanks a lot for the reply, It means a lot to as I am stuck on this issue. I'm not sure if attachment works here or not. I will paste the log in paste bin: http://pastebin.com/0w6a8iTB Regards <CAEmfXJ4KeqaPD_7k9pgFJNK3Ny4OsQ5X36M+r=Y6LQ1eYwy0pg(a)mail.gmail.com>

2 1

Extending the default 1800 seconds of session expiration
by kssyrr 06 Feb '16

06 Feb '16

2 2

why msg_apply_changes() from REQUEST_ROUTE only
by smititelu 05 Feb '16

05 Feb '16

Hello, I don't understand why msg_apply changes() can be called from REQUEST_ROUTE *only*. Can someone explain? Can this be extended to *FAILURE_ROUTES *also? Basically I need to call this from FAILURE_ROUTES in order to avoid double SDP for rtpengine offers. Searching in the mailing list I could see that there could be other means of handling this like keeping and reading the SDP from an avp(write/read_sdp_pv), but I find the above method pretty straightforward. Thanks, Stefan

3 4

Error or Bug: [t_msgbuilder.c:490]: build_local_reparse(): ERROR: build_local_reparse: HDR_EOH_T was not found
by Sid Mason 05 Feb '16

05 Feb '16

Hi We are receiving error on CANCEL request from client that is supposed to be relayed to an Asterisk Server: [t_msgbuilder.c:490]: build_local_reparse(): ERROR: build_local_reparse: HDR_EOH_T was not found Any help on this is much appreciated. below is the log: Regards Sid 0(6351) DEBUG: <core> [parser/msg_parser.c:606]: parse_msg(): SIP Request: 0(6351) DEBUG: <core> [parser/msg_parser.c:608]: parse_msg(): method: <CANCEL> 0(6351) DEBUG: <core> [parser/msg_parser.c:610]: parse_msg(): uri: < sip:0031XXXXXXXXX@voip03nl.voipsystem.com> 0(6351) DEBUG: <core> [parser/msg_parser.c:612]: parse_msg(): version: <SIP/2.0> 0(6351) DEBUG: <core> [parser/parse_via.c:1254]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK-d8754z-ff05107e83283f5e-1---d8754z->; state=6 0(6351) DEBUG: <core> [parser/parse_via.c:1254]: parse_via_param(): Found param type 235, <rport> = <n/a>; state=17 0(6351) DEBUG: <core> [parser/parse_via.c:2642]: parse_via(): end of header reached, state=5 0(6351) DEBUG: <core> [parser/msg_parser.c:496]: parse_headers(): parse_headers: Via found, flags=2 0(6351) DEBUG: <core> [parser/msg_parser.c:498]: parse_headers(): parse_headers: this is the first via 0(6351) DEBUG: <core> [receive.c:134]: receive_msg(): After parse_msg... 0(6351) DEBUG: <core> [parser/parse_addr_spec.c:894]: parse_addr_spec(): end of header reached, state=10 0(6351) DEBUG: <core> [parser/msg_parser.c:173]: get_hdr_field(): DEBUG: get_hdr_field: <To> [59]; uri=[sip:0031XXXXXXXXX@voip03nl.voipsystem.com] 0(6351) DEBUG: <core> [parser/msg_parser.c:175]: get_hdr_field(): DEBUG: to body ["0031XXXXXXXXX"<sip:0031XXXXXXXXX@voip03nl.voipsystem.com> 0(6351) DEBUG: <core> [parser/msg_parser.c:153]: get_hdr_field(): get_hdr_field: cseq <CSeq>: <2> <CANCEL> 0(6351) DEBUG: <core> [parser/msg_parser.c:187]: get_hdr_field(): DEBUG: get_hdr_body : content_length=0 0(6351) DEBUG: <core> [parser/msg_parser.c:89]: get_hdr_field(): found end of header {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) DEBUG: <core> [receive.c:177]: receive_msg(): preparing to run routing scripts... {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/etc/kamailio/kamailio.cfg] l=614 a=5 n=route {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/etc/kamailio/kamailio.cfg] l=734 a=25 n=xlog {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) ERROR: <script>: The IP is : 90.152.66.68 and uri {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/etc/kamailio/kamailio.cfg] l=745 a=16 n=if {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/etc/kamailio/kamailio.cfg] l=742 a=25 n=xlog {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) ERROR: <script>: Match the 0/24 IP -- the message was sent from REGUS OFFICE! {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/etc/kamailio/kamailio.cfg] l=767 a=16 n=if {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) DEBUG: <core> [socket_info.c:563]: grep_sock_info(): checking if host==us: 12==9 && [90.152.66.68] == [10.1.4.23] {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) DEBUG: <core> [socket_info.c:582]: grep_sock_info(): checking advertise if host==us: 12==22 && [90.152.66.68] == [voip03nl.voipsystem.com] {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) DEBUG: <core> [socket_info.c:563]: grep_sock_info(): checking if host==us: 12==9 && [90.152.66.68] == [10.1.4.23] {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) DEBUG: <core> [socket_info.c:582]: grep_sock_info(): checking advertise if host==us: 12==22 && [90.152.66.68] == [voip03nl.voipsystem.com] {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) DEBUG: <core> [forward.c:412]: check_self(): host != me {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/etc/kamailio/kamailio.cfg] l=761 a=16 n=if {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/etc/kamailio/kamailio.cfg] l=766 a=16 n=if {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/etc/kamailio/kamailio.cfg] l=761 a=24 n=pike_check_req {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) DEBUG: pike [ip_tree.c:286]: mark_node(): search on branch 90 (top=0x7f24a45a4688) {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) DEBUG: pike [ip_tree.c:299]: mark_node(): only first 4 were matched! {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) DEBUG: pike [pike_funcs.c:93]: pike_check_req(): src IP [90.152.66.68],node=0x7f24a45a4d78; hits=[0,0],[0,1] node_flags=6 func_flags=0 {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) DEBUG: pike [timer.c:46]: remove_from_timer(): 0x7f24a45a4d88 from 0x7f24a4579500(0x7f24a45a4d88,0x7f24a45a4d88) {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) DEBUG: pike [timer.c:33]: append_to_timer(): 0x7f24a45a4d88 in 0x7f24a4579500(0x7f24a4579500,0x7f24a4579500) {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/etc/kamailio/kamailio.cfg] l=782 a=16 n=if {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/etc/kamailio/kamailio.cfg] l=807 a=16 n=if {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/etc/kamailio/kamailio.cfg] l=782 a=25 n=mf_process_maxfwd_header {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) DEBUG: maxfwd [mf_funcs.c:66]: is_maxfwd_present(): max_forwards header not found! {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/etc/kamailio/kamailio.cfg] l=807 a=26 n=xlog {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) DEBUG: <script>: [REQINIT] (Method is CANCEL) r-uri ( sip:0031XXXXXXXXX@voip03nl.voipsystem.com) {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/etc/kamailio/kamailio.cfg] l=825 a=16 n=if {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/etc/kamailio/kamailio.cfg] l=811 a=25 n=is_method {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/etc/kamailio/kamailio.cfg] l=829 a=16 n=if {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/etc/kamailio/kamailio.cfg] l=825 a=26 n=sanity_check {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) DEBUG: <core> [parser/parse_addr_spec.c:172]: parse_to_param(): DEBUG: add_param: tag=0d61d760 {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) DEBUG: <core> [parser/parse_addr_spec.c:894]: parse_addr_spec(): end of header reached, state=29 {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) DEBUG: sanity [mod_sanity.c:248]: w_sanity_check(): sanity checks result: 1 {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/etc/kamailio/kamailio.cfg] l=617 a=5 n=route {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) ERROR: *** cfgtrace:request_route=[NATDETECT] c=[/etc/kamailio/kamailio.cfg] l=1104 a=55 n=force_rport {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) ERROR: *** cfgtrace:request_route=[NATDETECT] c=[/etc/kamailio/kamailio.cfg] l=1149 a=16 n=if {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) ERROR: *** cfgtrace:request_route=[NATDETECT] c=[/etc/kamailio/kamailio.cfg] l=1118 a=25 n=nat_uac_test {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) ERROR: *** cfgtrace:request_route=[NATDETECT] c=[/etc/kamailio/kamailio.cfg] l=1143 a=16 n=if {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) ERROR: *** cfgtrace:request_route=[NATDETECT] c=[/etc/kamailio/kamailio.cfg] l=1119 a=25 n=is_method {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) ERROR: *** cfgtrace:request_route=[NATDETECT] c=[/etc/kamailio/kamailio.cfg] l=1125 a=25 n=xlog {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) ERROR: <script>: [NATDETECT]: method is CANCEL{1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) ERROR: *** cfgtrace:request_r oute=[NATDETECT] c=[/etc/kamailio/kamailio.cfg] l=1142 a=16 n=if {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) ERROR: *** cfgtrace:request_route=[NATDETECT] c=[/etc/kamailio/kamailio.cfg] l=1132 a=24 n=is_first_hop {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) DEBUG: siputils [sipops.c:159]: w_is_first_hop(): no 2nd via found - first hop {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) ERROR: *** cfgtrace:request_route=[NATDETECT] c=[/etc/kamailio/kamailio.cfg] l=1138 a=26 n=xlog {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) DEBUG: <script>: [NATDETECT]: (is_first_hop() is true) set_contact_alias() Adds an “;alias=ip~port~transport” parameter to the contact URI{1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) ERROR: *** cfgtrace:request_route=[NATDETECT] c=[/etc/kamailio/kamailio.cfg] l=1139 a=24 n=set_contact_alias {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) ERROR: *** cfgtrace:request_route=[NATDETECT] c=[/etc/kamailio/kamailio.cfg] l=1146 a=39 n=setflag {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) ERROR: *** cfgtrace:request_route=[NATDETECT] c=[/etc/kamailio/kamailio.cfg] l=1151 a=26 n=xlog {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) DEBUG: <script>: [NATDETECT]: NATDETECT is done{1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) ERROR: *** cfgtrace:request_route=[NATDETECT] c=[/etc/kamailio/kamailio.cfg] l=1152 a=2 n=return {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/etc/kamailio/kamailio.cfg] l=620 a=5 n=route {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) ERROR: *** cfgtrace:request_route=[WITHINDLG] c=[/etc/kamailio/kamailio.cfg] l=842 a=16 n=if {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) ERROR: *** cfgtrace:request_route=[WITHINDLG] c=[/etc/kamailio/kamailio.cfg] l=837 a=24 n=has_totag {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) DEBUG: siputils [checks.c:97]: has_totag(): no totag {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) ERROR: *** cfgtrace:request_route=[WITHINDLG] c=[/etc/kamailio/kamailio.cfg] l=838 a=26 n=xlog {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) DEBUG: <script>: [WITHINDLG]: Has_totag() is false; Return 1;{1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) ERROR: *** cfgtrace:request_route=[WITHINDLG] c=[/etc/kamailio/kamailio.cfg] l=839 a=2 n=return {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/etc/kamailio/kamailio.cfg] l=632 a=16 n=if {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/etc/kamailio/kamailio.cfg] l=625 a=25 n=is_method {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/etc/kamailio/kamailio.cfg] l=629 a=16 n=if {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/etc/kamailio/kamailio.cfg] l=627 a=24 n=t_check_trans {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) DEBUG: tm [t_lookup.c:701]: t_lookupOriginalT(): DEBUG: t_lookupOriginalT: searching on hash entry 38426 {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) DEBUG: tm [t_lookup.c:409]: matching_3261(): DEBUG: RFC3261 transaction matched, tid=-d8754z-ff05107e83283f5e-1---d8754z- {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) DEBUG: tm [t_lookup.c:798]: t_lookupOriginalT(): DEBUG: t_lookupOriginalT: canceled transaction found (0x7f24a45a4fa8)! {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) DEBUG: tm [t_lookup.c:801]: t_lookupOriginalT(): DEBUG: t_lookupOriginalT completed {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) DEBUG: tm [tm.c:1043]: w_t_lookup_cancel(): lookup_original: t_lookupOriginalT returned: 0x7f24a45a4fa8 {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/etc/kamailio/kamailio.cfg] l=628 a=24 n=t_relay {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) DEBUG: tm [t_lookup.c:1312]: t_newtran(): DEBUG: t_newtran: msg id=12 , global msg id=11 , T on entrance=0xffffffffffffffff {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) DEBUG: tm [t_lookup.c:466]: t_lookup_request(): t_lookup_request: start searching: hash=38426, isACK=0 {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) DEBUG: tm [t_lookup.c:424]: matching_3261(): DEBUG: RFC3261 transaction matching failed {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) DEBUG: tm [t_lookup.c:648]: t_lookup_request(): DEBUG: t_lookup_request: no transaction found {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) DEBUG: tm [t_hooks.c:358]: run_reqin_callbacks_internal(): DBG: trans=0x7f24a45a9098, callback type 1, id 0 entered {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) DEBUG: <core> [md5utils.c:67]: MD5StringArray(): MD5 calculated: a4351b7a4b00df7d36f1c82b3d87c603 {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) DEBUG: tm [t_lookup.c:701]: t_lookupOriginalT(): DEBUG: t_lookupOriginalT: searching on hash entry 38426 {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) DEBUG: tm [t_lookup.c:409]: matching_3261(): DEBUG: RFC3261 transaction matched, tid=-d8754z-ff05107e83283f5e-1---d8754z- {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) DEBUG: tm [t_lookup.c:798]: t_lookupOriginalT(): DEBUG: t_lookupOriginalT: canceled transaction found (0x7f24a45a4fa8)! {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) DEBUG: tm [t_lookup.c:801]: t_lookupOriginalT(): DEBUG: t_lookupOriginalT completed {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) ERROR: tm [t_msgbuilder.c:490]: build_local_reparse(): ERROR: build_local_reparse: HDR_EOH_T was not found {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) ERROR: tm [t_msgbuilder.c:494]: build_local_reparse(): ERROR: build_local_reparse: cannot build CANCEL request {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) ERROR: tm [t_cancel.c:310]: cancel_branch(): ERROR: attempt to build a CANCEL failed {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) ERROR: tm [t_fwd.c:1388]: e2e_cancel(): ERROR: cancel error {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) DEBUG: <core> [mem/shm_mem.c:101]: _shm_resize(): WARNING:vqm_resize: resize(0) called {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) DEBUG: tm [t_reply.c:1591]: cleanup_uac_timers(): DEBUG: cleanup_uac_timers: RETR/FR timers reset {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) DEBUG: tm [t_reply.c:647]: _reply_light(): DEBUG: reply sent out. buf=0x7f24aee459c8: SIP/2.0 500 cancel e..., shmem=0x7f24a45ab880: SIP/2.0 500 cancel e {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) DEBUG: tm [t_reply.c:657]: _reply_light(): DEBUG: _reply_light: finished {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) DEBUG: tm [t_funcs.c:362]: t_relay_to(): SER: new transaction fwd'ed {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/etc/kamailio/kamailio.cfg] l=629 a=2 n=exit {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) DEBUG: <core> [usr_avp.c:631]: destroy_avp_list(): destroying list (nil) {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) DEBUG: <core> [usr_avp.c:631]: destroy_avp_list(): destroying list (nil) {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) DEBUG: <core> [usr_avp.c:631]: destroy_avp_list(): destroying list (nil) {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) DEBUG: <core> [usr_avp.c:631]: destroy_avp_list(): destroying list (nil) {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) DEBUG: <core> [usr_avp.c:631]: destroy_avp_list(): destroying list (nil) {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) DEBUG: <core> [usr_avp.c:631]: destroy_avp_list(): destroying list (nil) {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) DEBUG: <core> [xavp.c:446]: xavp_destroy_list(): destroying xavp list (nil) {1 2 CANCEL MmU5NDBhMTcxYzc1YWNmZGJiNGQ3NmU4NjU2M2I2NTc.} 0(6351) DEBUG: <core> [receive.c:278]: receive_msg(): cleaning up

2 1

Kamailio strange behaviour
by Gholamreza Sabery 05 Feb '16

05 Feb '16

I am using Kamailio 4.1.x with NAT enabled. When I want to REGISTER using my desktop client which is Linphone everything is find with any username. But on Android some users can REGISTER and someusers can not. I captured SIP packets and the only difference between packets on Android devices and packets on Desktop client is that on my desktop "received" and "rport" are not added by Kamailio in response messages. Any idea on what is happening here?

2 3

Extending the default 1800 seconds expiration of presence
by kssyrr 04 Feb '16

04 Feb '16

1 0

Sip routing
by David Bakker 04 Feb '16

04 Feb '16

I want to be able to route sip invites from my on prem skype to either an on prem jabber server or out to office 365. It depends on who the recipient is. I am sure Kamailio can do this but I am absolutely at a loss on where to start. I have tried a few different configs but I basically have to break our office communication to test. I am looking for some guidance on what to do. Thanks in advance!

2 1

Kamailio v4.4 - development frozen
by Daniel-Constantin Mierla 04 Feb '16

04 Feb '16

Hello, development (master) branch of Kamailio enters now in pre-release phase for version 4.4.0. Therefore, no new feature should be pushed to master until we create a dedicated branch for 4.4 (expected to be in about 4 weeks or so). If in doubt to push or not a commit to master, push it first on a personal branch (or attach to an email) and discuss it on sr-dev. The new modules can be a bit more dynamic if there is need to get them to the right shape (e.g., like decision to rename functions, parameters or adjust database structure). We hope to get many people involved in testing, to reach a stable state before releasing 4.4.0. If you want to get involved and need assistance, don't hesitate to write to mailing lists. Based on my quick review, there were no major changes to old core components, meaning we should have preserved quite a good level of stability overall. Help with updating the wiki page for migration from 4.3 to 4.4 as well as what is new in 4.4 is very appreciated. I will post updates about them very soon. Many thanks to everyone involved in development of 4.4 and the early testers that played with master branch so far. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 18-20, 2016 Berlin, Germany - http://www.kamailioworld.com

1 0

Kamailio Cluster
by Gholamreza Sabery 03 Feb '16

03 Feb '16

I am trying to create a cluster of Kamailio servers. I want users that are registered on one server to be able to contact and call users on other servers. How can I implement this in Kamailio? PS: My servers are in a private network and I do not have DNS.

10 21

rtpengine port allocation & recycling
by Alex Balashov 03 Feb '16

03 Feb '16

Hello, I have a high-volume installation with 100% RTP relay using rtpengine. We run rtpengine with a high and low port range of 8000 - 65000. Generally the peak concurrent channel count is around 2000-4000 ports, so in theory, so we should have enough ports for all calls currently in process. In practice, however, we get a nearly constant stream of these errors from Kamailio: Feb 2 15:52:58 sip /usr/local/sbin/kamailio[16438]: ERROR: rtpengine [rtpengine.c:1541]: rtpp_function_call(): proxy replied with error: Error rewriting SDP When I trace this back to rtpengine logs on the server, I see: --- Feb 2 15:55:47 rtphost rtpengine[14710]: [FSRIGZPJGXZWPNO.52364.907376806] Received command 'offer' from 215.111.222.101:54100 Feb 2 15:55:47 rtphost rtpengine[14710]: [FSRIGZPJGXZWPNO.52364.907376806] Creating new call Feb 2 15:55:47 rtphost rtpengine[14710]: [FSRIGZPJGXZWPNO.52364.907376806] Failed to get 2 consecutive UDP ports for relay Feb 2 15:55:47 rtphost rtpengine[14710]: [FSRIGZPJGXZWPNO.52364.907376806] Error allocating media ports Feb 2 15:55:47 rtphost rtpengine[14710]: [FSRIGZPJGXZWPNO.52364.907376806] Protocol error in packet from 215.111.222.101:54100: Error rewriting SDP [d3:sdp285:v=0#015#012o=- 1631132006999214513 4263823554000404972 IN IP4 45.201.202.203#015#012s=VOS2009#015#012c=IN IP4 45.201.202.203#015#012t=0 0#015#012m=audio 29034 RTP/AVP 18 8 0 101#015#012a=rtpmap:18 G729/8000#015#012a=fmtp:18 annexb=no#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-15#015#0123:ICE6:remove7:replacel6:origin18:session-connectione7:call-id31:FSRIGZPJGXZWPNO.52364.90737680613:received-froml3:IP415:215.111.222 ... Feb 2 15:55:47 rtphost rtpengine[14710]: [FSRIGZPJGXZWPNO.52364.907376806] ... 5.125e8:from-tag32:709abb985b675e1497597a5bd44ca2d67:command5:offere] --- This is despite nearly no rtpengine engagement: [root@csrprtpus1 log]# cat /proc/rtpengine/0/status Refcount: 1 Control PID: 14709 Targets: 0 Buckets: 0 As far as I can see, the cause is that rtpengine is not deallocating ports: --- [root@rtphost log]# lsof -n | grep -i rtp rtpengine 14710 root *542u IPv6 1082109949 0t0 UDP *:39910 rtpengine 14710 root *543u IPv6 1082109950 0t0 UDP *:39911 rtpengine 14710 root *544u IPv6 1082218768 0t0 UDP *:41469 rtpengine 14710 root *545u IPv6 1082058934 0t0 UDP *:46424 rtpengine 14710 root *546u IPv6 1082366799 0t0 UDP *:22915 rtpengine 14710 root *547u IPv6 1082288397 0t0 UDP *:57245 rtpengine 14710 root *548u IPv6 1082428020 0t0 UDP *:24311 rtpengine 14710 root *549u IPv6 1082366800 0t0 UDP *:23266 rtpengine 14710 root *550u IPv6 1082798144 0t0 UDP *:22743 rtpengine 14710 root *551u IPv6 1082605782 0t0 UDP *:47845 rtpengine 14710 root *552u IPv6 1082062515 0t0 UDP *:52766 rtpengine 14710 root *553u IPv6 1082648709 0t0 UDP *:58266 rtpengine 14710 root *554u IPv6 1082122218 0t0 UDP *:54359 ... --- There are _lots_ of these: [root@rtphost log]# lsof -n | grep -i rtpengine | grep 'UDP \*:' | wc -l 56964 So, I suppose most of these just aren't being freed up. These are short-duration calls, and the vast majority of them are CANCEL'd at various stages of completion. My approach to invocation of rtpengine in the config is: 1) rtpengine_offer() in branch_route[] on outbound call: set_rtpengine_set("1"); rtpengine_offer("replace-origin replace-session-connection ICE=remove"); add_rr_param(";proxy_media=yes"); setflag(PROXY_MEDIA); This can happen multiple times, as many branches may be attempted before a gateway is found to take the call. 2) Symmetrical rtpengine_answer() in onreply_route: if(isflagset(PROXY_MEDIA) && has_body("application/sdp") && $var(origin_b2bua) == 0) { set_rtpengine_set("1"); rtpengine_answer("replace-origin replace-session-connection ICE=remove"); } 3) rtpengine_delete() in event of CANCEL: if(is_method("CANCEL")) { # Kill RTP stream if this call was using rtpengine. set_rtpengine_set("1"); rtpengine_delete(); # Implicitly forward the CANCEL and drop out of script # if the corresponding INVITE transaction exists. if(!t_relay_cancel()) { # If we are still here, the INVITE transaction # was found but an error occurred. sl_send_reply("500", "Internal Server Error"); exit; } # If we are still here, this is also bad because it means # the corresponding INVITE transaction no longer exists. xlog("L_INFO", 'action=R-MAIN-REQUEST | ret=CANCEL_NO_TRANS_MATCH'); exit; 4) rtpengine_delete() in case of BYE: if(has_totag()) { if(loose_route()) { if(is_method("BYE")) { if(check_route_param("proxy_media=yes")) { set_rtpengine_set("1"); rtpengine_delete(); } } else if(is_method("INVITE")) { # update rtpengine_offer() etc. } ... } else { ... } } Any obvious problem here? Thanks! -- Alex -- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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