sr-users March 2016

sr-users@lists.kamailio.org

88 participants
127 discussions

by Igor Olhovskiy

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

8 years, 7 months

HOW send 4XX to the caller when the callee response to bye is 481 Call leg/transaction does not exist

by Humberto H

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

8 years, 7 months

How can I send 404 to the caller instead of BYE?

by Humberto H

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; }

8 years, 7 months

stateful messages in a dialog

by Al S

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

8 years, 7 months

Record-Route header

by jignesh gandhi

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

8 years, 7 months

Re: [SR-Users] Proxy configuration

by Jignesh Gandhi

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 >*****************************************

8 years, 7 months

timeout function

by Al S

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

8 years, 7 months

TCP connection select problem

by 张顺通

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?

8 years, 7 months

db_cassandra status and plans

by Andreas Granig

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

8 years, 7 months

acc_cdrs empty

by Nelson Migliaro

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.

8 years, 7 months

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sr-users March 2016