- sr-users - kamailio.org

Wrong Call-ID and tags in response to OPTIONS
by Jean-Marie Baran 10 Aug '15

10 Aug '15

Hi, I am trying to get Kamailio working with a third party SIP provider. This provider pings Kamailio with OPTIONS messages, but Kamailio does not answer correctly: the Call-ID and the tags change where they should not. Here is an example: Third party SIP server: OPTIONS sip:false.sip.provider:5060 SIP/2.0 Via: SIP/2.0/UDP 197.17.144.4:5060;branch=z9hG4bK8e45ac1463a From: <sip:197.17.144.4>;tag=1669836943 To: <sip:false.sip.provider> Date: Fri, 07 Aug 2015 12:34:37 GMT Call-ID: a012e080-5c41a5ce-8e5-38512c6(a)197.17.144.4 User-Agent: Cisco-CUCM10.5 CSeq: 101 OPTIONS Contact: <sip:197.17.144.4:5060> Max-Forwards: 0 Content-Length: 0 Kamailio's answer: SIP/2.0 200 OK Via: SIP/2.0/UDP 197.17.144.4:5060;branch=z9hG4bK93e5f072f5f;received=174.37.217.1 From: <sip:197.17.144.4>;tag=298911055 To: <sip:false.sip.provider>;tag=b27e1a1d33761e85846fc98f5f3a7e58.8c24 Call-ID: 2defcb00-5c41aa17-93f-38512c6(a)197.17.144.4 CSeq: 101 OPTIONS Server: kamailio (4.3.1 (x86_64/linux)) Content-Length: 0 You can see that the Call-ID change from "a012e080-5c41a5ce-8e5-38512c6" to "2defcb00-5c41aa17-93f-38512c6". Likewise, the From tag change ("1669836943" to "298911055"). What could be the reason for this ? Moreover I think Kamailio should add the field Allow (with INVITE,ACK,BYE,CANCEL,INFO,OPTIONS,REFER,SUBSCRIBE,NOTIFY), but it does not. I should mention that I had to modify /kamailio.cfg/ to have Kamailio answer to OPTIONS messages (I added *the code in bold*): if (*!is_method("OPTIONS") && *!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS") && uri==myself && $rU==$null) { *#* sl_send_reply("200","Keepalive"); *options_reply();* exit; } Maybe this hack replaces an expected behavior that I failed to see/configure ? Thanks, Jean-Marie.

3 3

Kamailio 4.3 and XCAP
by Safdar Khan 10 Aug '15

10 Aug '15

Hello sir/mam How can i configure xcap with kamailio 4.3. I am finding difficulties in routing logic of xcap_server module amd xcap_client module. Will you please share proper documentation of these two technology together.

1 0

LCR Routing - Cost based routing
by Arun Kumar 09 Aug '15

09 Aug '15

Hi is there any way to achive following usecase using carrier route / LCR / droude module i have 10 different termination gateways, each gateways will have more than 10000 destinations and each gateways will have different rate, for example 91 Destination ( there will be 10000 entries like 91 ) gw1 gw2 gw3 gw4 gw5 0.001 0.007 0.003 0.002 0.003 now need to logic to load the these destination with cost and sort gateways in order of longest match of dialled number,cost, priority for the dialled destination, which will not use too much CPU and memory Regards Arun

4 9

Kamailio crash when imc_list_rooms
by SamyGo 09 Aug '15

09 Aug '15

Hi, I'm working with IMC module and noticed that when I execute fifo command "imc_list_rooms" it crashes. [root@kamailio75 ]# kamctl fifo imc_list_rooms my Kamailio version is 4.2.1 Right now the way IMC module works is that a SIP UA has to send a chat room command to create a new room, get participants added etc. If I try to add users directly from DB the kamailio don't recognize the room or participants unless restart. Would it be possible to reload the DB tables for IMC via MI command, a big plus if we can do room creation, user addition via MI as well ! Thanks, Sammy

1 0

acc Module missing BYE in database
by Dirk Teurlings - SIGNET B.V. 07 Aug '15

07 Aug '15

Hi again, Running kamailio 4.2.5 with the acc module to do accounting in a MySQL database. We are now running into the issue that sometimes there is no BYE line when there is an INVITE. Digging into this I noticed that the dialog:end event does get triggered, and that CNXCC consumes the call properly. So the BYE message does get sent, but acc doesn't seem to pick it up some of the times. It occurs around maybe once every 100 calls. Anybody got any ideas about this? -- Cheers, Dirk Teurlings

2 3

Kamailio example trace needed UAR/UAA
by Jan Devai 07 Aug '15

07 Aug '15

Hi Experts, we have trouble with the last SIP Registration from I-CSCF to S-CSCF, see below: *The flow is as follows:* UE HSS P-CSCF I-CSCF S-CSCF -----------Sip Register-----------> -------Sip Register-------> <--------------------------UAR--------------- --------------------------UAA---------------> ----SIP Register---> Almost a week ago, I've open a thread about this issue called "Error forwarding to SCSCF", and so far we did not get any further with it :-( I was wondering does somebody have similar Signaling that he could share with us? It could be either with an external or OpenHSS, does not matter at this stage. It would definitely move us forward with at least the signaling comparison as a start point. Maybe the UAA answer and anything happening before is wrong. Any reference, similar working trace would be greatly appreciated. Thank you in advance Jan

2 1

About Acc_radius rad_sttaus Function
by Yasin CANER 07 Aug '15

07 Aug '15

3 4

SDP IPv4 has been concatenate - no audio
by Loic Chabert 07 Aug '15

07 Aug '15

Hello everyone, I'm trying put kamailio in front of asterisk server farm. Fow now, 2 asterisk servers are running and i'm trying to make some basic calls between two UACc. All asterisk servers has been ofuscaded from public internet using 10.189.122.0/24 network. All trafic must be passed throught asterisk so RTPproxy is used to (and used for rtp bridging). Kamailio and rtpproxy is running with public IP address, and private ip address (mhomed=1) But a wired thing append on my SDP body: c line have two rtpproxy public addresses concatenate (see my capture attached). Any reason for this ? Only invite method from my asterisk contains 2 publics IP addresses concatenated. Does it mean than rtp_manage as been executed twice ? Thanks in advance. Regards.

3 6

Problem adding json module
by Joao Alves 07 Aug '15

07 Aug '15

Hi, Have just installed the kamailio 4.3 (from github) and was trying to include the additional modules that I require. I've edited the modules.lst and changed the extra modules to compile "include_modules= db_mysql utils json". During compilation (using "make all") getting the following warnings in relation with the json module. CC (gcc) [M json.so] json_funcs.o json_funcs.c: In function 'json_get_field': json_funcs.c:58:2: warning: implicit declaration of function 'is_error' [-Wimplicit-function-declaration] json_funcs.c:64:2: warning: 'json_object_object_get' is deprecated (declared at /usr/local/include/json-c/json_object.h:303) [-Wdeprecated-declarations] And later, when running the config file: ERROR: <core> [sr_module.c:574]: load_module(): could not open module </usr/local/lib64/kamailio/modules/json.so>: libjson-c.so.3: cannot open shared object file: No such file or directory 0(2344) : <core> [cfg.y:3432]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 195, column 12-20: failed to load module ERROR: bad config file (1 errors) I have already installed json-c modules and confirmed this installation by doing: root@ID14337:/usr/local/kamailio-4.3/kamailio# pkg-config --cflags json-c -I/usr/local/include/json-c root@ID14337:/usr/local/kamailio-4.3/kamailio# pkg-config --libs json-c -L/usr/local/lib -ljson-c root@ID14337:/usr/local/include/json-c# ls arraylist.h json.h json_inttypes.h json_object_private.h linkhash.h bits.h json_c_version.h json_object.h json_tokener.h printbuf.h debug.h json_config.h json_object_iterator.h json_util.h random_seed.h root@ID14337:/usr/local/include/json-c# cd ../../lib root@ID14337:/usr/local/lib# ls libjson-c.a libjson-c.la libjson-c.so libjson-c.so.3 libjson-c.so.3.0.0 pkgconfig python2.6 python2.7 root@ID14337:/usr/local/lib# So any ideas of what could be causing this to fail? Thanks Joao Alves This message and the information contained herein is proprietary and confidential and subject to the Amdocs policy statement, you may review at http://www.amdocs.com/email_disclaimer.asp

2 6

Kamailio w/ external routing server
by Eric Babcock 07 Aug '15

07 Aug '15

Hey Everyone, I've been banging my head against the desk for days trying to get this working, and I'm sure it's something simple that I'm not doing, but maybe someone else can add some input. Here is the best way to describe my call flow. Customer -> Kamailio -> Routing Server (302) -> Kamailio -> Vendors (from the 302 header) I have tried to configure dispatch, but that hasn't worked out for me. My routing server does the authentication, LRN lookup, and provides the vendor routes in the sip header. So when I pass the call to the routing server, it has to contain the customer information. This setup has to be stateful so I can write CDRs for billing to MySQL. The reason I tried using dispatch, is because I have multiple routing servers I would like to load balance across or even failover if one goes down. Can anyone help me out, or at least help point me in the right direction? Thanks! -Eric -- View this message in context: http://sip-router.1086192.n5.nabble.com/Kamailio-w-external-routing-server-… Sent from the Users mailing list archive at Nabble.com.

3 4

Errors when starting Kamailio with Websocket Module
by Babu Raja 07 Aug '15

07 Aug '15

Hello, I incuded the module for websocket into my Kamailio 4.3 on my Ubuntu 12.04 server environement. For installing the packages and websocket I followed the instructions from the following link: http://www.kamailio.org/wiki/packages/debs http://nil.uniza.sk/sip/kamailio/configuring-kamailio-4x-websocket After starting the Kamailio I got 26 errors like: can not find command "handle_ruri_alias" or Syntax errors for: route[REGISTRAR] => square brackets.. ($du == "") => parameter "==" unknown.. the whole error log I will paste later.. Can anyone help me?

3 4

RLS Notify sent to wrong transport
by M S 07 Aug '15

07 Aug '15

Hi, I have a Kamailio v4.3.0 with SVN Rev. c6aa95 running an RLS based presence setup. Everything works fine when end-user has UDP transport, however, if user has TCP or TLS transport then i aggregated NOTIFY sent by RLS gives send error, -- WARNING: <core> [forward.c:231]: get_send_socket2(): protocol/port mismatch (forced tls:X.X.X.X:5061, to udp:Y.Y.Y.Y:12345) -- (X.X.X.X is server IP and Y.Y.Y.Y is end user IP). Not sure if it is configuration issue or some bug, so posting it to both mailing list. Please help. Thank you.

2 8

Kamailio 4.3.1 and Debian 7
by Mathieu Boyer 07 Aug '15

07 Aug '15

Hello, i've got big issues since upgrade of Debian packages (debian repository http://deb.kamailio.org/kamailio wheezy)<http://deb.kamailio.org/kamailio> on 4.3.1. Started with rr.so can't load without outbound.so, location.so find version mismatch (6 vs 8) etc ... Have you got a quick workaround in order to make it run properly ? dpkg -l |grep kamailio ii kamailio 4.3.1+wheezy amd64 very fast and configurable SIP proxy ii kamailio-extra-modules:amd64 4.3.1+wheezy amd64 extra modules for Kamailio ii kamailio-mysql-modules:amd64 4.3.1+wheezy amd64 MySQL database connectivity module for Kamailio ii kamailio-outbound-modules:amd64 4.3.1+wheezy amd64 Outbound module for Kamailio ii kamailio-utils-modules:amd64 4.3.1+wheezy amd64 Provides a set utility functions for Kamailio Error logs : Aug 6 21:24:18 frontsip01 /usr/sbin/kamailio[4083]: ERROR: dispatcher [dispatcher.c:768]: ds_warn_fixup(): failover functions used, but required AVP parameters are NULL -- feature disabled Aug 6 21:24:18 frontsip01 /usr/sbin/kamailio[4085]: ERROR: db_mysql [km_dbase.c:121]: db_mysql_submit_query(): driver error on query: Unknown column 'server_id' in 'field list' Aug 6 21:24:18 frontsip01 /usr/sbin/kamailio[4085]: ERROR: <core> [db_query.c:132]: db_do_query_internal(): error while submitting query Aug 6 21:24:18 frontsip01 /usr/sbin/kamailio[4085]: ERROR: usrloc [udomain.c:424]: preload_udomain(): db_query (1) failed Aug 6 21:24:18 frontsip01 /usr/sbin/kamailio[4085]: ERROR: usrloc [ul_mod.c:439]: child_init(): child(1): failed to preload domain 'location' Aug 6 21:24:18 frontsip01 /usr/sbin/kamailio[4085]: ERROR: <core> [sr_module.c:900]: init_mod_child(): Error while initializing module usrloc (/usr/lib/x86_64-linux-gnu/kamailio/modules/usrloc.so) Aug 6 21:24:18 frontsip01 /usr/sbin/kamailio[4085]: ERROR: <core> [pt.c:329]: fork_process(): init_child failed for process 1, pid 4085, "udp receiver child=0 sock=XXXX:5063"? Thank you for your help BR Mathieu

3 2

Rtpengine/rtpproxy and zrtp
by Alexandru Covalschi 07 Aug '15

07 Aug '15

Sorry if writing to wrong mailing list, I am very limited to traffic now amd don't know if there is any for rtpproxy/rtpengine. My question is - can they support ZRTP at least in pass-through mode? Will rtpengine fail on trying to recognize unknown SDP fields?

2 2

Kamailio v4.2.6 Released
by Daniel-Constantin Mierla 06 Aug '15

06 Aug '15

Hello, Kamailio SIP Server v4.2.6 stable release is out. This is a maintenance release of the previous stable branch, 4.2, that includes fixes since release of v4.2.5. There is no change to database schema or configuration language structure that you have to do on installations of v4.2.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.2.6 (or to 4.3.x series). For more details about version 4.2.6 (including links and guidelines to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2015/07/kamailio-v4-2-6-released/ RPM, Debian/Ubuntu packages will be available soon as well. Note: the latest stable branch is 4.3, at this moment with its latest release v4.3.1. See more details about it at: * http://www.kamailio.org/w/kamailio-v4-3-0-release-notes/ Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/miconda - http://www.linkedin.com/in/miconda

2 3

Kamailio Redis support
by Solutions Solutions 06 Aug '15

06 Aug '15

Hi All, i want to replace MySQL with redis database in kamalio,but in kamalio supports transactions with MySQL ,ie saving data in MySql DB.but while trying to run kamailio with redis ,transaction are not happening,is there any module that i have to enable in kamalio for No SQL databases transaction support.

2 1

Re: [SR-Users] Remote SIP Registration
by Keith 05 Aug '15

05 Aug '15

Hi, I have a carrier who require registration, I have looked at the uac module but can't see how to do this. Anyone point me in the right direction? Thanks Keith

2 10

Difference between corex module's 'alias_subdomains' and 'alias' core parameter
by Iwan Aucamp 05 Aug '15

05 Aug '15

>From the descriptions these seem similar: corex alias_subdomains ( http://kamailio.org/docs/modules/devel/modules/corex.html#idp18017160): Register a domain and all its sub-domains to match the “myself” condition. It can be set many times. Its full format is: 'proto:domain:port', allowing to set restrictions on protocol and port as well. Protocol and port are optional. alias core parameter ( http://www.kamailio.org/wiki/cookbooks/devel/core#alias): Parameter to set alias hostnames for the server. It can be set many times, each value being added in a list to match the hostname when 'myself' is checked. Is the only difference that corex alias_subdomains can include 'proto' ? -- Iwan Aucamp

2 1

Question about how location server_id is supposed to be used.
by Ding Ma 05 Aug '15

05 Aug '15

Looking for some doc that describes how the new columns in 4.3.x location table is supposed to be used. Can find some email discussions about this, but seems no formal doc exists. Would appreciate if someone can provide an explanation. Thanks

3 3

Re: [SR-Users] R: Mutiple branch forking and voicemail
by Daniel-Constantin Mierla 05 Aug '15

05 Aug '15

Indeed usually there are few options for same functionality, good to hear you found a solution. Cheers, Daniel On 03/08/15 14:53, Marino Mileti wrote: > Thanks Daniel, > > I've found another way using a seturi for mediaserver that will be executed > after all branch (using t_load_contacts and t_next_contacts) > In this way I can complete my scenario in the right way > > -----Messaggio originale----- > Da: sr-users [mailto:sr-users-bounces@lists.sip-router.org] Per conto di > Daniel-Constantin Mierla > Inviato: lunedì 3 agosto 2015 12:37 > A: Kamailio (SER) - Users Mailing List > Oggetto: Re: [SR-Users] Mutiple branch forking and voicemail > > Hello, > > On 31/07/15 11:31, Marino Mileti wrote: >> Hello, >> >> I would like to implement a voicemail services but i'm trying >> everything without good results :( >> >> My scenario is: >> >> - A make a parallel forking to B(1001) and C(1002) >> - If timeout, call is routed to voicemail >> >> I've done something like this...very simple >> >> seturi("sip:1001@172.20.40.103"); >> >> append_branch("1002(a)172.20.40.103"); >> route(RELAY) >> >> on failure_route.. >> if (t_check_status("486|408")) { >> $ru = "sip:8000@172.20.40.103:5090"; >> >> route(RELAY); >> >> exit; >> >> } >> >> The scenario is working but on log there are some error... >> t_should_relay_response >> >> I would like that when a branch reach the failure route, all other >> branches will be cancelled..and call is routed to VM. >> Can anyone help me? >> >> > When the failure_route is executed, no branch is active anymore. > > To be able to assist further, you have to send here the errors you get in > the log, to understand what is wrong. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla > http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > Book: SIP Routing With Kamailio - http://www.asipto.com > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > --- > Questa e-mail è stata controllata per individuare virus con Avast antivirus. > https://www.avast.com/antivirus > > > _______________________________________________ > 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

1 0

Maintaining order of messages in a sip dialog on forwarding
by Daniel Tryba 05 Aug '15

05 Aug '15

Since switching to 4.2.5 (from 3.x) a customer has problems with a buggy user agent (IMHO). What I am seeing now is: provider kamailio customer INVITE1-> INVITE1-> <-Trying <-Trying <-200 OK <-200 OK ACK1-> INVITE2-> INVITE2-> ACK1-> ACK1 arrives 0.007s before INVITE2, but INVITE2 is send 0.004s before ACK1. It looks like this wasn't happening with Kamailio 3.x, it worked fine then (apparently). Can this reordering be (gracefully) prevented somehow (ignoring the fact that during transport reordering can happen)? A Q&D hack looks to be just usleep the reinvite a little. The endpoint is essentially ignoring the reinvite before (and after) the ACK. After a timeout INVITE2 is canceled and after that INVITE1 will be ended with a BYE from provider. kamailio-customer communication is UDP provider-kamailio is tcp when message >= 1300b. INVITE1 is TCP, INVITE2 and ACK1 are UDP.

2 5

VIA header question.
by Chad 04 Aug '15

04 Aug '15

Hi list, I need a little help, I am a business owner trying to get Kamailio up and running as a SIP load balancer. I hired a Kamailio consultant to help me do so, but Kamailio is not working and I am getting conflicting information. My Kamailio consultant and my VOIP provider are telling me 2 different things and I don't know which one is right. Kamailio sends SIP traffic to the VOIP provider with 2 VIA headers like this (in this order): Via: SIP/2.0/UDP 10.10.10.254;branch=z9hG4bK7291.6a0bbd2e8fd639a47d7d2de606779e47.0. Via: SIP/2.0/UDP 209.170.201.25:5060;received=10.10.10.102;branch=z9hG4bK742dc03d;rport=5060. The VOIP provider says that is incorrect because they are supposed to reply back to the topmost VIA header so they reply to the 10.10.10.254 IP (which is not public) and the call ends. The VOIP provider says Kamailio should send the VIA headers like this instead: Via: SIP/2.0/UDP 209.170.201.25:5060;received=10.10.10.102;branch=z9hG4bK742dc03d;rport=5060. Via: SIP/2.0/UDP 10.10.10.254;branch=z9hG4bK7291.6a0bbd2e8fd639a47d7d2de606779e47.0. My Kamailio consultant says the way we are sending it is right and that the VOIP provider is processing the call incorrectly. I read that the SIP proxy is supposed to remove the internal header from the 1st example above based on this RFC: https://tools.ietf.org/html/rfc3261#section-16.7 Item: 3. Via "The proxy removes the topmost Via header field value from the response." If that applies to this situation (which I don't know if it does) then Kamailio should be removing the 10.10.10.254 VIA line and only sending 1 VIA header like this: Via: SIP/2.0/UDP 209.170.201.25:5060;received=10.10.10.102;branch=z9hG4bK742dc03d;rport=5060. Which would sort of make the VOIP provider right in that the topmost VIA line would then be the external IP, but how they said to fix it (reversing the VIA lines) is wrong. Does anyone know what the right answer is here? Please let me know. THANKS for your help. -- ^C

3 3

MSILO: Archiving every message
by SamyGo 03 Aug '15

03 Aug '15

Hi, Tried finding anything as weird as I am trying to do on mailing list but couldn't. The idea is to save a copy of the messages on database. MSILO module can keep the message as long as the destination user is offline, or doesn't support method:MESSAGE or expiry timeout. Any ideas will be appreciated. Best Regards, Sammy

2 2

Re: [SR-Users] Mutiple branch forking and voicemail
by Daniel-Constantin Mierla 03 Aug '15

03 Aug '15

Hello, On 31/07/15 11:31, Marino Mileti wrote: > Hello, > > I would like to implement a voicemail services but i'm trying everything > without good results :( > > My scenario is: > > - A make a parallel forking to B(1001) and C(1002) > - If timeout, call is routed to voicemail > > I've done something like this...very simple > > seturi("sip:1001@172.20.40.103"); > > append_branch("1002(a)172.20.40.103"); > route(RELAY) > > on failure_route.. > if (t_check_status("486|408")) { > $ru = "sip:8000@172.20.40.103:5090"; > > route(RELAY); > > exit; > > } > > The scenario is working but on log there are some error... > t_should_relay_response > > I would like that when a branch reach the failure route, all other branches > will be cancelled..and call is routed to VM. > Can anyone help me? > > When the failure_route is executed, no branch is active anymore. To be able to assist further, you have to send here the errors you get in the log, to understand what is wrong. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com

2 1

www.kamailio.org unreachable via ipv6
by Daniel Tryba 03 Aug '15

03 Aug '15

(www.)kamailio.org is unreachable at this time (the whole morning) on 2a00:d60:0:400::2. ipv4 is responsing. Never noticed whether connectivity via ipv6 was working in the past or this is new and misconfigured.

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