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Database errors with usrloc db_mode=1 and dmq_usrloc
by Joel Serrano 25 Feb '20

25 Feb '20

Hello, I have a setup with 2 kamailio in active/idle, locations are shared via dmq_usrloc, but database is also used as a cache where other systems can access and query "real-time" data. The config we have is: # ----- usrloc params ----- modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 1) modparam("usrloc", "db_load", 0) modparam("usrloc", "db_insert_update", 1) modparam("usrloc", "db_timer_clean", 1) modparam("usrloc", "server_id_filter", 1) modparam("usrloc", "timer_procs", 2) modparam("usrloc", "timer_interval", 60) # ----- dmq_usrloc params ----- modparam("dmq_usrloc", "enable", 1) modparam("dmq_usrloc", "sync", 1) modparam("dmq_usrloc", "batch_msg_contacts", 50) # 50 contacts / message modparam("dmq_usrloc", "batch_size", 10000) # 10000 contacts / batch modparam("dmq_usrloc", "batch_usleep", 500000) # one batch every 500ms With this setup, we assume: 1- almost real-time updated location info in the database. 2- as long as one of the two Kam is up, location information will not be lost (as they will sync with each other via DMQ) 3- if both servers go down, locations are lost (as they will not reload from database on startup). Our tests work well with the above config, with one exception, on the idle node, we see a lot of (only happens on the node receiving location via dmq): Feb 21 00:11:25 csbc02 csbc[13916]: ERROR: <core> [db.c:481]: db_use_table(): invalid parameter value Feb 21 00:11:25 csbc02 csbc[13916]: ERROR: usrloc [ucontact.c:1136]: db_update_ucontact_ruid(): sql use_table failed Feb 21 00:11:25 csbc02 csbc[13916]: ERROR: usrloc [ucontact.c:1657]: update_contact_db(): failed to update database I added debug logs to see if I could understand the reason: Feb 21 14:38:34 csbc02 csbc[30796]: DEBUG: dmq_usrloc [usrloc_sync.c:478]: usrloc_dmq_handle_msg(): dmq message received from sip:usrloc@10.2.1.181:5050 Feb 21 14:38:34 csbc02 csbc[30796]: DEBUG: dmq_usrloc [usrloc_sync.c:423]: usrloc_dmq_execute_action(): Received DMQ_UPDATE. Update contact info... Feb 21 14:38:34 csbc02 csbc[30796]: DEBUG: dmq_usrloc [usrloc_sync.c:75]: add_contact(): aor: 1063362 Feb 21 14:38:34 csbc02 csbc[30796]: DEBUG: dmq_usrloc [usrloc_sync.c:76]: add_contact(): ci->ruid: uloc-1-5e502cd3-5629-1 Feb 21 14:38:34 csbc02 csbc[30796]: DEBUG: dmq_usrloc [usrloc_sync.c:77]: add_contact(): aorhash: 928261667 Feb 21 14:38:34 csbc02 csbc[30796]: DEBUG: dmq_usrloc [usrloc_sync.c:115]: add_contact(): '1063362' Not found in usrloc Feb 21 14:38:34 csbc02 csbc[30796]: DEBUG: dmq_usrloc [usrloc_sync.c:117]: add_contact(): Insert record Feb 21 14:38:34 csbc02 csbc[30796]: DEBUG: usrloc [ul_callback.h:83]: run_ul_callbacks(): contact=0x7fe41ff41f40, callback type 1/15, id 0 entered Feb 21 14:38:34 csbc02 csbc[30796]: DEBUG: dmq_usrloc [usrloc_sync.c:800]: dmq_ul_cb_contact(): Callback from usrloc with type=1 Feb 21 14:38:34 csbc02 csbc[30796]: DEBUG: dmq_usrloc [usrloc_sync.c:826]: dmq_ul_cb_contact(): Contact received from DMQ... skip Feb 21 14:38:34 csbc02 csbc[30796]: ERROR: <core> [db.c:481]: db_use_table(): invalid parameter value Feb 21 14:38:34 csbc02 csbc[30796]: ERROR: usrloc [ucontact.c:669]: db_insert_ucontact(): sql use_table failed Feb 21 14:38:34 csbc02 csbc[30796]: ERROR: usrloc [urecord.c:598]: insert_ucontact(): failed to insert in database Feb 21 14:38:34 csbc02 csbc[30796]: DEBUG: dmq_usrloc [usrloc_sync.c:121]: add_contact(): Insert ucontact Feb 21 14:38:34 csbc02 csbc[30796]: DEBUG: dmq_usrloc [usrloc_sync.c:124]: add_contact(): Release record Feb 21 14:38:34 csbc02 csbc[30796]: DEBUG: dmq_usrloc [usrloc_sync.c:126]: add_contact(): Unlock udomain Feb 21 14:38:34 csbc02 csbc[30796]: DEBUG: sl [sl.c:305]: send_reply(): reply in stateless mode (sl) Feb 21 14:38:34 csbc02 csbc[30796]: DEBUG: <core> [core/msg_translator.c:161]: check_via_address(): (10.2.1.181, 10.2.1.181, 0) Can anyone help me understand what's the problem? I have the server_id setting, and the modparam server_id_filter, so shouldn't the receiving node just ignore any database tasks? Thanks, Joel.

3 12

Using the To header to route a call
by Dovid Bender 24 Feb '20

24 Feb '20

Hi, We are fighting with a vendor and trying to explain to them that call routing should be based on the RURI on the INVITE and *not* the To header. I went through RFC 3261 and while there are many "indicators" and it seems like its screaming that we should be using the RURI, is anyone aware of any specific text (that I may have missed) where it says specifically call routing should be done based on the RURI and *NOT* the to header? TIA. Dovid https://tools.ietf.org/html/rfc3261#page-78 1) 8.1.1.1 Request-URI *The initial Request-URI of the message SHOULD be set to the value of the URI in the To field.* - Note that it says SHOULD but is NOT mandatory. Also this is for the *INITIAL* request. *This may or may not be the ultimate recipient of the request.* - Again this is the To field. The RURI is what we are supposed to follow when we want to know where the call should be delivered to. 2) 8.1.2 Sending the Request -Usage of the URI from the *To and From fields in the original request within subsequent requests is done for backwards compatibility with RFC 2543*, which used the URI for dialog identification. *In this specification, only the tags are used for dialog identification*. It is expected that mandatory reflection of the original To and From URI in mid-dialog requests will be deprecated in a subsequent revision of this specification. - While this is for subsequent requests (and not an original one) you see that the only point of the To field is to identify the dialog. 3) 8.2.2.1 To and Request-URI The To header field identifies the *original recipient of the request* designated by the user identified in the From field. - Note that the TO field identifies the ORIGINAL recipient, this **CAN** change. This is basically telling you that you *can't* rely on the To header for routing. *However, the Request-URI identifies the UAS that is to process the request*. - As you can see the Request-RU and *NOT* the TO header tells you where the request should go. 5) 12.2.1.1 Generating the Request a. The UAC uses the *remote target* and route set to build the *Request-URI and Route header field of the request*. - This is self explanatory. it shows that the RURI is using for routing and *NOT* the to header. b. The procedures in Section 8.1.2 will normally result in the request being *sent to the address* indicated by the topmost Route header field value or the *Request-URI* if no Route header field is present. Subject to certain restrictions, they allow the request to be sent to an alternate address (such as a default outbound proxy not represented in the route set). Again self explanatory. 6) 16.5 Determining Request Targets Each target in the set is *represented as a URI*. - This clearly shows that how we route is based on the RURI and *NOT* the to header. 7) 26.5 Privacy Thus it MAY be desirable for privacy reasons to create a To header field that differs from the Request-URI. - Again this shows that the to may not match up the RURI. Think of the To header as the contact in someones phone and the RURI as where we are calling. The to header may be bob(a)1.1.1.1 but the URI is where we are actually calling which is 12346(a)1.1.1.1 or bob_smith(a)1.1.1.1.1

3 2

Kamailio presence when account not equal extension
by Zhan Bazarov 24 Feb '20

24 Feb '20

Hello, I'm using presence and pua modules, it works perfectly when username is equal to extension... But now I have case when username is not equal to the extension.. Like: extension: 1111 username: aythnbghhgt How I can implement this? Any ideas? -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

2 3

CentOS 8 RPMs missing from OBS
by Anthony Alba 21 Feb '20

21 Feb '20

The 5.3.2-4 builds for CentOS 8 are not available here: https://download.opensuse.org/repositories/home:/kamailio:/v5.3.x-rpms/Cent… Is https://rpm.kamailio.org/centos/8/5.3/5.3.2/x86_64/ aka Pulp the official release location now? (With -0 builds) For CentOS 7 Pulp has 5.3.2-0 and OBS has 5.3.2-4. The version mismatch is confusing. Cheers Anthony

2 1

Kamailio presence when account not equal extension
by Zhan Bazarov 21 Feb '20

21 Feb '20

Hello, I'm using presence and pua modules, it works perfectly when username is equal to extension... But now I have case when username is not equal to the extension.. Like: extension: 1111 username: aythnbghhgt How I can implement this? Any ideas? -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

1 1

acc_radius
by volga629 21 Feb '20

21 Feb '20

2 1

Understanding some usrloc modparams
by Joel Serrano 20 Feb '20

20 Feb '20

Hello, I have an installation using usrloc + db_mode=1. If I want on db_mode=1 or db_mode=2 the expired contacts to be cleaned up I have to enable the modparam db_timer_clean: https://www.kamailio.org/docs/modules/devel/modules/usrloc.html#usrloc.p.db… The docs say: "*Enable (1) or disable (0) cleaning of expired db records on timer basis for db_mode WRITE-BACK and WRITE-THROUGH. It uses the secondary timer process.*" I have the following doubts: 1- What does the reference "it uses *the* secondary timer process" mean? Is there a specific secondary timer just for this purpose? 2- Is #1 related to the modparam timer_procs? (so it uses one of those when enabled?) 3- Why/When would you need a timer_proces modparam with a value higher than "1"? Thanks, Joel.

4 6

Active-Active Registrar setup with users behind NAT persistence in event of server failure
by Jonathan Hunter 19 Feb '20

19 Feb '20

Hi Guys, I have a quick question wondered if anyone could provide some suggestions please?! We have a setup of two registrars in EC2 (kamailio 4.4 running on debian VMs) which act in an active/active fashion with a number of devices behind NAT where we use the DMQ dmq_t_replicate function to send the registration details between each server and append the PATH parameter so the registrar the device registered with is always associated with the client device. This all works fine, the only issue is persistency should we have a failure of one of the registrars, be it process or server and recovering the correct PATH detail so the end device can still be reached. I have tried this with separate databases with each registrar, which works when using cache and db but I ideally want the same database for the two servers and not change the architecture at this time unless it is a necessary. I have also tested with dmq_usrloc but this again provides the usrlocation details of the notification servers in the group(not path), again which I understand but I wondered if there is anything else I could try or its just a question of in the event of a failure half the registrations will be effected until it recovers and they re-register again if I dont want to have separate databases (location tables) or I add more network elements into the flow, such as HA pair or load balancer but just want to make sure I have exhausted all options or consider the risk before I look to do this. Many thanks! Jon

2 2

IMS Third party Register issue
by Lanvin ADON 19 Feb '20

19 Feb '20

Hi All, I'm facing an issue with the third party registration in the ISC interface between the SCSCF and the AS. The AS is triggered for deregistration but not for registration. Attached are logs for a register that not works and a deregister that works Regards Lanvin ADON

2 3

Documentation Broken Link
by Sam Ware 19 Feb '20

19 Feb '20

On the http://www.kamailio.org/w/documentation/ page, the Docbook tutorials <http://sip-router.org/docbook/sip-router/branch/master/> link is broken. -- Sam D Ware

2 2

xlog() format change from 5.3 to 5.4-nightly
by Joel Serrano 19 Feb '20

19 Feb '20

Hello, I've noticed that the output format of xlog() changes from 5.3 to 5.4-nightly. v5.3 xlog(): Feb 13 11:56:35 csbc01 csbc[27333]: NOTICE: {1 24727 REGISTER b1d374f6-9b8c53ff(a)A.B.C.D} <script>: [log] auth successful v5.4-nightly xlog(): Feb 13 11:56:35 csbc01 csbc[27333]: NOTICE: {1 24727 REGISTER b1d374f6-9b8c53ff(a)A.B.C.D} <script>: xlog_helper(): [log] auth successful Is there a way to avoid adding the extra "*xlog_helper():*" string? We have a lot of scripts that parse the K logs for different reasons and I would like to avoid having to update them all to match the new format. I've tried adding *modparam("xlog", "prefix", "")* but it only affects the "<script>:" part and not the "xlog_helper():" part. Thanks, Joel.

3 4

ds_is_from_list doesn't accept groupid from a variable?
by Björn Bylander 14 Feb '20

14 Feb '20

Hi, According to https://kamailio.org/docs/modules/5.3.x/modules/dispatcher.html#dispatcher.…, ds_is_from_list should accept an integer variable for the group id parameter. The following simple example doesn¹t work however: $var(foo) = 5; ds_is_from_list($var(foo)); Kamailio logs ³function ds_is_from_list: parameter 1 is not constant². Is this a bug or have I misinterpreted the documentation somehow? Thanks in advance, Björn Bylander

2 1

Call-ID Uniqueness and safety?
by Mark Boyce 14 Feb '20

14 Feb '20

Hi all Over coffee we’ve been discussing the uniqueness and safety of call-id this morning. Lots of things like RTPEngine recording use call-id to generate filenames (yes it does escape them first :-). Not to mention dialog tracking (also uses from/to tags) etc. Which got us thinking. As call-id is determined by the first SIP packet that normally comes from a UA, should we care if this UA was broken or worse malicious? What would happen if they were repeated? For example we may struggle to determine a recordings owner in RTPEngine (file name format is callid+random.wav). For security should we be rejecting requests where we’ve seen the call-id before? All theoretical over coffee, as it was that or start working! Just wondering what peoples thoughts were? Cheers Mark

4 7

ACC module
by Voip support 13 Feb '20

13 Feb '20

Dear Community, I have setup ACCDB module to save missed / failed and successful calls. My issue is that i am getting duplicated entries for INVITE / BYE in acc database table. For example: id;method;from_tag;to_tag;callid;sip_code;sip_reason;time;src_ip;dst_user;dst_domain;src_user;src_domain;cdr_id 231838;INVITE;21-7FC1D85;1914472180948750;KKyIdaaJceAu7pzN(a)172.16.0.35;200;OK;2020-02-11 19:16:28;22.33.44.55;nlb124#78019989897056;11.11.11.11;asterisk;172.16.0.35;0 231839;INVITE;21-7FC1D85;1914472180948750;KKyIdaaJceAu7pzN(a)172.16.0.35;200;OK;2020-02-11 19:16:28;22.33.44.55;nlb124#78019989897056;11.11.11.11;asterisk;172.16.0.35;0 231840;INVITE;21-7FC1D85;1914472180948750;KKyIdaaJceAu7pzN(a)172.16.0.35;200;OK;2020-02-11 19:16:28;22.33.44.55;nlb124#78019989897056;11.11.11.11;asterisk;172.16.0.35;0 231841;BYE;21-7FC1D85;1914472180948750;KKyIdaaJceAu7pzN(a)172.16.0.35;200;OK;2020-02-11 19:16:36;22.33.44.55;;190.196.153.130;asterisk;172.16.0.35;0 231842;BYE;21-7FC1D85;1914472180948750;KKyIdaaJceAu7pzN(a)172.16.0.35;200;OK;2020-02-11 19:16:36;22.33.44.55;;190.196.153.130;asterisk;172.16.0.35;0 231844;BYE;21-7FC1D85;1914472180948750;KKyIdaaJceAu7pzN(a)172.16.0.35;200;OK;2020-02-11 19:16:36;22.33.44.55;;190.196.153.130;asterisk;172.16.0.35;0 Following way: # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. * if you enable this parameter, be sure the enable "append_fromtag" * in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) #modparam("acc", "db_extra", # "src_user=$fU;src_domain=$fd;src_ip=$si;" # "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;dst_user=$rU;dst_domain=$rd;src_ip=$si") # our lines modparam("acc", "cdrs_table", "acc_cdrs") modparam("acc", "cdr_on_failed", 1) #!endif Then in routing config i set setflag(FLT_ACC FLT_ACCMISSED FLT_ACCFAILED in several places. My question is - each setflag cause an entry to be saved in DB? How can i debug and resolve the issue of duplicates. I am trying to use a CDR script found somewhere on siremis sites to do the accounting but it's making trouble because due to duplicates in acc table i am getting cdrs unique key violations. Secondary question is - if a BYE is not received / lost is there any way to send a check triggered by kamailio to each caller and callee side to check if dialog is still active. I would not like to use session timers (as endpoints need to support it). Best regards, Tom

2 7

URI Mapping in Presence?
by Mark Boyce 12 Feb '20

12 Feb '20

Hi All Looking for the best way to achieve mapping from requested uri to actual uri in presence subscription. The scenario I’ve been presented with is that there is a direct relation between extension number and sip user but they are not the same thing. So for example; Extension 10 is mapped to SIP user sip-SomethingLong Extension 11 is mapped to SIP user sip-SomethingEvenLonger When extension 10 subscribes to 11 we end up with something like this; From: “Extension 10" <sip:sip-SomethingLong@ourpbx.com:5060>;tag=12345678 To: <sip:11@ourpbx.com:5060>;tag=987654321-c8bd I *think* what I need to end up with in presence db is; presently_uri = sip:sip-SomethingEvenLonger@ourpbx.com to_user = 10 (Is that right?) However I can’t see a standard way to achieve this. Manage_subscribe will let me override the watcher uri but not the watched. Anyone see a simple way to achieve this? Thanks Mark

2 2

Generating provisional responses from kamailio instance
by George Diamantopoulos 12 Feb '20

12 Feb '20

Hello all, I would like to use rtpengine's media playback capabilities in an early media scenario. The use case is as follows: * Kamailio receives an incoming INVITE * During INVITE processing, kamailio config script determines that an early media message should be played back and then the call should fail with 4xx * To that effect, Kamailio sends out a locally-generated 183 Progress and instructs rtpengine to playback an audio file * After the audio file has finished playing, kamailio responds with 4xx to the INVITE Is this possible at all? Thanks! BR, George

3 2

Kamailio crashes after Contact expires event route
by Sergiu Pojoga 12 Feb '20

12 Feb '20

Hi there, After below even route is triggered, Kamailio crashes with the following in logs: Feb 12 08:03:59 kamailio[24671]: INFO: <script>: Received HTTP/RPC request from IP=127.0.0.1 Feb 12 08:03:59 kamailio[24671]: WARNING: <script>: Contact myusername(a)sip.mydomain.net has expired Feb 12 08:03:59 kamailio[24671]: INFO: drouting [drouting.c:483]: rpc_reload(): RPC command received! Feb 12 08:03:59 kamailio[24675]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 59 Feb 12 08:03:59 kamailio[24614]: ALERT: <core> [main.c:745]: handle_sigs(): child process 24671 exited by a signal 11 Feb 12 08:03:59 kamailio[24614]: ALERT: <core> [main.c:748]: handle_sigs(): core was not generated Feb 12 08:03:59 kamailio[24614]: INFO: <core> [main.c:770]: handle_sigs(): terminating due to SIGCHLD Feb 12 08:03:59 kamailio[24675]: INFO: <core> [main.c:825]: sig_usr(): signal 15 received event_route[usrloc:contact-expired] { xlog("L_WARN", "Contact $ulc(exp=>aor) has expired\n"); # Remove the sip: from the front of the Received URI $var(exp_received) = $(ulc(exp=>received){s.substr,4,0}); sql_query("db", "UPDATE `dr_gateways` SET address = '127.0.0.1' WHERE address = '$var(exp_received)'", "rz"); sql_result_free("rz"); jsonrpc_exec('{"jsonrpc": "2.0", "method": "drouting.reload", "id": 1}'); } If I comment out the sql and jsonrpc_exec lines above - crash does not happen. Let me know if core dump is needed. Thanks, --Sergiu

2 1

RTP Engine in docker
by Voip support 12 Feb '20

12 Feb '20

Dear Community, I would like to use rtpengine but had a very hard time to do the compilation under ubuntu 16.04 and 18.04. On Ubuntu 18.04 after already compiled the rtpengine i was unable to install the deb packages. Many different errors occurred. I finally tried to install RTP engine on debian 10 and i was able to install it. I am thinking of 2 use scenarios: - handling many concurrent calls like using rtpproxy for normall traffic - make WebRTC to legacy RTP transcoding (convert WebRTC SDP to legacy SDP to use with non webrtc compliant sip server) For the first scenario i imagine that it would be far better to run in-kernel mode because of performance. For the second scenario i think userspace daemon should be fine ( i expect not much traffic maximum 50-100 calls). However my question is what is the correct order of installing the deb packages. Which packages do i really need. For running rtpengine in docker could i use Debian 10 OS and compile rtpengine and install just " ngcp-rtpengine-daemon_6.2.0.0+0~mr6.2.0.0_amd64.deb" ? (the in this case the host running docker can be any linux distribution?) If i would like to run rtpengine in docker in kernel mode - is it possible or i need to use same linux distribution in host and docker container (because of the kernel match?) Best regards, Tom

5 6

Dispatcher error with rtpptoxy
by Gaurav Bmotra 12 Feb '20

12 Feb '20

hi i m using kamailio 5.2.1 in Kubernetese on Host network it was working file but now the following issue i n getting call connecting but audio both way issue here is the logs that got -------------------------------- 2(14) ERROR: {1 2 INVITE EQEiv5Mjs-eV1OCGv6Potw..} rtpproxy [rtpproxy.c:2586]: force_rtp_proxy(): incorrect port 0 in reply from rtp proxy 4(16) ERROR: {2 2 INVITE EQEiv5Mjs-eV1OCGv6Potw..} rtpproxy [rtpproxy.c:2586]: force_rtp_proxy(): incorrect port 0 in reply from rtp proxy 4(16) NOTICE: {2 2 INVITE EQEiv5Mjs-eV1OCGv6Potw..} acc [acc.c:279]: acc_log_request(): ACC: transaction answered: timestamp=1578636040;method=INVITE;from_tag=88c92018;to_tag=2c3c0a16;call_id=EQEiv5Mjs-eV1OCGv6Potw..;code=200;reason=OK;src_user=200;src_domain=testcalls.aone.one;src_ip=157.39.9.195;dst_ouser=300;dst_user=300;dst_domain=106.78.70.57 4(16) ERROR: {2 2 INVITE EQEiv5Mjs-eV1OCGv6Potw..} db_mysql [km_dbase.c:126]: db_mysql_submit_query(): driver error on query: Unknown column 'src_user' in 'field list' (1054) 4(16) ERROR: {2 2 INVITE EQEiv5Mjs-eV1OCGv6Potw..} <core> [db_query.c:243]: db_do_insert_cmd(): error while submitting query 4(16) ERROR: {2 2 INVITE EQEiv5Mjs-eV1OCGv6Potw..} acc [acc.c:477]: acc_db_request(): failed to insert into database 6(18) ERROR: {2 2 INVITE EQEiv5Mjs-eV1OCGv6Potw..} rtpproxy [rtpproxy.c:2586]: force_rtp_proxy(): incorrect port 0 in reply from rtp proxy 8(20) NOTICE: {2 3 BYE EQEiv5Mjs-eV1OCGv6Potw..} acc [acc.c:279]: acc_log_request(): ACC: transaction answered: timestamp=1578636041;method=BYE;from_tag=88c92018;to_tag=2c3c0a16;call_id=EQEiv5Mjs-eV1OCGv6Potw..;code=200;reason=OK;src_user=200;src_domain=testcalls.aone.one;src_ip=157.39.9.195;dst_ouser=300;dst_user=300;dst_domain=106.78.70.57 8(20) ERROR: {2 3 BYE EQEiv5Mjs-eV1OCGv6Potw..} db_mysql [km_dbase.c:126]: db_mysql_submit_query(): driver error on query: Unknown column 'src_user' in 'field list' (1054) 8(20) ERROR: {2 3 BYE EQEiv5Mjs-eV1OCGv6Potw..} <core> [db_query.c:243]: db_do_insert_cmd(): error while submitting query 8(20) ERROR: {2 3 BYE EQEiv5Mjs-eV1OCGv6Potw..} acc [acc.c:477]: acc_db_request(): failed to insert into database 3(15) ERROR: {1 2 INVITE LH6sikNzpP9rXXMJvH20wQ..} rtpproxy [rtpproxy.c:2586]: force_rtp_proxy(): incorrect port 0 in reply from rtp proxy 9(21) NOTICE: acc [acc.c:279]: acc_log_request(): ACC: call missed: timestamp=1578636141;method=INVITE;from_tag=6b032a77;to_tag=;call_id=LH6sikNzpP9rXXMJvH20wQ..;code=408;reason=Request Timeout;src_user=200;src_domain=testcalls.aone.one;src_ip=157.39.9.195;dst_ouser=300;dst_user=300;dst_domain=106.78.70.57 9(21) ERROR: db_mysql [km_dbase.c:126]: db_mysql_submit_query(): driver error on query: Unknown column 'src_user' in 'field list' (1054) 9(21) ERROR: <core> [db_query.c:243]: db_do_insert_cmd(): error while submitting query 9(21) ERROR: acc [acc.c:477]: acc_db_request(): failed to insert into database 9(21) NOTICE: {1 2 INVITE LH6sikNzpP9rXXMJvH20wQ..} <script>: RTF_DISPATCH: INVITE 300 [LH6sikNzpP9rXXMJvH20wQ..] 9(21) NOTICE: {1 2 INVITE LH6sikNzpP9rXXMJvH20wQ..} <script>: RTF_DISPATCH: WARNING sip: 49.14.97.169:63353 is broken and marked as inactive 3(15) ERROR: {1 2 INVITE 2gY1N8nGCT02Ut7iAEpKWg..} rtpproxy [rtpproxy.c:2586]: force_rtp_proxy(): incorrect port 0 in reply from rtp proxy 8(20) ERROR: {2 2 INVITE 2gY1N8nGCT02Ut7iAEpKWg..} rtpproxy [rtpproxy.c:2586]: force_rtp_proxy(): incorrect port 0 in reply from rtp proxy 8(20) NOTICE: {2 2 INVITE 2gY1N8nGCT02Ut7iAEpKWg..} acc [acc.c:279]: acc_log_request(): ACC: transaction answered: timestamp=1578638074;method=INVITE;from_tag=b9979f2a;to_tag=c2952a71;call_id=2gY1N8nGCT02Ut7iAEpKWg..;code=200;reason=OK;src_user=200;src_domain=testcalls.aone.one;src_ip=157.39.9.195;dst_ouser=300;dst_user=300;dst_domain=106.78.71.32 8(20) ERROR: {2 2 INVITE 2gY1N8nGCT02Ut7iAEpKWg..} db_mysql [km_dbase.c:126]: db_mysql_submit_query(): driver error on query: Unknown column 'src_user' in 'field list' (1054) ------------------------------- plz help me thank you -- *Regards:* Gaurav Kumar

3 2

Same Via Tag for INVITE and ACK on S/L loadbalancer
by Sebastian Damm 11 Feb '20

11 Feb '20

Hi, I stumbled upon an interop problem with a carrier. We have the following scenario: Gateway --> Loadbalancer --> Carrier The loadbalancer generates a Via header for each request. But since it is stateless, the Via tag is generated for each request. As a consequence, the Via tag in the ACK differs from the one in the INVITE. And one carrier doesn't handle those ACKs if the Via tag differs. Is there a way to force the creation of a "deterministic" Via branch tag? For example, building it from a hash over call-id and from-tag or something like that? Thanks in advance Sebastian -- Sebastian Damm Voice Engineer __________________________________________ sipgate GmbH

5 10

Different REGISTER expires based on User-Agent
by mayamatakeshi 11 Feb '20

11 Feb '20

Hi, is it possible to reply with a different expires for a REGISTER depending on some condition like the User-Agent? I am thinking maybe I can use: $ulc(profile=>expires) = 600 (but my kamailio dev machine is out of service and I cannot test this right now).

5 8

Kamailio v5.1.10 Released
by Daniel-Constantin Mierla 10 Feb '20

10 Feb '20

Hello, Kamailio SIP Server v5.1.10 stable release is out. This is a maintenance release of the old stable branch, 5.1, that includes fixes since the release of v5.1.9. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.1.x. Deployments running previous v5.1.x versions are strongly recommended to be upgraded to v5.1.10, to v5.2.x or 5.3.x series. For more details about version 5.1.10 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2020/02/kamailio-v5-1-10-released/ RPM, Debian/Ubuntu packages will be available soon as well. Note: the branch 5.1 is an old stable branch, v5.1.10 being the last planned release out of it, to mark the end of official maintenance. The latest stable branches are now 5.2 and 5.3, at this time with v5.3.2 being the latest stable release. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - April 27-29, 2020 -- www.kamailioworld.com

1 0

Planning the release of v5.1.10
by Daniel-Constantin Mierla 10 Feb '20

10 Feb '20

Hello, I am considering to release Kamailio v5.1.10 on Thursday, Jan 30, 2020, to mark the end of official packaging from branch 5.1. At this moment the last two stable branches are 5.2 and 5.3. If anyone is aware of some commits for fixes that should be backported to branch 5.1, write to sr-dev. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - March 9-11, 2020, Berlin - www.asipto.com Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com

1 2

asterisk kamailio mobile network
by Gintaras Kazakevicius 07 Feb '20

07 Feb '20

Hello, need help. Call from mobile to asterisk: Mobile network provider <trunk> Kamailio <-> Asterisk --- working Call from asterisk to mobile Asterisk <-> Kamailio <trunk> Mobile network provider --- not working How to configure? Maybe you have working example of this solution?

2 1

DIAMETER (server & avp helper)
by Steve Bucklin 07 Feb '20

07 Feb '20

Anyone familiar with: ims_diameter_server cdp cdp_avp I have set Kamailio up to connect with SMSC using Diameter. Initially just want to capture requests in JSON and examine. My trace tells me that I am sending a good message for DIRECT_DEBITING (436). I know the ocs module does not support this - only trying to "take apart" the message. When I get the message, I experience the following: Feb 7 13:07:58 ip-172-31-45-179 /sbin/kamailio[18048]: ERROR: cdp [diameter_avp.c:645]: AAAUngroupAVPS(): hss3g_ungroup_avps: source buffer to short!! Cannot read the whole AVP header! Feb 7 13:07:58 ip-172-31-45-179 /sbin/kamailio[18048]: CRITICAL: cdp [diameter_avp.c:702]: AAAUngroupAVPS(): AVP:<> I suspect this is due to vendor, but not experienced here. The docs seem to suggest the 3gpp is supported, and the diameter xml is set for 0 & 10415? Steve

1 0

Broken link on https://www.kamailio.org/wiki/cookbooks/devel/selects
by Björn Bylander 07 Feb '20

07 Feb '20

Hi, In the section "Complete Select List" on X there's a link to http://sip-router.org/docbook/sip-router/branch/master/select_list/select_l… which doesn't exist anymore. Can anyone tell me where the current list is located? Thanks in advance, Björn Bylander

2 1

Best practice to figure out one way rtp delay
by Karsten Horsmann 06 Feb '20

06 Feb '20

Hi List, I have a new setup with two Kamailios installations. One serves siptrunks from the internet and one is for internal routing. So far so good. At the end I have an 3rd party b2bua that receives and sends calls via the both Kamailios. For testing purposes I setup an freeswitch with beep and then echo application to the caller. My call flow are like this Call 1 to kam1 sbc then internal kam2 and b2bua. B2bua make then new call no 2 vice versa to PSTN freeswitch with echo. After this b2bua bridges the calls together. This generates for the caller an beep and echo. The interesting thing is now, that Call 1 gets an hearable delay of 1 second. But only in the rtp steam from me to the caller. The second calls seems equal of timing. Since is a really new complete setup of hardware and stuff, the question of its working before is answered with an no. Now my question to you guys. How can I get an measurable method to finding the delaying parts (could be network, servers, applications etc). Only capturing on one place don't did the trick for me. Thanks for your hints Cheers Karsten

3 4

Binary Websocket messages
by Andre Pires dos Santos 06 Feb '20

06 Feb '20

Hello guys, we are new to kamailio so sorry basic or wrong points. We started using kamailio 5.3.2 and rtpengine as our sip to webrtc solution and SIP.js as webrtc client. We noticed that most of the times webrtc client receives UTF-8 messages. But sometimes webrtc client receives binary messages. When we get binary messages SIP.js cannot parse the message. I couldn't find how to force kamailio to send only UTF-8 messages. Is it possible? We have other issues but I think it will be better for the list to break other points in different messages. Thanks! Santos

2 8

IMS RX AAR issues
by Pafel 06 Feb '20

06 Feb '20

Hello, I am using kamailio 5.3 + rtpengine module + sipwise rtpengine in a IMS test environment. I have and issue with ims_qos module and Rx AAR command. PCSCF is configured with #!define WITH_RX ##!define WITH_RX_REG #!define WITH_RX_CALL I have 2 test phones connected. Both of them use ipv6. PCSCF has ipv4 and ipv6 address. All other nodes - SCSCF, ICSCF use ipv4. Simple scenario UE1 calls UE2. UE2 is replying with 183 session progress and then RX AAR is sent by PCSCF to PCRF/PGW. The problem is that AVPs Media-Sub-Component Flow-Description include ipv4 address of the proxy and ipv6 address of the UE. Example: permit out 17 from 192.168.1.1 34998 to fd18::1:2dca:3303:1286:257d 50010 permit in 17 from fd18::1:2dca:3303:1286:257d 50010 to 192.168.1.1 34998 But this is not acceptable for the PGW (different ip versions). The SDP between UE and PCSCF is rewritten by rtpengine/sipwise. Looking at the ims_qos module I found that the address of the proxy is taken from the session original request SDP media ip/port. Having in mind that rtpengine/sipwise proxy is rewritting it is there any way I can set the correct ipv6 address and port of the proxy? I've tried to get the rewritten media address from SDP (in logs) but without success: if(sdp_get_line_startswith("$avp(cline)", "c=")) { xlog("L_INFO", "SDP media ip: $(avp(cline){s.select,2, })\n"); } if(sdp_get_line_startswith("$avp(mline)", "m=")) { xlog("L_INFO", "SDP media port: $(avp(mline){s.select,1, })\n"); } I am always getting the ipv4 media address and port of the proxy. Any idea how I can set and use the rewritten by the rtpengine/sipwise media ip and proxy for the RX AAR? Thanks in advance, Pavel Siderov

1 1

Re: [SR-Users] Pre-built Debian Package for Kemix Module?
by Yufei Tao 05 Feb '20

05 Feb '20

Hi Daniel and Victor, Thank you very much for the very quick and helpful response, as always! Best regards, Yufei

1 0

Pre-built Debian Package for Kemix Module?
by Yufei Tao 05 Feb '20

05 Feb '20

Hi, I am trying to use KEMI lua that needs kemix module (for KSR.kx.<...>). I wonder if there is any pre-built Debian package that include the kemix module, something like kamailio-<...>-modules, that I can install? Or do I need to build from source with kemix in included_modules? On a related question, for the kamailio-<...>-modules packages how do I find out what modules are included in each of these package? E.g. things like kamailio-lua-modules, kamailio-mysql-modules etc. are easy to guess, while kamailio-nth, kamailio-utils-modules etc. are not so easy. Thank you! Yufei

3 3

config file packaging
by Sergey Safarov 05 Feb '20

05 Feb '20

Hi At present time kamailio config file packaged with kamailio binaries. So projects that use kamailio cannot simple package "/etc/kamailio/kamailio.cfg" file. This leads to packages conflict on RPM dists. As options we can separate "kamailio default config" and "kamailio binaries" to packages. 1) kamailio-cfg-default 2) kamailio-base This allow not install "kamailio-default" config when project provide own config file. For backward compatibility we create "kamailio" empty package that have two dependencies "kamailio-cfg-default" and "kamailio-base". What you think?

2 2

How to implement simple presence in kamailio active-active cluster.
by Zhan Bazarov 05 Feb '20

05 Feb '20

Hello! We have next setup, which I've desribed in: http://sip-router.1086192.n5.nabble.com/Bad-request-when-INVITE-relayed-to-… Presence works fine in setup with single kamailio, without any loaded presence modules... So, I want to something wtih SUBSCRIBE and NOTIFY methods... But when I making a relay on second kamailio instance I receive 407 proxy auth required from this instance... So, its how I'm trying to implement this: part of code from route[LOCATION] case -2: if (is_method("SUBSCRIBE|NOTIFY")) { $ru = "sip:" + $tU + "@" + "ip_of_second_kam:5078"; t_relay("ip_of_second_kam", 5078); exit; } else { send_reply("405", "Method Not Allowed"); exit; } Any ideas how I can do that? Thanks. -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

1 0

Re: [SR-Users] P-CSCF not starting- Kamailio 5.3
by Henning Westerholt 04 Feb '20

04 Feb '20

Hi Lanvin, (please keep the sr-users list in CC) the modules works on my machine: henning@linux:~/repositories/kamailio> killall kamailio ; ./src/kamailio -f etc/kamailio.cfg -L src/modules/ -Y /tmp -a no -E -e -d Listening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 192.168.188.25 [192.168.188.25]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 tcp: 192.168.188.25 [192.168.188.25]:5060 Aliases: henning@linux:~/repositories/kamailio> ps aux |grep kamailio henning 15948 0.1 0.1 166776 14900 ? S 15:33 0:00 ./src/kamailio -f etc/kamailio.cfg -L src/modules/ -Y /tmp -a no -E -e -d henning 15949 0.0 0.0 166776 4224 ? S 15:33 0:00 ./src/kamailio -f etc/kamailio.cfg -L src/modules/ -Y /tmp -a no -E -e -d [...] Here the diff to the default cfg in git master: diff --git a/etc/kamailio.cfg b/etc/kamailio.cfg index c0e618719..dbc911fb0 100644 --- a/etc/kamailio.cfg +++ b/etc/kamailio.cfg @@ -309,6 +309,14 @@ loadmodule "pike.so" loadmodule "debugger.so" #!endif + +loadmodule "ims_dialog" +loadmodule "ims_usrloc_pcscf" +loadmodule "ims_ipsec_pcscf" +loadmodule "ims_registrar_pcscf" +modparam("ims_dialog", "dlg_flag", 4) + + Cheers, Henning Am 04.02.20 um 11:55 schrieb Lanvin ADON: > > > loadmodule "ims_dialog" > loadmodule "ims_usrloc_pcscf" > #loadmodule "ims_registrar_pcscf" > #!ifdef WITH_IPSEC > loadmodule "ims_ipsec_pcscf" > #!endif > loadmodule "ims_registrar_pcscf" > > > > > Feb 4 11:53:27 pcscf kamailio[17616]: 0(17618) ERROR: > ims_registrar_pcscf [ims_registrar_pcscf_mod.c:268]: mod_init(): can't > bind ims_ipsec_pcscf > Feb 4 11:53:27 pcscf kamailio[17616]: 0(17618) ERROR: <core> > [core/sr_module.c:849]: init_mod(): Error while initializing module > ims_registrar_pcscf > (/opt/pcscf/lib64/kamailio/modules/ims_registrar_pcscf.so) > Feb 4 11:53:27 pcscf kamailio[17616]: ERROR: error while initializing > modules > Feb 4 11:53:28 pcscf kamailio[17620]: 0(17622) ERROR: > ims_registrar_pcscf [ims_registrar_pcscf_mod.c:268]: mod_init(): can't > bind ims_ipsec_pcscf > Feb 4 11:53:28 pcscf kamailio[17620]: 0(17622) ERROR: <core> > [core/sr_module.c:849]: init_mod(): Error while initializing module > ims_registrar_pcscf > (/opt/pcscf/lib64/kamailio/modules/ims_registrar_pcscf.so) > Feb 4 11:53:28 pcscf kamailio[17620]: ERROR: error while initializing > modules > Feb 4 11:53:28 pcscf kamailio[17623]: 0(17625) ERROR: > ims_registrar_pcscf [ims_registrar_pcscf_mod.c:268]: mod_init(): can't > bind ims_ipsec_pcscf > Feb 4 11:53:28 pcscf kamailio[17623]: 0(17625) ERROR: <core> > [core/sr_module.c:849]: init_mod(): Error while initializing module > ims_registrar_pcscf > (/opt/pcscf/lib64/kamailio/modules/ims_registrar_pcscf.so) > Feb 4 11:53:28 pcscf kamailio[17623]: ERROR: error while initializing > modules > Feb 4 11:53:29 pcscf kamailio[17626]: 0(17628) ERROR: > ims_registrar_pcscf [ims_registrar_pcscf_mod.c:268]: mod_init(): can't > bind ims_ipsec_pcscf > Feb 4 11:53:29 pcscf kamailio[17626]: 0(17628) ERROR: <core> > [core/sr_module.c:849]: init_mod(): Error while initializing module > ims_registrar_pcscf > (/opt/pcscf/lib64/kamailio/modules/ims_registrar_pcscf.so) > Feb 4 11:53:29 pcscf kamailio[17626]: ERROR: error while initializing > modules > Feb 4 11:53:29 pcscf kamailio[17629]: 0(17631) ERROR: > ims_registrar_pcscf [ims_registrar_pcscf_mod.c:268]: mod_init(): can't > bind ims_ipsec_pcscf > Feb 4 11:53:29 pcscf kamailio[17629]: 0(17631) ERROR: <core> > [core/sr_module.c:849]: init_mod(): Error while initializing module > ims_registrar_pcscf > (/opt/pcscf/lib64/kamailio/modules/ims_registrar_pcscf.so) > Feb 4 11:53:29 pcscf kamailio[17629]: ERROR: error while initializing > modules > > > > > > > > > On Tue, Feb 4, 2020 at 11:46 AM Henning Westerholt <hw(a)skalatan.de > <mailto:hw@skalatan.de>> wrote: > > Hello Lanvin, > > try to load it before and see if it starts then. > > Cheers, > > Henning > > -- > > Henning Westerholt – https://skalatan.de/blog/ > > Kamailio services – https://gilawa.com <https://gilawa.com/> > > *From:* Lanvin ADON <lanvin.adon(a)gmail.com > <mailto:lanvin.adon@gmail.com>> > *Sent:* Tuesday, February 4, 2020 11:45 AM > *To:* Henning Westerholt <hw(a)skalatan.de <mailto:hw@skalatan.de>> > *Subject:* Re: [SR-Users] P-CSCF not starting- Kamailio 5.3 > > Hello Henning > > ims_ipsec_pcscf is loaded after. > > loadmodule "ims_dialog" > loadmodule "ims_usrloc_pcscf" > loadmodule "ims_registrar_pcscf" > #!ifdef WITH_IPSEC > loadmodule "ims_ipsec_pcscf" > #!endif > > Thanks > > Lanvin ADON > > On Tue, Feb 4, 2020 at 11:06 AM Henning Westerholt <hw(a)skalatan.de > <mailto:hw@skalatan.de>> wrote: > > Hello, > > just a guess, but do you load the ims_ipsec_pcscf module > before the ims_registrar_pcscf module? > > Cheers, > > Henning > > -- > > Henning Westerholt – https://skalatan.de/blog/ > > Kamailio services – https://gilawa.com <https://gilawa.com/> > > *From:* sr-users <sr-users-bounces(a)lists.kamailio.org > <mailto:sr-users-bounces@lists.kamailio.org>> *On Behalf Of > *Lanvin ADON > *Sent:* Tuesday, February 4, 2020 8:54 AM > *To:* sr-users(a)lists.kamailio.org > <mailto:sr-users@lists.kamailio.org> > *Subject:* [SR-Users] P-CSCF not starting- Kamailio 5.3 > > Hi All, > > I'm installing an IMS core with Kamailio 5.3 release. The > P-CSCF is not starting with the following error message. > > Feb 4 07:01:32 pcscf kamailio[13878]: 0(13880) DEBUG: <core> > [core/sr_module.c:638]: find_mod_export_record(): export of > <bind_ims_ipsec_pcscf> not found (flags 0) > Feb 4 07:01:32 pcscf kamailio[13878]: 0(13880) ERROR: > ims_registrar_pcscf [ims_registrar_pcscf_mod.c:268]: > mod_init(): can't bind ims_ipsec_pcscf > Feb 4 07:01:32 pcscf kamailio[13878]: 0(13880) ERROR: <core> > [core/sr_module.c:849]: init_mod(): Error while initializing > module ims_registrar_pcscf > (/opt/pcscf/lib64/kamailio/modules/ims_registrar_pcscf.so) > Feb 4 07:01:32 pcscf kamailio[13878]: ERROR: error while > initializing modules > > For the configuration file, I'm using the example files in the > source. > > > > -- > > Regards > > Lanvin ADON > > > > -- > > Cordialement, > > Lanvin ADON > > > > -- > > */ > /* > > Cordialement, > > Lanvin ADON -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com Kamailio Merchandising - https://skalatan.de/merchandising/

1 0

Re: [SR-Users] P-CSCF not starting- Kamailio 5.3
by Henning Westerholt 04 Feb '20

04 Feb '20

Hello Lanvin, try to load it before and see if it starts then. Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: Lanvin ADON <lanvin.adon(a)gmail.com> Sent: Tuesday, February 4, 2020 11:45 AM To: Henning Westerholt <hw(a)skalatan.de> Subject: Re: [SR-Users] P-CSCF not starting- Kamailio 5.3 Hello Henning ims_ipsec_pcscf is loaded after. loadmodule "ims_dialog" loadmodule "ims_usrloc_pcscf" loadmodule "ims_registrar_pcscf" #!ifdef WITH_IPSEC loadmodule "ims_ipsec_pcscf" #!endif Thanks Lanvin ADON On Tue, Feb 4, 2020 at 11:06 AM Henning Westerholt <hw(a)skalatan.de<mailto:hw@skalatan.de>> wrote: Hello, just a guess, but do you load the ims_ipsec_pcscf module before the ims_registrar_pcscf module? Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: sr-users <sr-users-bounces(a)lists.kamailio.org<mailto:sr-users-bounces@lists.kamailio.org>> On Behalf Of Lanvin ADON Sent: Tuesday, February 4, 2020 8:54 AM To: sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> Subject: [SR-Users] P-CSCF not starting- Kamailio 5.3 Hi All, I'm installing an IMS core with Kamailio 5.3 release. The P-CSCF is not starting with the following error message. Feb 4 07:01:32 pcscf kamailio[13878]: 0(13880) DEBUG: <core> [core/sr_module.c:638]: find_mod_export_record(): export of <bind_ims_ipsec_pcscf> not found (flags 0) Feb 4 07:01:32 pcscf kamailio[13878]: 0(13880) ERROR: ims_registrar_pcscf [ims_registrar_pcscf_mod.c:268]: mod_init(): can't bind ims_ipsec_pcscf Feb 4 07:01:32 pcscf kamailio[13878]: 0(13880) ERROR: <core> [core/sr_module.c:849]: init_mod(): Error while initializing module ims_registrar_pcscf (/opt/pcscf/lib64/kamailio/modules/ims_registrar_pcscf.so) Feb 4 07:01:32 pcscf kamailio[13878]: ERROR: error while initializing modules For the configuration file, I'm using the example files in the source. -- Regards Lanvin ADON -- Cordialement, Lanvin ADON

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P-CSCF not starting- Kamailio 5.3
by Lanvin ADON 04 Feb '20

04 Feb '20

Hi All, I'm installing an IMS core with Kamailio 5.3 release. The P-CSCF is not starting with the following error message. Feb 4 07:01:32 pcscf kamailio[13878]: 0(13880) DEBUG: <core> [core/sr_module.c:638]: find_mod_export_record(): export of <bind_ims_ipsec_pcscf> not found (flags 0) Feb 4 07:01:32 pcscf kamailio[13878]: 0(13880) ERROR: ims_registrar_pcscf [ims_registrar_pcscf_mod.c:268]: mod_init(): can't bind ims_ipsec_pcscf Feb 4 07:01:32 pcscf kamailio[13878]: 0(13880) ERROR: <core> [core/sr_module.c:849]: init_mod(): Error while initializing module ims_registrar_pcscf (/opt/pcscf/lib64/kamailio/modules/ims_registrar_pcscf.so) Feb 4 07:01:32 pcscf kamailio[13878]: ERROR: error while initializing modules For the configuration file, I'm using the example files in the source. -- Regards Lanvin ADON

2 1

parallel forking in combination with rtpengine
by michael.willow＠web.de 04 Feb '20

04 Feb '20

Hi everbody, I encountered an issue with my Kamailio in combination with Rtpengine. I wondered if anybody got a solution or a hint as this issue is only in combination with special parallel forked calls. I sent a call from my SIP gateway via a Kamailio to another SIP provider. The other SIP provider or the B-user seems to use a parallel forking as I receive 183 Session Progress and 180 Ringing with different to-tags. My kamailio is using a Rtpengine to relay the media from my SIP gateway to the other SIP provider. As Kamailio receives two different to-tags that include a SDP body rtpengine_manage makes Rtpengine to change RTP port. Some short example (cut down to the important messages): a-gateway kamailio b-provider | INVITE Port 16890 | INVITE Port 20400 | | ------------------------> | --------------------------> | | | | | 180 to-tag 1 / no SDP | 180 to-tag 1 / no SDP | | <------------------------ | <------------------------- | | | | | 183 to-tag 2 / Port 20408 | 183 to-tag 2 / Port 46298 | | <------------------------ | <------------------------- | | | | | | | | RTP | RTP | | Port 16890 <---- Port 20408 | Port 20400 <---- Port 46298 | | | | | | | | RTP | RTP | | Port 16890 ----> Port 20408 | Port 20400 ----> Port 46298 | | | | | | | | 180 to-tag 2 / Port 20408 | 180 to-tag 0 / Port 46298 | | <------------------------ | <------------------------- | | | | | | | | 200 OK to-tag 1 / Port 20408 | 200 OK to-tag 1 / Port 46298 | | <------------------------ | <------------------------- | | | | | | | | RTP | RTP | | Port 16890 <---- Port 20408 | Port 20400 <---- Port 46298 | | | | | | | | RTP | RTP | | Port 16890 ----> Port 20408 | Port 20418 ----> Port 46298 | Kamailio receives INVITE from a-gateway with rtp port 16890 and rtpengine rewrites it with port 20400 to b-provider. b-provider sends 180 ringing from to-tag 1 and no SDP which is just passed through kamailio to a-gateway. b-provider sends 183 progress from to-tag 2 and SDP with port 46298 to kamailio. rtpengines rewrites the sdp port to 20408 which is passed to a-gateway. b-provider start sending RTP via the negotiated ports which is passed correctly to a-gateway. b-provider sends 180 ringing from to-tag 2 with the same SDP. Rewritten and passed to a-gateway. b-provider sends 200 OK on Invite. This 200 Ok is from to-tag 1 but including the same SDP endpoint as to-tag 2 used before. Media from b-provider via rtpengine to a-gateway is passed the same way as before. Media from a-gateway to rtpengine is passed the same way as before. Media from rtpengine is passed with a different source port to b-provider. As this port was not announced in the callflow b-provider blocks the traffic. Any ideas how to fix this issue or force rtpengine to stay on the same port? Thanks Michael

1 0

P-CSCF not starting- Kamailio 5.3
by Lanvin ADON 04 Feb '20

04 Feb '20

Hi All, I'm installing an IMS core with Kamailio 5.3 release. The P-CSCF is not starting with the following error message. Feb 4 07:01:32 pcscf kamailio[13878]: 0(13880) DEBUG: <core> [core/sr_module.c:638]: find_mod_export_record(): export of <bind_ims_ipsec_pcscf> not found (flags 0) Feb 4 07:01:32 pcscf kamailio[13878]: 0(13880) ERROR: ims_registrar_pcscf [ims_registrar_pcscf_mod.c:268]: mod_init(): can't bind ims_ipsec_pcscf Feb 4 07:01:32 pcscf kamailio[13878]: 0(13880) ERROR: <core> [core/sr_module.c:849]: init_mod(): Error while initializing module ims_registrar_pcscf (/opt/pcscf/lib64/kamailio/modules/ims_registrar_pcscf.so) Feb 4 07:01:32 pcscf kamailio[13878]: ERROR: error while initializing modules For the configuration file, I'm using the example files in the source. -- Regards Lanvin ADON

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Call merging issues on cellular network
by Gaurav Bmotra 03 Feb '20

03 Feb '20

Hi I m using kamailio 5.3 I m trying to merge the call on cellular network but is not working. As on WiFi it is working fine . Any one can help me Thank you

2 1

Kamailio TLS server hello without any client certificate request
by Patrick Murphy 03 Feb '20

03 Feb '20

Hey guys - I'm using the following settings on my Kamailio config: modparam("tls", "low_mem_threshold2", 1024) modparam("tls", "tls_force_run", 1) modparam("tls", "tls_disable_compression", 1) modparam("tls", "ssl_max_send_fragment", 4096) modparam("tls", "renegotiation", 0) modparam("tls", "low_mem_threshold1", 2048) modparam("tls", "require_certificate", 1) When a client attempts to connect to Kamailio, the server hello it receives does *not* contain the client certificate request. And I've noticed that kamailio starts with: tls [tls_domain.c:694]: set_verification(): TLSs<default>: No client certificate required and no checks performed tls [tls_domain.c:320]: fill_missing(): TLSs<default>: require_certificate=0 Complete version output: kamailio 5.1.1 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. Any hints?

2 2

Dialog module - default_timeout behavior
by Gholamreza Sabery 03 Feb '20

03 Feb '20

Hi, According to the documentation default_timeout is "The default dialog timeout (in seconds), in the absence of a custom value provided in an AVP". There is also a "send_bye" option which can automatically send a BYE message on dialog timeout. Suppose that a user sends an INVITE and the call is successful (200). He talks more than default_timeout seconds (so there is not BYE for more than default_timeout seconds). This means that when the user actually sends BYE, the dialog is already terminated by the dialog module? Regards

3 4

sqlops multiple databases
by Ali Taher 03 Feb '20

03 Feb '20

Hello, Is there a way in sqlops module to use multiple postgres databases based on routing logic ? Here is what I'm using right now: modparam("sqlops","sqlcon","cb=>postgres://postgres:postgres@localhost:5432/db1") Thanks Ali Taher

2 1

Bad request when INVITE relayed to another kamailio
by Zhan Bazarov 31 Jan '20

31 Jan '20

We have Kamailio-cluster via route53(round-robin) some-domain.net we have two kamailio with public IP's phone1 is registered on kam1 phone2 is registered on kam2 when we are calling from phone1 to phone2 callflow looks: phone1 => kam1 => asterisk => kam1 => t_relay(address of second kamailio:5078) => kam2 => phone2 it works perfectly, but in case when we are using polycom as phone2 - we are getting 404 response from polycom... *Invite from second kamailio * 2020/01/20 10:31:21.799327 10.199.240.19:5078 -> 37.17.41.5:49811 INVITE sip:jyu3xsfkrz6c5qn@10.3.0.116;transport=tcp SIP/2.0 Record-Route: <sip:10.199.240.19:5078;transport=tcp;r2=on;lr;nat=yes;rtp=bridge> Record-Route: <sip:10.199.240.191:5078;r2=on;lr;nat=yes;rtp=bridge> Record-Route: <sip:10.199.240.135:5078;lr> Via: SIP/2.0/TCP some-domain.net:5078;branch=z9hG4bK9ca3.93c2345f3eb1d4b0e1244e722a9bfb6e.0 Via: SIP/2.0/UDP some-domain.net:5078;rport=5078;received=10.199.240.135;branch=z9hG4bK9ca3.89b66f1dd6e86a5922180b8ed8475072.0 Via: SIP/2.0/UDP 10.199.240.179:7060;received=10.199.240.179;rport=7060;branch=z9hG4bKPj269365e0-798d-404b-bb77-2ad78472905c From: "Penny" <sip:1015@10.199.240.179>;tag=ba402508-a640-409f-ba30-dffdfe499f43 To: <sip:jyu3xsfkrz6c5qn@10.199.240.135> Contact: <sip:asterisk@10.199.240.179:7060;alias=10.199.240.179~7060~1> Call-ID: c3a406ca-9ac7-423d-9697-06d0603f48d5 CSeq: 22619 INVITE Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER Supported: timer, replaces, norefersub Session-Expires: 1800 Min-SE: 90 P-Asserted-Identity: "Penny" <sip:1015@10.199.240.179> Max-Forwards: 68 User-Agent: Awesome Calling Platform 3.0 Content-Type: application/sdp Content-Length: 463 *Response from POLYCOM * 2020/01/20 10:31:22.054766 37.17.41.5:49811 -> 10.199.240.19:5078 SIP/2.0 400 Bad Request Via: SIP/2.0/TCP some-domain.net:5078;branch=z9hG4bK9ca3.93c2345f3eb1d4b0e1244e722a9bfb6e.0 Via: SIP/2.0/UDP some-domain.net:5078;rport=5078;received=10.199.240.135;branch=z9hG4bK9ca3.89b66f1dd6e86a5922180b8ed8475072.0 Via: SIP/2.0/UDP 10.199.240.179:7060;received=10.199.240.179;rport=7060;branch=z9hG4bKPj269365e0-798d-404b-bb77-2ad78472905c From: "Penny" <sip:1015@10.199.240.179>;tag=ba402508-a640-409f-ba30-dffdfe499f43 To: <sip:jyu3xsfkrz6c5qn@10.199.240.135>;tag=8BC58304-83D9B045 CSeq: 22619 INVITE Call-ID: c3a406ca-9ac7-423d-9697-06d0603f48d5 Record-Route: <sip:10.199.240.19:5078;transport=tcp;r2=on;lr;nat=yes;rtp=bridge>, <sip:10.199.240.19:5078;r2=on;lr;nat=yes;rtp=bridge>, <sip:10.199.240.235:5078;lr> User-Agent: PolycomVVX-VVX_450-UA/5.8.0.12848 Accept-Language: en Content-Length: 0 Any ideas how to fix it? -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

3 6

$vn vs $var
by Alex Balashov 31 Jan '20

31 Jan '20

Hi, I must have missed the discussion about it somewhere, but what is the inspiration for the new $vn variable container? According to the documentation, the only difference is that it can hold $null values, whereas variables initialised in $var to $null are coerced into an integer value of 0. But why was this a concern? Why is it necessary? Is it not confusing to users? Thanks, -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 1

rtpengine relay ip and port in avp
by Pafel 31 Jan '20

31 Jan '20

Thanks for the info Yuri! I tried several scenarios but without success. My sip proxy has ipv4 and ipv6 address. The other nodes use ipv4. When there is a request for a ipv6 UAC/customer comming to the proxy the rtpengine/sipwise is rewriting the sdp with the correct ipv6 address of the proxy. So I am trying to get the ipv6 address of the proxy from the sdp (which is always the same ipv6 address) and the port (which is always different). I've enabled modparam("rtpengine", "write_sdp_pv", "$avp(sdp)") And I have in request section: rtpengine_manage($avp(rtpproxy_offer_flags)); set_body("$avp(sdp)", "application/sdp"); xlog("L_INFO", "Request SDP $avp(sdp))\n"); And the problem is that I am always getting the ipv4 address and port but not the rewritten by the proxy ipv6 address and port. Any idea? Regards, Pavel

1 0

Re: [SR-Users] modifying via header from public IP to private IP
by vinod mn 31 Jan '20

31 Jan '20

Thank's david in my kamailio configuration I am using *listen=udp:192.168.131.69:5060 <http://192.168.131.69:5060/> advertise 221.20.20.12:5060 <http://221.20.20.12:5060/>* (as mentioned by you) and when the kamailio sends an invite to sip-gateway I am seeing this sip-gateway ip is *192.168.132.67 * and this gateway don't have any public-ip assigned .. Via: *SIP/2.0/UDP 221.20.20.12:5060 <http://221.20.20.12:5060/>;* branch=z9hG4bKaa6a.14c65f5675077ae7b0df70b02516979b.0 Via: SIP/2.0/UDP 192.168.113.112 :50386;received= 192.168.113.112 ;branch=z9hG4bK-d87543-550b984016109617-1--d87543-;rport=50386 I want private ip instead of public ip in the first via -- Thanks and regards Vinod.M.N

3 4

kamailio
by Genildo Nascimento 30 Jan '20

30 Jan '20

Someone managed installed kamailio with IPV6

4 5

modifying via header from public IP to private IP
by vinod mn 30 Jan '20

30 Jan '20

Hi , I have a cloud server, when I make call from a sip phone (registered with kamailio), in the INVITE header I am seeing the via header with public IP, is there any way that I can modify via header to send only the private IP. Please let me know, is it possible to replace. [image: image.png] the via headers that kamailio is sending to sip-gateway is sip-phone ip:192.168.113.112 Via: SIP/2.0/UDP *221.20.20.12:5060 <http://221.20.20.12:5060>* ;branch=z9hG4bKaa6a.14c65f5675077ae7b0df70b02516979b.0 Via: SIP/2.0/UDP 192.168.113.112 :50386;received= 192.168.113.112 ;branch=z9hG4bK-d87543-550b984016109617-1--d87543-;rport=50386 how to modify the first via header to have private-IP instead of public-IP.. I tried all the replace functions but it is not working. -- Thanks and regards Vinod.M.N

2 1

How to make a distinction between sip and tel uri
by Duarte Rocha 29 Jan '20

29 Jan '20

Greetings, I have a client that sends me two P-Asserted-Identity headers, one with SIP URI and one with tel URI. The pseudo variable $ai returns the topmost P-Asserted-Identity header and I only need the one with the SIP URI. How should i proceed ? Is there a way to delete the one with tel URI? the header removal functions i find only work with header name. I could also work with something different than $ai for this scenario. Best Regards,

2 1

Re-write contact header
by egemen ulus 29 Jan '20

29 Jan '20

Hi everyone, I have asterisk farm and kamailio which located front of asterisks farm as a proxy. When someone send a invite from outside to my kamailio (without record route), i forward it to my Asterisk farm. Then My one of Asterisk reply it as 200 ok. Then my kamailio forward this 200 ok to remote side with record route. But in my 200 ok message, contact header contains my Asterisk IP address. That is why remote side tries to send ACK to Asterisk. How to rewrite my contact header as my kamailio (SBC) address ? Actually I tried fixed nated contact and advertise ip but I couldn't make it worked. I am really new on that. Kamailio cfg file is attached. Please help! Egemen Get Outlook for Android<https://aka.ms/ghei36>

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