- sr-users - kamailio.org

Looking for a VoIP & Kamailio Dev-Ops
by Henning Westerholt 06 Jul '20

06 Jul '20

Dev-Ops VoIP & Kamailio GILAWA is looking for a new colleague to join our team! Tasks: Deploying, developing, and integrating our software based on Kamailio Testing and validation of our customer solutions Solve complex customer problems in the VoIP field Fix Bugs and create extensions in the Kamailio project Requirements: Good background in VoIP telephony and Kamailio Stack Experience with signalling standards (SIP, RTP etc..) and troubleshooting Knowledge and experience in one of the following languages: C/C++, Bash, PHP Experience with Debian or RedHat/CentOS Linux, knowledge of Ansible would be a plus Fluent in English Based in European or neighbouring time zone (e.g. EMEA area) Self-organized and strong communication skills Given the current situation this is a remote position. If you are interested, please send me a (direct) e-mail with an CV or Linkedin link, please include your availability and salary expectations as well. Best regards, Henning Westerholt -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com<https://gilawa.com/>

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Kamailio 5.3.5, usrloc: Error inserting location (integer out of range)
by Stefan 03 Jul '20

03 Jul '20

Hi Maillist Kamailio-Version: 5.3.5 built from source with postgres-db. I get an Error if I try to add a static location with kamctl, see my example: kamctl ul add 1234 sip:1234@10.10.10.10:5060 NULL NULL Log output: Jul 1 09:28:20 kamailio-srv02 /sbin/kamailio[3892]: ERROR: db_postgres [km_dbase.c:266]: db_postgres_submit_query(): 0x7fbe5398c148 PQsendQuery Error: ERROR: integer out of range#012 Query: insert into location (username,contact,expires,q,callid,cseq,flags,cflags,user_agent,path,methods,last_modified,ruid,reg_id,server_id,connection_id,keepalive,partition ) values ('1234','sip:1234@10.10.10.10:5060','1970-01-01 00:00:00',1.000000 ,'dfjrewr12386fd6-343(a)kamailio.rpc',1,0,0,'kamailio SIP Router - RPC Server','NULL',2147483648,'1970-01-01 00:00:00','ulcx-5ef99724-f37-1',0,0,0,0,22) truncated output of kamctl ul show: "AoRs": [{ "Info": { "AoR": "1234", "HashID": 924651761, "Contacts": [{ "Contact": { "Address": "sip:1234@10.10.10.10:5060", "Expires": "permanent", "Q": 1, "Call-ID": "dfjrewr12386fd6-343(a)kamailio.rpc", "CSeq": 1, "User-Agent": "kamailio SIP Router - RPC Server", "Received": "[not set]", "Path": "NULL", "State": "CS_NEW", "Flags": 0, "CFlags": 0, "Socket": "[not set]", "Methods": -2147483648, "Ruid": "ulcx-5ef99724-f37-1", "Instance": "[not set]", "Reg-Id": 0, "Server-Id": 0, "Tcpconn-Id": 0, "Keepalive": 0, "Last-Keepalive": 0, "Last-Modified": 0 } }] } }, { So the log tells me: integer out of range. I think it's this one: "Methods": -2147483648, This let's me thinking that this is a bug. Someone else? Regards Stefan

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403 Forbidden error You must register first with a S-CSCF
by Pavithra Mohanraja 03 Jul '20

03 Jul '20

Hi, I am facing below error when i try to make a call using pjsip .. Registration is successful. I am following the open 5gs VolTE setup. I have not changed any configurations in any of the config files. But I am facing below issue .. Can you please help me what could be the reason. *reason=403 (Forbidden - You must register first with a S-CSCF)* Could anyone please help me .. I am ready to provide anyother info if required . Thanks, Pavithra

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Error in SCSCF
by Pavithra Mohanraja 02 Jul '20

02 Jul '20

Hi , I am facing below error in s-cscf when i send a BYE from alice to bob in kamailio IMS. The problem is in only call termination .Kindly help . Error log in scscf : Jun 30 08:40:22 scscf kamailio[557]: DEBUG: <core> [core/mem/q_malloc.c:526]: qm_free(): freeing frag. 0x7f4746f89148 alloc'ed from core: core/rvalue.c: rval_new_empty(224) Jun 30 08:40:22 scscf kamailio[557]: DEBUG: <core> [core/socket_info.c:630]: grep_sock_info(): checking if host==us: 11==11 && [10.244.0.57] == [10.244.0.54] Jun 30 08:40:22 scscf kamailio[557]: DEBUG: <core> [core/socket_info.c:637]: grep_sock_info(): checking if port 4080 (advertise 0) matches port 5060 Jun 30 08:40:22 scscf kamailio[557]: DEBUG: <core> [core/name_alias.h:62]: grep_aliases(): matching (0:10.244.0.57:5060) vs. (0:sip.example.com:0) Jun 30 08:40:22 scscf kamailio[557]: DEBUG: <core> [core/name_alias.h:62]: grep_aliases(): matching (0:10.244.0.57:5060) vs. (0:sip.example.com:5060) Jun 30 08:40:22 scscf kamailio[557]: DEBUG: <core> [core/name_alias.h:62]: grep_aliases(): matching (0:10.244.0.57:5060) vs. (0: scscf.sip.example.com:4080) Jun 30 08:40:22 scscf kamailio[557]: DEBUG: <core> [core/forward.c:423]: check_self(): *host (0:10.244.0.57:5060 <http://10.244.0.57:5060>) != me* Jun 30 08:40:22 scscf kamailio[557]: ERROR: <script>: sip:bob at sip.example.com Jun 30 08:40:22 scscf kamailio[557]: ERROR: <script>: *Domain not served 403* Jun 30 08:40:22 scscf kamailio[557]: ERROR: <script>: *BYE *( sip:alice@sip.example.com (10.244.0.52:4060) to sip:bob@sip.example.com, -ARSjM.TTZgxhzwDjKRN0DtENWD4bKl0) Jun 30 08:40:22 scscf kamailio[557]: DEBUG: <core> [core/mem/q_malloc.c:374]: qm_malloc(): qm_malloc(0x7f4746dd7010, 30) called from core: core/ut.c: as_asciiz(161) I am not getting what it checks .. 10.244.0.57 -> UE bob 10.244.0.52 -> pcscf 10.244.0.53 -> icscf 10.244.0.54 -> scscf 10.244.0.56 -> UE alice Kindly help Thanks, Pavithra

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Sql_result_free() not cleaning properly
by Duarte Rocha 01 Jul '20

01 Jul '20

Greetings, I'm trying to clean a SQL result using "sql_result_free()" but i'm not having success. Can you please verify if i'm doing anything wrong here? This is my code : xerr("Pre clean [0] = Iterator 0 = $xavp(Cli_PICKER_Default_Search_Result[0]=>Code)"); sql_result_free("Cli_PICKER_Default_Search_Result"); xerr("Pos clean [0] = Iterator 0 = $xavp(Cli_PICKER_Default_Search_Result[0]=>Code)"); The result is the following : ERROR: <script>:Pre clean [0] = Iterator 0 = 111111111111 ERROR: <script>:Pos clean [0] = Iterator 0 = 111111111111 Am i using sql_result_free correctly or doing something wrong? Best Regards,

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Re: [SR-Users] [EXT] RE: kamailio 522 S-CSCF failing to start -UPDATE
by Martin W Woscek 30 Jun '20

30 Jun '20

Problem: The mysql db issues with the scscf db include: table version, disappearing impu and/or UE data retained in tables when none are registered Workarround: In order to get over those various mysql issues listed causing the S-CSCF not to come up properly as boot up, 1. create a fresh new scscf database (kamadbctl [drop|create] scscf and import the msql script from the scscf configs) 2. back up the new scscf db 3. modify the init.d startup script for the scscf to restore the fresh db upon each start/restart of the S-CSCF /etc/init.d/kamailio_scscf .. case "$1" in start|debug) mysql -u root --database=scscf < /etc/kamailio/ims_databases/scscf.sql The S-CSCF has been stable and dependable on each boot. _Martin From: Henning Westerholt <hw(a)skalatan.de> Sent: Monday, June 8, 2020 4:50 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Cc: Martin W Woscek <mwoscek(a)mitre.org> Subject: [EXT] RE: kamailio 522 S-CSCF failing to start Hello, the error message below is from the DB startup version check. So do you have sometimes DB errors on this virtual machine? 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 Martin W Woscek Sent: Monday, June 8, 2020 2:46 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org>> Subject: [SR-Users] kamailio 522 S-CSCF failing to start Hello, There are times that our S-CSCF module just wont start. I’ve tried correlating the issue with some operation such as not de-registering an IMS client (Boghe or imsdroid) prior to shutting down the Kamailio server. But this isn’t always the case. Being that we are running this an a Ubuntu Virtual Machine, simply overwriting the image with a backed up VM is the temporary solution. I’ve dropped and created the db for S (kamdbctl) and rearranged the order of loading the presence module with same result. Status of the S-CSCF daemon on failure to start: # systemctl status kamailio_scscf ● kamailio_scscf.service - LSB: Start the Kamailio SIP proxy server Loaded: loaded (/etc/init.d/kamailio_scscf; generated) Active: inactive (dead) Docs: man:systemd-sysv-generator(8) Jun 08 08:22:55 ubuntu /usr/local/sbin/kamailio[16511]: INFO: cdp [cdp_mod.c:255]: cdp_exit(): CDiameterPeer child stopping ... Jun 08 08:22:55 ubuntu /usr/local/sbin/kamailio[16511]: INFO: cdp [cdp_mod.c:257]: cdp_exit(): ... CDiameterPeer child stopped Jun 08 08:22:55 ubuntu /usr/local/sbin/kamailio[16511]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Jun 08 08:22:55 ubuntu kamailio_scscf[16495]: * already running Jun 08 08:22:55 ubuntu kamailio_scscf[16495]: ...done. Jun 08 08:22:55 ubuntu systemd[1]: Started LSB: Start the Kamailio SIP proxy server. Jun 08 08:24:03 ubuntu systemd[1]: Stopping LSB: Start the Kamailio SIP proxy server... Jun 08 08:24:03 ubuntu kamailio_scscf[16517]: * Stopping Kamailio IMS S-CSCF: kamailio_scscf Jun 08 08:24:03 ubuntu kamailio_scscf[16517]: ...done. Jun 08 08:24:03 ubuntu systemd[1]: Stopped LSB: Start the Kamailio SIP proxy server. root@ubuntu:~/IMS_manual_scripts# systemctl start kamailio_scscf root@ubuntu:~/IMS_manual_scripts# systemctl status kamailio_scscf ● kamailio_scscf.service - LSB: Start the Kamailio SIP proxy server Loaded: loaded (/etc/init.d/kamailio_scscf; generated) Active: active (exited) since Mon 2020-06-08 08:27:23 EDT; 58s ago Docs: man:systemd-sysv-generator(8) Process: 16543 ExecStart=/etc/init.d/kamailio_scscf start (code=exited, status=0/SUCCESS) Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: ERROR: <core> [db.c:457]: db_check_table_version(): invalid version 0 for table presentity found, expected 5 (check table structure and ta Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: ERROR: presence [presence.c:367]: mod_init(): error during table version check Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: ERROR: <core> [core/sr_module.c:898]: init_mod(): Error while initializing module presence (/usr/local/lib64/kamailio/modules/presence.so) Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: WARNING: <core> [core/mem/q_malloc.c:480]: qm_free(): WARNING: free(0) called from cdp_avp: cdp_avp_mod.c: cdp_avp_destroy(226) Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: INFO: cdp [cdp_mod.c:255]: cdp_exit(): CDiameterPeer child stopping ... Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: INFO: cdp [cdp_mod.c:257]: cdp_exit(): ... CDiameterPeer child stopped Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Jun 08 08:27:23 ubuntu kamailio_scscf[16543]: * already running Jun 08 08:27:23 ubuntu kamailio_scscf[16543]: ...done. Jun 08 08:27:23 ubuntu systemd[1]: Started LSB: Start the Kamailio SIP proxy server. Any insight is appreciated. Thanks, _Martin

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default config and options for Kamailio 5.4.x
by Daniel-Constantin Mierla 30 Jun '20

30 Jun '20

Hello, with the release of Kamailio 5.4.0 approaching, I opened an item in the tracker to collect suggestions about what should be tuned in the default config file. If you feel like something should be there, add your comment to: * https://github.com/kamailio/kamailio/issues/2381 Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

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Re: [SR-Users] RTP voice failover
by Yufei Tao 30 Jun '20

30 Jun '20

Hi, RTP FO is possible, e.g. with keepalived+rtpengine+Redis as people suggested. Here's the guide for configuring it which I find very useful: https://github.com/sipwise/rtpengine/wiki/Redis-keyspace-notifications A possible way is to have a pair of rtpengine instances in master-slave setup. When master goes down, RTP continues on slave (new master) after a few seconds gap, when the virtual IP(s) migrate over. Another useful link: https://www.kamailio.org/events/2016-KamailioWorld/Day2/20-Pawel.Kuzak-High… Good luck! Yufei <https://github.com/sipwise/rtpengine/wiki/Redis-keyspace-notifications>

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kamailio and rtpengine in docker
by David Villasmil 29 Jun '20

29 Jun '20

Hello guys, I've got a kamailio on a docker and kamailio on another docker on different boxes. The issue i'm having is that when kamailio tries to enable rtpengine it tries pinging it. The ping arrives at the actual container where rtpengine is running, but for some reason i haven't figured out, rtpengine (or the container) is not responding to the ping. I'm writing here in the hopes that someone has already implemented this an come across this and a solved (for sure) I'd appreciate some help. Many thanks! David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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Valid URL characters in the RURI
by Sam Ware 29 Jun '20

29 Jun '20

I'm using Sanity Checks. Is an underscore (_) not allowed in the host portion of the RURI? I'm on version 5.3. I reviewed the pase_uri function in the core parse_uri.c but I am not super proficient in C. case URI_HOST_P: switch(*p) { check_host_end; default: if(!isalnum(*p) && (*p != '.') && (*p != '-') && !uri_host_char_allowed(*p)) { goto error_bad_host; } } break; I see it calls out periods (.) and dashes (-) but not underscores. I'm wondering why, being underscores are allowed in URLs. -- Sam D Ware

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Changing Debug Level on the fly
by Sam Ware 29 Jun '20

29 Jun '20

I am trying to change the debug level without restarting the service. My research says this is right but I get an error message. $ sudo -u kamailio kamcmd dbg.set_mod_level core 2 error: 500 - cannot store parameter -- Sam D Ware

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SIP OPTIONS between UE and P-CSCF
by BALL SUN 29 Jun '20

29 Jun '20

Hi We have setup UE with Kamailio IMS, and after the registration is successful. The 2nd SIP REGISTER is sent from UE to P-CSCF through IPSEC tunnel. But we found that the SIP OPTIONS (I believed it is a health check) from P-CSCF to UE is not through the IPSEC tunnel. our question is 1. Should SIP OPTIONS run inside the IPSEC tunnel or it should be outside? 2. if it can be configured to run inside the IPSEC tunnel, what configuration can we change to achieve it? - RBK

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How to add parameter ;user=phone in RURI
by Minh Phan 28 Jun '20

28 Jun '20

Hi List, I'm having trouble adding the parameter user=phone in the RURI. I have tried to use the function add_uri_param from siputils module but it doesn't work (see below). Here is the code I use to add user=phone parameter to FROM, TO and RURI uac_replace_from("", "sip:$var(normalized_number)@sip.domain.com ;user=phone"); uac_replace_to("", "sip:$var(normalized_number)@sip.domain.com;user=phone"); $rU = $var(normalized_number); *add_uri_param*("user=phone"); Thanks in advance for any hints, Best regards, Minh Phan

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RTP voice failover
by Evgeniy 27 Jun '20

27 Jun '20

Is that possible to implement RTP voice fail-over? In case RTP node goes down - the call should not be interrupted. This is kinda strange - but client want to have super redundant system. For example use 2 RTP nodes and simultaneously send 2 RTP stream over 2 nodes (yes - network traffic will increase) Or maybe is a way to recover UDP connections after a crash and replace node state. Maybe use a Kubernetes tools for that purpose ? Here what i found in mailing-list - https://lists.kamailio.org//pipermail/sr-users/2015-December/091006.html Many thanks for any hints to this topic.

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Kamailio IMS with ENUM
by BALL SUN 26 Jun '20

26 Jun '20

Hi Is there any information that can help to setup ENUM query in Kamailio IMS? what we wanted to achieve is to when A in domain A (the local domain) makes an INVITE to B, it will check the domain of B from ENUM backend. if it is not on the local domain, it will route to remote IMS core. Thanks - RBK

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Does kamailio sip domain makes sense if i defined some alias
by PICCORO McKAY Lenz 26 Jun '20

26 Jun '20

in kamctlrc i defined an DOMAIN but later that ip was updated.. so my question: it make sense inclusivelly if already defined more allias in kamailiorc.cfg file? in case of more complicated response,, anybody can explain me about that why? -- Lenz McKAY Gerardo (PICCORO) http://qgqlochekone.blogspot.com

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Re: [SR-Users] [EXT] Re: Kamailio I-CSCF not sending SIP:200 OK messages to Asterisk (to tag) -RESOLVED
by Martin W Woscek 26 Jun '20

26 Jun '20

Hello, Just wanted to update what fixed our issue. The "contact_user" which is "asterisk" that we were using for the Kamailio context for outgoing SIP trunk/channel in the pjsip.conf file of the Asterisk 16 server was not a valid user in the IMS HSS database (FHoSS implementation). However not being an expert in the IMS, IMHO, the IMS should have kicked out this INVITE as soon as credentials were checked at the front door of the IMS, but rather Kamailio sent it from I-CSCF to S-CSCF to P-CSCF, which is the correct flow, until the P-CSCF sent it back to I-CSCF which then resulted in a 604 HSS user unknown being sent thru the reverse direction back to Asterisk. Anyway, its always good to end Friday on a high note. Thanks, _Martin From: sr-users <sr-users-bounces(a)lists.kamailio.org> On Behalf Of Martin W Woscek Sent: Wednesday, June 24, 2020 1:32 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org>; miconda(a)gmail.com Subject: Re: [SR-Users] [EXT] Re: Kamailio I-CSCF not sending SIP:200 OK messages to Asterisk (to tag) Just to clarify what we are seeing, here is a packet from the INVITE from the S-CSCF (6060) to the P-CSCF(4060). I have eliminated the S-CSCF as this packet looks correct. The P-CSCF is not handling the request URI correctly, and simply sends it back to the I-CSCF where I-CSCF cannot find the user in the HSS based on the ruri updated by the S-CSCF with the local IP address of the UE (192.168.20.1). The P-CSCF should send this INVITE to the UE (703222888(a)192.168.20.1<mailto:703222888@192.168.20.1>) No. Arrival Time Source Destination Source Port Destination Port SIP to tag SIP from tag Method Record-Route Userinfo Request-URI Media Port Protocol Length Request-URI Host Port Owner Username Media Type Media Format Origin-Host Type slice_type SIP from address User Part SIP from address Host Part SIP to address User Part SIP to address Host Part Contact URI User Part Contact URI Host Part Request-URI User Part Request-URI Host Part Request-URI Host Port Info 1553 Jun 17, 2020 09:31:21.722451703 Eastern Daylight Time 192.168.20.128 192.168.20.128 6060 4060 d490a30c-42a8-4fc0-9fe9-b7a21537386d INVITE mt sip:7032228888@192.168.20.1:63871;transport=udp 10626,13370 SIP/SDP 2263 63871 - audio,video ITU-T G.711 PCMU,DynamicRTP-Type-101,0,101,101,DynamicRTP-Type-99,99,99 7033331111 asterisk.irisims.org 7032228888 ims522.irisims.org asterisk 192.168.20.131 7032228888 192.168.20.1 63871 Request: INVITE sip:7032228888@192.168.20.1:63871;transport=udp | Frame 1553: 2263 bytes on wire (18104 bits), 2263 bytes captured (18104 bits) on interface any, id 0 Linux cooked capture Internet Protocol Version 4, Src: 192.168.20.128, Dst: 192.168.20.128 User Datagram Protocol, Src Port: 6060, Dst Port: 4060 Session Initiation Protocol (INVITE) Request-Line: INVITE sip:7032228888@192.168.20.1:63871;transport=udp SIP/2.0 Method: INVITE Request-URI: sip:7032228888@192.168.20.1:63871;transport=udp Request-URI User Part: 7032228888 Request-URI Host Part: 192.168.20.1 Request-URI Host Port: 63871 [Resent Packet: False] Message Header Record-Route: <sip:mt@192.168.20.128:6060;lr=on;ftag=d490a30c-42a8-4fc0-9fe9-b7a21537386d;did=7d2.e0f1> Route: <sip:term@pcscf522.irisims.org;lr> Via: SIP/2.0/UDP 192.168.20.128:6060;branch=z9hG4bK285b.2a57abaad57cfd55bc18a23d4e9cefc8.0 Via: SIP/2.0/UDP 192.168.20.128;branch=z9hG4bK285b.793ba7dcf9fcd18ddef225b83896340e.1 Via: SIP/2.0/UDP 192.168.20.131:5060;received=192.168.20.131;rport=5060;branch=z9hG4bKPjef451437-0a4f-4732-ac54-318740956335 From: <sip:7033331111@asterisk.irisims.org>;tag=d490a30c-42a8-4fc0-9fe9-b7a21537386d To: <sip:7032228888@ims522.irisims.org> Contact: <sip:asterisk@192.168.20.131:5060> Call-ID: 39c15c42-b0b7-48b5-b4bf-ac7a99fbe660 [Generated Call-ID: 39c15c42-b0b7-48b5-b4bf-ac7a99fbe660] CSeq: 19314 INVITE Route: <sip:ims522.irisims.org:5060;lr> Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER Supported: 100rel, timer, replaces, norefersub Session-Expires: 1800 Min-SE: 90 Max-Forwards: 68 User-Agent: Asterisk PBX 16.6.0 Content-Type: application/sdp Content-Length: 1121 Message Body From: sr-users <sr-users-bounces(a)lists.kamailio.org<mailto:sr-users-bounces@lists.kamailio.org>> On Behalf Of Martin W Woscek Sent: Wednesday, June 24, 2020 11:21 AM To: miconda(a)gmail.com<mailto:miconda@gmail.com>; Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org>> Subject: Re: [SR-Users] [EXT] Re: Kamailio I-CSCF not sending SIP:200 OK messages to Asterisk (to tag) Hi, The issue is the P should be sending to the UE not back to the I like the failed flow: Asterisk16-[phone] --->ICSCF--->S-CSCF--->P-CSCF--->I-CSCF ---> [604 HSS user unknown] to Asterisk16[phone] The successful call should look like: Asterisk16-[phone] --->ICSCF--->S-CSCF--->P-CSCF--->Kamailio-[UE] Does anyone have a example P-cscf scripts including the mt.cfg or other script file that make this work? The reverse calls work just fine. Thanks, _Martin From: Daniel-Constantin Mierla <miconda(a)gmail.com<mailto:miconda@gmail.com>> Sent: Wednesday, June 24, 2020 4:53 AM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org>>; Martin W Woscek <mwoscek(a)mitre.org<mailto:mwoscek@mitre.org>> Subject: [EXT] Re: [SR-Users] Kamailio I-CSCF not sending SIP:200 OK messages to Asterisk (to tag) Hello, Kamailio is not generating the 200ok for INVITE (calls), it just sends out what it received from end point, after consuming own Via header (of course, other headers can be changed based on config rules, but Kamailio is not in charge of setting To-tag). Cheers, Daniel On 19.06.20 21:19, Martin W Woscek wrote: Hello all, SIP/UE (boghe or imsdroid) client registered to Kamailio makes call to an Asterisk registered SIP phone is successful: [UE]-Kamailio--->INVITE--->Asterisk16-[phone] [UE]-Kamailio--->200 OK with to_tag--->Asterisk16-[phone] But in reverse direction for a call, Kamailio does not return the SIP OK so no to_tag is sent so call fails to ring and complete: [phone]-Asterisk16 --->INVITE--->Kamailio Where the UE device never receives the INVITE, Asterisk never gets and 200 OK message with the to_tag from the I-CSCF, the call flow itself gets lost in Kamailio, where the P-CSCF sends final INVITE to I-CSCF and and ultimately a 604 HSS user unknown message is sent back to Asterisk from the I-CSCF. Basically Im using the default "sample" configs for both the P and the I-CSCF. Our sauce is in the S-CSCF for out going calls that originate by a registered UE. Any insight or sample Kamailio configuration that Im lacking? Has anyone done this and could share the asterisk and Kamailio script snippets that make it possible. Thanks, _Martin _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla -- www.asipto.com<http://www.asipto.com> www.twitter.com/miconda<http://www.twitter.com/miconda> -- www.linkedin.com/in/miconda<http://www.linkedin.com/in/miconda> Funding: https://www.paypal.me/dcmierla

1 0

Question about textops
by Mojtaba 26 Jun '20

26 Jun '20

Hello, Some confusing things while developing in Kamailio: In is_method_f function in textops module, the code checks the HDR_CSEQ_F in msg also, and if the method name appears in CSEQ_F, The result is returned TRUE. These question will arise why in this function, the CSEQ field in message will check? if(parse_headers(msg, HDR_CSEQ_F, 0)!=0 || msg->cseq==NULL) { LM_ERR("cannot parse cseq header\n"); return -1; /* should it be 0 ?!?! */ } if(m->s==0) return (get_cseq(msg)->method_id&m->len)?1:-1; else return (get_cseq(msg)->method_id==METHOD_OTHER && get_cseq(msg)->method.len==m->len && (strncasecmp(get_cseq(msg)->method.s, m->s, m->len)==0))?1:-1; I think it would be nice to add new function like: is_cseq_method_f in module to avoid some problem understanding. -- --Mojtaba Esfandiari.S

2 2

rtpengine inserted random line into sdp
by Andrew Chen 26 Jun '20

26 Jun '20

Hi forum, I'm starting my rtpengine project and I'm facing a strange problem with rtpengine. I am seeing this in the SDP part of the INVITE: a=rtcp:52021 a=rtcp-mux 2001:470:7:3A7:0:0:0:2 a=direction:active a=oldmediaip:54.153.25.234 As you can see there is a random insert of the local kamailio IP in there and I'm have tried the following to remove it but failed to succeed: - Remove in the offer the combination of ICE and replace-session-connection flags - Switched between rtpengine_offer and rtpengine_manage - Restarted ngcp-rtpengine-daemon process - Used sdpops function sdp_remove_line_by_preifx() Currently running Kamailio 5.1.2. I am running out of ideas. Any suggestions would be greatly appreciated. -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

2 4

Advertise concept
by Social Boh 25 Jun '20

25 Jun '20

Hello, maybe i don't understood completely the use of Advertise parameter. We use it only if Kamailio is behind a NAT and Kamailio have to comunicate over is PublicIP with provider, other SIP proxy and PBX but never to accept Users REGISTER. Is it right? Regards -- --- I'm SoCIaL, MayBe

2 3

Server error on LIR select next S-CSCF
by BALL SUN 25 Jun '20

25 Jun '20

Hi I have a setup using Kamailio IMS, and when trying to make a SIP call between clients, S-CSCF returns " SIP/2.0 500 Server error on LIR select next S-CSCF", do you have any idea on how to fix? - RBK

2 5

kamailio - 5.3.2 - Possible memleak in Core Module
by José Seabra 25 Jun '20

25 Jun '20

Hello there. During my stress tests against our kamailio servers I detected that it was running out of shm memory. edge-sip-proxy[6119]: ERROR: ESP_LOG: 10405249-1580(a)10.225.121.206: <core> [core/mem/q_malloc.c:298]: qm_find_free(): qm_find_free(0x7ff370920000, 4224); Free fragment not found! edge-sip-proxy[6119]: ERROR: ESP_LOG: 10405249-1580(a)10.225.121.206: <core> [core/mem/q_malloc.c:432]: qm_malloc(): qm_malloc(0x7ff370920000, 4224) called from core: core/sip_msg_clone.c: sip_msg_shm_clone(496), module: core; Free fragment not found! edge-sip-proxy[6119]: ERROR: ESP_LOG: 10405249-1580(a)10.225.121.206: <core> [core/sip_msg_clone.c:499]: sip_msg_shm_clone(): could not allocate shared memory from shm pool edge-sip-proxy[6119]: ERROR: ESP_LOG: 10405249-1580(a)10.225.121.206: tm [t_lookup.c:1293]: new_t(): out of mem: edge-sip-proxy[6119]: ERROR: ESP_LOG: 10405249-1580(a)10.225.121.206: tm [t_lookup.c:1439]: t_newtran(): new_t failed edge-sip-proxy[6119]: ERROR: ESP_LOG: 10405249-1580(a)10.225.121.206: sl [sl_funcs.c:392]: sl_reply_error(): stateless error reply used: I'm terribly sorry, server error occurred (1/SL) The output of kamctl stats shmem confirms that kamailio is running out of memory: { "jsonrpc": "2.0", "result": [ "shmem:fragments = 36427", "shmem:free_size = 14123552", "shmem:max_used_size = 268435456", "shmem:real_used_size = 254311904", "shmem:total_size = 268435456", "shmem:used_size = 215787904" ], "id": 10901 } Then I was check what was the module that were consuming more memory with kamcmd mod.stats all shm and the output of this command showed me that Core module was consuming most of shm memory, as you can see bellow is the output of this command shows that Core - build_req_buf_from_sip_req is consuming most of shm. Module: core { sip_msg_shm_clone(496): 4872856 create_avp(175): 24768 msg_lump_cloner(986): 154376 xavp_new_value(106): 780896 build_req_buf_from_sip_req(2187): 183984624 counters_prefork_init(211): 36864 cfg_clone_str(130): 96 cfg_shmize(217): 848 main_loop(1303): 8 init_pt(106): 8 init_pt(105): 8 init_pt(104): 5256 register_timer(995): 232 cfg_register_ctx(47): 64 init_tcp(5021): 8192 init_tcp(5015): 32768 init_tcp(5007): 8 init_tcp(5000): 8 init_tcp(4993): 8 init_tcp(4987): 8 init_tcp(4975): 8 init_avps(90): 8 init_avps(89): 8 init_dst_blacklist(438): 16384 init_dst_blacklist(430): 8 timer_alloc(498): 96 init_dns_cache(361): 8 init_dns_cache(352): 16384 init_dns_cache(344): 16 init_dns_cache(336): 8 init_timer(267): 8 init_timer(266): 16384 init_timer(265): 8 init_timer(264): 8 init_timer(253): 8 init_timer(221): 8 init_timer(210): 278544 init_timer(209): 8 init_timer(197): 8 cfg_child_cb_new(829): 64 sr_cfg_init(361): 8 sr_cfg_init(354): 8 sr_cfg_init(347): 8 sr_cfg_init(335): 8 sr_cfg_init(323): 8 qm_shm_lock_init(1202): 8 Total: 190229920 } Then I stopped the sipp that was running against kamailio in order to see if the memory consumed by *build_req_buf_from_sip_req* decreased but it didn't, so, it seems that there is a memleak in this version. If the information that I'm sending in this email is not enough to identify the root cause of this, please let me know if there is anything else that I can do here to help identify better what is the main reason. Than you Best Regards -- Cumprimentos José Seabra

2 2

siptrace: methods blacklist modparam
by Nuno Ferreira 25 Jun '20

25 Jun '20

Hello, We are a heavy Homer/Hepic (tks QXIP team) users and on our setup, there are some SIP methods we don’t want to get into the database or send to the capture server because there’s no value-added for call troubleshooting purposes (OPTIONS, KDMQ, HTTP (from JSONRPC)…) Typically we achieve the message filtering using siptrace in “manual” mode and control what to send from the script. This works fine for some situations, but there are a few we need to have the automatic mode on: - topos enabled: using manual mode we don’t get the “wire” version of the signaling - TCP connections: there are a few situations where the destination port is traced with 0 - I believe this is due to the fact the TCP socket not established when the message is collected There are other technics to filter like using a HEP relay proxy dropping unwanted traffic, but having a native kamailio functionality would make things much easier. With that in mind, I’ve created a PR (#2374 <https://github.com/kamailio/kamailio/pull/2374>) that allows to blacklist methods we don’t want to send to the capture server when using automatic trace mode. I believe this is a very helpful change for the community as well. To block SIP OPTIONS and KDMQ we just need to declare a new modparam: modparam("siptrace", "trace_on", 1) modparam("siptrace", "methods_blacklist_auto", 16896) Let me know what you think about it. Nuno -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

2 1

Re: [SR-Users] Kamailio-5.3.3. NAT and RTP not working
by supreeth herle 25 Jun '20

25 Jun '20

Please try to keep the mailing list in loop so that it helps others as well facing this issue. I believe this was resolved in 5.3.3 branch not sure why you are facing this issue. Anyways, make sure that modules loaded in kamailio.cfg of PCSCF is in following order loadmodule "ims_dialog" loadmodule "ims_usrloc_pcscf" #!ifdef WITH_IPSEC loadmodule "ims_ipsec_pcscf" #!endif loadmodule "ims_registrar_pcscf" Best Regards, Supreeth On Wed, 24 Jun 2020 at 13:36, Pavithra Mohanraja <pavimohan3004(a)gmail.com> wrote: > Hi Supreeth, > > I am getting the below error > > ms_usrloc_pcscf > [/usr/lib/x86_64-linux-gnu/kamailio/modules/ims_usrloc_pcscf.so] > 0(455) DEBUG: ims_registrar_pcscf [ims_registrar_pcscf_mod.c:264]: > mod_init(): Successfully bound to PCSCF Usrloc module > 0(455) DEBUG: <core> [core/sr_module.c:632]: find_mod_export_record(): > found export of <bind_ims_ipsec_pcscf> in module ims_ipsec_pcscf > [/usr/lib/x86_64-linux-gnu/kamailio/modules/ims_ipsec_pcscf.so] > 0(455) ERROR: ims_ipsec_pcscf [cmd.c:85]: bind_ipsec_pcscf(): > configuration error - trying to bind to ipsec pscscf module before being > initialized > 0(455) ERROR: <core> [core/sr_module.c:849]: init_mod(): Error while > initializing module ims_registrar_pcscf > (/usr/lib/x86_64-linux-gnu/kamailio/modules/ims_registrar_pcscf.so) > ERROR: error while initializing modules > 0(455) ERROR: ims_ipsec_pcscf [ims_ipsec_pcscf_mod.c:305]: mod_destroy(): > Error destroying spi generator > 0(455) ERROR: ims_ipsec_pcscf [ims_ipsec_pcscf_mod.c:309]: mod_destroy(): > Error destroying port generator > 0(455) DEBUG: tm [t_funcs.c:84]: tm_shutdown(): start > 0(455) DEBUG: tm [t_funcs.c:87]: tm_shutdown(): emptying hash table > 0(455) DEBUG: tm [t_funcs.c:89]: tm_shutdown(): removing semaphores > 0(455) DEBUG: tm [t_funcs.c:91]: tm_shutdown(): destroying tmcb lists > > > Kindly help > > On Wed, Jun 24, 2020 at 5:03 PM supreeth herle <herlesupreeth(a)gmail.com> > wrote: > >> There is no other IPSec configuration needed apart from running Kamailio >> P-CSCF and UE sending a SIP Register with Security-Client information for >> IPSec creation. Kamailio IMS IPSec module creates IPSec tunnels using >> Netlink library. >> >> Best Regards, >> Supreeth >> >> On Wed, 24 Jun 2020 at 13:29, Pavithra Mohanraja <pavimohan3004(a)gmail.com> >> wrote: >> >>> Hi , >>> >>> Could you please tell how to configure ipsec (general configuration not >>> related to kamailio). >>> >>> Thanks, >>> Pavithra >>> >>> On Wed, Jun 24, 2020 at 2:12 PM supreeth herle <herlesupreeth(a)gmail.com> >>> wrote: >>> >>>> reason=404 (Not Found - destination user not found on this S-CSCF)] >>>>> >>>> >>>> Please check your IMPU for the IMS user in the IMS HSS (e.g. FHoSS or >>>> any other HSS you are using to enter IMS user details). SCSCF is not able >>>> to find a registered user with that IMPU (i am assuming that you are >>>> getting this response while calling) >>>> >>>> Increase the debug level on S-CSCF and check out which IMPU is being >>>> referred by SCSCF >>>> >>>> Best Regards, >>>> Supreeth >>>> >>>> On Wed, 24 Jun 2020 at 10:26, Pavithra Mohanraja < >>>> pavimohan3004(a)gmail.com> wrote: >>>> >>>>> Hi, >>>>> >>>>> I am facing below error in kamailio-ims 5.3.4 version. >>>>> >>>>> reason=404 (Not Found - destination user not found on this S-CSCF)] >>>>> >>>>> could you please help me how should i proceed with and where the >>>>> registration details will be stored in kamailio ims. >>>>> >>>>> It fails at this line in kamailio.cfg file in scscf . >>>>> >>>>> route[FINAL_TERM] { >>>>> if (lookup("location")) { >>>>> if (uri==myself) { >>>>> if (!t_newtran()) { >>>>> sl_reply_error(); >>>>> exit; >>>>> } >>>>> xlog("Destination Not Found ----\n"); >>>>> t_reply("404","Not Found - destination user >>>>> not found on this S-CSCF"); >>>>> exit; >>>>> } >>>>> >>>>> here lookup(location) means what it means. Could you please help me >>>>> >>>>> >>>>> >>>>> Thanks, >>>>> Pavithra >>>>> >>>>> On Thu, May 14, 2020 at 9:59 PM supreeth herle < >>>>> herlesupreeth(a)gmail.com> wrote: >>>>> >>>>>> Hello Pavithra, >>>>>> >>>>>> I have written a guide for how to integrate an IMS using Kamailio >>>>>> with open5gs EPC, here is the link >>>>>> https://open5gs.org/open5gs/docs/tutorial/02-VoLTE-setup/. You can >>>>>> find steps to setup rtpengine in that guide. >>>>>> >>>>>> Regarding requirement of rtpengine, its mandatory if you want to use >>>>>> the scripts in that example folder without modification. Also, read the >>>>>> READE of sipwise/rtpengine github repo, its very informative (in a gist >>>>>> rtpengine offer much more feature than rtpproxy) >>>>>> >>>>>> why can't we have sip client to sip client rtps flowing like how it >>>>>>> is happening without NAT. >>>>>>> >>>>>> >>>>>> For this, i would definitely suggest to read about SIP working. This >>>>>> is a perfect link https://github.com/onmyway133/blog/issues/284 >>>>>> which offers extensive material (short and very interesting to read with >>>>>> examples) related to SIP. >>>>>> >>>>>> Best Regards, >>>>>> Supreeth >>>>>> >>>>>> >>>>>> On Thu, 14 May 2020 at 17:51, Pavithra Mohanraja < >>>>>> pavimohan3004(a)gmail.com> wrote: >>>>>> >>>>>>> Hi , >>>>>>> >>>>>>> I didn't configure rtpengine ..instead of that only I used >>>>>>> rrpproxy..will it not work? >>>>>>> >>>>>>> If not please tell me how to configure rtpengine and also tell me if >>>>>>> it is really madatory .. why can't we have sip client to sip client rtps >>>>>>> flowing like how it is happening without NAT. >>>>>>> >>>>>>> That's my doubt ..please clarify .. >>>>>>> >>>>>>> Thanks. >>>>>>> >>>>>>> On Thu, May 14, 2020, 8:57 PM supreeth herle < >>>>>>> herlesupreeth(a)gmail.com> wrote: >>>>>>> >>>>>>>> HI, >>>>>>>> >>>>>>>> Hold on a moment, >>>>>>>> >>>>>>>> >>>>>>>>> 10.40.10.3 - pcscf in port 4060 (pcscf with NAT enabled in >>>>>>>>> pcscf.cfg (!#define WITH_NAT) and configured rtpproxy) >>>>>>>>> 10.40.10.12 - icscf in port 4070 >>>>>>>>> 10.40.10.5 - scscf in port 4080 >>>>>>>>> 10.40.10.30 - hss >>>>>>>>> >>>>>>>> >>>>>>>> So i assume you are using the examples provided in the ims folder >>>>>>>> of kamailio repo right? Isn't it using RTPEngine and not RTPProxy? Or have >>>>>>>> you modified the P-CSCF configuration to work with RTPProxy? >>>>>>>> >>>>>>>> Best regards, >>>>>>>> Supreeth >>>>>>>> >>>>>>>> On Thu, 14 May 2020 at 16:39, Pavithra Mohanraja < >>>>>>>> pavimohan3004(a)gmail.com> wrote: >>>>>>>> >>>>>>>>> Hi , >>>>>>>>> I have attached working rtp pcap file. But it is getting reverted >>>>>>>>> back to me . >>>>>>>>> >>>>>>>>> As you said , i have tried restarting rtpproxy several times. but >>>>>>>>> it didnt work . >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> On Thu, May 14, 2020 at 7:11 PM supreeth herle < >>>>>>>>> herlesupreeth(a)gmail.com> wrote: >>>>>>>>> >>>>>>>>>> Thanks for the pcap. Sorry, to ask this but can you also send >>>>>>>>>> pcap of scenario where you see RTP packets? >>>>>>>>>> >>>>>>>>>> Also, did you restart the rtpproxy after altering the >>>>>>>>>> /etc/default/rtpproxy ? >>>>>>>>>> >>>>>>>>>> i.e. >>>>>>>>>> >>>>>>>>>> $ systemctl restart rtpproxy >>>>>>>>>> >>>>>>>>>> Best Regards, >>>>>>>>>> Supreeth >>>>>>>>>> >>>>>>>>>> On Thu, 14 May 2020 at 14:51, Pavithra Mohanraja < >>>>>>>>>> pavimohan3004(a)gmail.com> wrote: >>>>>>>>>> >>>>>>>>>>> Hi Supreeth, >>>>>>>>>>> I have attached pcap files . kindly check and let me know. >>>>>>>>>>> filter of rtp packets and check. >>>>>>>>>>> >>>>>>>>>>> *filter : sip || rtp* >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> On Thu, May 14, 2020 at 5:15 PM supreeth herle < >>>>>>>>>>> herlesupreeth(a)gmail.com> wrote: >>>>>>>>>>> >>>>>>>>>>>> Hi Pavithra, >>>>>>>>>>>> >>>>>>>>>>>> Thanks for the clarification. Can you please attach a pcap of >>>>>>>>>>>> the call where you are not seeing the RTP packets? And, btw is the call >>>>>>>>>>>> getting dropped or something in that scenario? >>>>>>>>>>>> >>>>>>>>>>>> Regards, >>>>>>>>>>>> Supreeth >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> On Thu, 14 May 2020 at 13:34, Pavithra Mohanraja < >>>>>>>>>>>> pavimohan3004(a)gmail.com> wrote: >>>>>>>>>>>> >>>>>>>>>>>>> Hi Supreeth, >>>>>>>>>>>>> >>>>>>>>>>>>> I have kamailio ims installed in 4 vms inside openstack. >>>>>>>>>>>>> >>>>>>>>>>>>> 10.40.10.3 - pcscf in port 4060 (pcscf with NAT enabled in >>>>>>>>>>>>> pcscf.cfg (!#define WITH_NAT) and configured rtpproxy) >>>>>>>>>>>>> 10.40.10.12 - icscf in port 4070 >>>>>>>>>>>>> 10.40.10.5 - scscf in port 4080 >>>>>>>>>>>>> 10.40.10.30 - hss >>>>>>>>>>>>> >>>>>>>>>>>>> so all these vms are running in same subnet. >>>>>>>>>>>>> I have a floating ip associated to pcscf. >>>>>>>>>>>>> Now I have two zoiper clients outside openstack >>>>>>>>>>>>> zoiper1 - 10.0.2.15 (NAT Enabled Network adaptor) >>>>>>>>>>>>> zoiper2 - 10.0.2.15 >>>>>>>>>>>>> >>>>>>>>>>>>> when i am making a call from zoiper1 to zoiper2, call is going >>>>>>>>>>>>> properly . SIP messages are coming properly in wireshark packets. >>>>>>>>>>>>> when i filter for RTP packets in wireshark, it is not coming. >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> when i have the zoiper clients inside openstack itself ,i can >>>>>>>>>>>>> able to see RTP Packets in wireshark between two sip clients without any >>>>>>>>>>>>> NAT related configurations done. >>>>>>>>>>>>> >>>>>>>>>>>>> Kindly help me. >>>>>>>>>>>>> Thanks. >>>>>>>>>>>>> >>>>>>>>>>>>> On Thu, May 14, 2020 at 4:49 PM supreeth herle < >>>>>>>>>>>>> herlesupreeth(a)gmail.com> wrote: >>>>>>>>>>>>> >>>>>>>>>>>>>> Can you explain some more when you mean by "RTP alone is not >>>>>>>>>>>>>> coming."? And, how are you checking/verifying that? >>>>>>>>>>>>>> >>>>>>>>>>>>>> Regards, >>>>>>>>>>>>>> Supreeth >>>>>>>>>>>>>> >>>>>>>>>>>>>> On Thu, 14 May 2020 at 13:12, Pavithra Mohanraja < >>>>>>>>>>>>>> pavimohan3004(a)gmail.com> wrote: >>>>>>>>>>>>>> >>>>>>>>>>>>>>> Hi , >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> Yes I have configured as you mentioned only . In my >>>>>>>>>>>>>>> pcscf only i have rtpproxy also configured. >>>>>>>>>>>>>>> Floating IP i have given in listen address also as >>>>>>>>>>>>>>> listen = 10.40.10.3:4060 advertise 10.45.4.22:4060 >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> 10.40.10.3 - private ip >>>>>>>>>>>>>>> 10.45.4.22 - floating ip >>>>>>>>>>>>>>> Still RTP alone is not coming. >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> Is it the only way of configuring NAT ? >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> Thanks. >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> On Thu, May 14, 2020 at 4:37 PM supreeth herle < >>>>>>>>>>>>>>> herlesupreeth(a)gmail.com> wrote: >>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> Hello Pavithra, >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> Have you tried advertising the floating IP of the VM >>>>>>>>>>>>>>>> running RTPProxy as follows? >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> Edit /etc/default/rtpproxy file as follows: >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> # Additional options that are passed to the daemon. >>>>>>>>>>>>>>>> EXTRA_OPTS="-l 172.24.15.30 -d DBUG:LOG_LOCAL0" >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> here, "-l <PUBLIC_IP>" >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> Hope it helps. >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> Regards, >>>>>>>>>>>>>>>> Supreeth >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> On Thu, 14 May 2020 at 13:01, Pavithra Mohanraja < >>>>>>>>>>>>>>>> pavimohan3004(a)gmail.com> wrote: >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> Hi , >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> Could anybody please give me the approach how to configure >>>>>>>>>>>>>>>>> NAT in kamailio IMS. >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> I have installed kamailio IMS in openstack environment >>>>>>>>>>>>>>>>> which has private ips for all four vms (each component in one vm). >>>>>>>>>>>>>>>>> pcscf has one floating IP attached. I have enabled "WITH_NAT" and >>>>>>>>>>>>>>>>> configured rtpproxy . >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> when i make a call from outside of openstack using 2 >>>>>>>>>>>>>>>>> zoiper clients with outbound proxy keeping floating ip of pcscf, I am >>>>>>>>>>>>>>>>> getting 200 ok with all SIP packets correctly flowing but there is no RTP >>>>>>>>>>>>>>>>> Packet. >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> When i try the same within openstack environment without >>>>>>>>>>>>>>>>> using public ip, I am getting RTP Packets. >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> Kindly tell me the procedure of NAT Enablement or do i >>>>>>>>>>>>>>>>> need to do anything other than this. >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> Kindly help because i am stuck with this for long time. >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> Thanks. >>>>>>>>>>>>>>>>> _______________________________________________ >>>>>>>>>>>>>>>>> Kamailio (SER) - Users Mailing List >>>>>>>>>>>>>>>>> sr-users(a)lists.kamailio.org >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> _______________________________________________ >>>>>>>>>>>>>>>> Kamailio (SER) - Users Mailing List >>>>>>>>>>>>>>>> sr-users(a)lists.kamailio.org >>>>>>>>>>>>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>> _______________________________________________ >>>>>>>>>>>>>>> Kamailio (SER) - Users Mailing List >>>>>>>>>>>>>>> sr-users(a)lists.kamailio.org >>>>>>>>>>>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>>>>>>>>>>>>> >>>>>>>>>>>>>> _______________________________________________ >>>>>>>>>>>>>> Kamailio (SER) - Users Mailing List >>>>>>>>>>>>>> sr-users(a)lists.kamailio.org >>>>>>>>>>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>>>>>>>>>>>> >>>>>>>>>>>>> _______________________________________________ >>>>>>>>>>>>> Kamailio (SER) - Users Mailing List >>>>>>>>>>>>> sr-users(a)lists.kamailio.org >>>>>>>>>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>>>>>>>>>>> >>>>>>>>>>>> _______________________________________________ >>>>>>>>>>>> Kamailio (SER) - Users Mailing List >>>>>>>>>>>> sr-users(a)lists.kamailio.org >>>>>>>>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>>>>>>>>>> >>>>>>>>>>> _______________________________________________ >>>>>>>>>>> Kamailio (SER) - Users Mailing List >>>>>>>>>>> sr-users(a)lists.kamailio.org >>>>>>>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>>>>>>>>> >>>>>>>>>> _______________________________________________ >>>>>>>>>> Kamailio (SER) - Users Mailing List >>>>>>>>>> sr-users(a)lists.kamailio.org >>>>>>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>>>>>>>> >>>>>>>>> _______________________________________________ >>>>>>>>> Kamailio (SER) - Users Mailing List >>>>>>>>> sr-users(a)lists.kamailio.org >>>>>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>>>>>>> >>>>>>>> _______________________________________________ >>>>>>>> Kamailio (SER) - Users Mailing List >>>>>>>> sr-users(a)lists.kamailio.org >>>>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>>>>>> >>>>>>> _______________________________________________ >>>>>>> Kamailio (SER) - Users Mailing List >>>>>>> sr-users(a)lists.kamailio.org >>>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>>>>> >>>>>> _______________________________________________ >>>>>> Kamailio (SER) - Users Mailing List >>>>>> sr-users(a)lists.kamailio.org >>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>>>> >>>>>

2 5

Re: [SR-Users] [EXT] Re: Kamailio I-CSCF not sending SIP:200 OK messages to Asterisk (to tag)
by Martin W Woscek 24 Jun '20

24 Jun '20

Hi, The issue is the P should be sending to the UE not back to the I like the failed flow: Asterisk16-[phone] --->ICSCF--->S-CSCF--->P-CSCF--->I-CSCF ---> [604 HSS user unknown] to Asterisk16[phone] The successful call should look like: Asterisk16-[phone] --->ICSCF--->S-CSCF--->P-CSCF--->Kamailio-[UE] Does anyone have a example P-cscf scripts including the mt.cfg or other script file that make this work? The reverse calls work just fine. Thanks, _Martin From: Daniel-Constantin Mierla <miconda(a)gmail.com> Sent: Wednesday, June 24, 2020 4:53 AM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org>; Martin W Woscek <mwoscek(a)mitre.org> Subject: [EXT] Re: [SR-Users] Kamailio I-CSCF not sending SIP:200 OK messages to Asterisk (to tag) Hello, Kamailio is not generating the 200ok for INVITE (calls), it just sends out what it received from end point, after consuming own Via header (of course, other headers can be changed based on config rules, but Kamailio is not in charge of setting To-tag). Cheers, Daniel On 19.06.20 21:19, Martin W Woscek wrote: Hello all, SIP/UE (boghe or imsdroid) client registered to Kamailio makes call to an Asterisk registered SIP phone is successful: [UE]-Kamailio--->INVITE--->Asterisk16-[phone] [UE]-Kamailio--->200 OK with to_tag--->Asterisk16-[phone] But in reverse direction for a call, Kamailio does not return the SIP OK so no to_tag is sent so call fails to ring and complete: [phone]-Asterisk16 --->INVITE--->Kamailio Where the UE device never receives the INVITE, Asterisk never gets and 200 OK message with the to_tag from the I-CSCF, the call flow itself gets lost in Kamailio, where the P-CSCF sends final INVITE to I-CSCF and and ultimately a 604 HSS user unknown message is sent back to Asterisk from the I-CSCF. Basically Im using the default "sample" configs for both the P and the I-CSCF. Our sauce is in the S-CSCF for out going calls that originate by a registered UE. Any insight or sample Kamailio configuration that Im lacking? Has anyone done this and could share the asterisk and Kamailio script snippets that make it possible. Thanks, _Martin _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla -- www.asipto.com<http://www.asipto.com> www.twitter.com/miconda<http://www.twitter.com/miconda> -- www.linkedin.com/in/miconda<http://www.linkedin.com/in/miconda> Funding: https://www.paypal.me/dcmierla

1 1

Kamailio IMS using returned SRV records for transport slelection
by BALL SUN 24 Jun '20

24 Jun '20

Hi All I am new to Kamailio IMS, and I just setup a testing environment where having P-CSCF, I-CSCF and S-CSCF. I would like to use make below works 1. SIP client <-> P-CSCF (either UDP or TCP, based on the transport tag) 2. P-CSCF <-> I-CSCF (TCP by default) 3. I-CSCF <-> S-CSCF (TCP by default) For point 2, I observed that P-CSCF actually will send DNS SRV query to DNS server for resolving _sip._udp. and _sip._tcp. I believed we can just disable the "_sip._udp." so that P-CSCF will always use TCP to communication wih I-CSCF. However, for point 3, I didnt notice I-CSCF send DNS SRV query, My question is how we can force I-CSCF to do that or is there any configuration that we should enable such that I-CSCF always use TCP to S-CSCF? - RBK

2 2

SECFILTER reload from Siremis
by Social Boh 24 Jun '20

24 Jun '20

Hello list, is posible to reload SECFILTER data directly from Siremis? or the only way is: kamcmd secfilter.reload Regards -- --- I'm SoCIaL, MayBe

3 3

dialog's track_cseq_updates seems to break uac_auth() in this configuration
by George Diamantopoulos 24 Jun '20

24 Jun '20

Hello again, This might turn out to be a bug, but I'm posting here first because my config is fairly complicated and it might not be the sanest it could be. The problem manifests as follows: with the sole change being the enabling of track_cseq_updates for the dialog module, kamailio fails to process the 100 Trying it receives following authentication initiated by the uac_auth() function. Without the track_cseq_updates option enabled, the call proceeds successfully, albeit with the CSeq for the auth-carrying INVITE having the same value as the original. Here's a simplified version of the configuration routes involved in this: request_route { ... route(DISPATCH_PROVIDER); ... } route[DISPATCH_PROVIDER] { if ( !ds_select_dst("10", "8") ) { t_send_reply("503", "Downstream carrier unavailable"); exit; } t_on_branch_failure("DISPATCH_PROVIDER_FAILOVER"); t_on_branch("PROVIDER_BRANCH"); t_on_failure("DISPATCH_PROVIDER_FAILURE"); route(RELAY); exit; } failure_route[DISPATCH_PROVIDER_FAILURE] { if (t_is_canceled()) { exit; } if ( t_check_status("401|407") ) { $avp(arealm) = "authrealm.com"; $avp(auser) = "authusername"; $avp(apass) = "verys3kr1t"; uac_auth(); t_on_branch("PROVIDER_BRANCH"); t_on_branch_failure("DISPATCH_PROVIDER_FAILOVER"); t_on_failure("DISPATCH_PROVIDER_FAILURE"); $du = "sip:" + $T_rpl($si) + ":" + $T_rpl($sp); // I'll explain why this is here later in this e-mail t_relay(); exit; } } branch_route[PROVIDER_BRANCH] { uac_replace_from("$dlg_var(my_new_from)"); uac_replace_to("$dlg_var(my_new_to)"); $rU = "my new RURI user"; } event_route[tm:branch-failure:DISPATCH_PROVIDER_FAILOVER] { if (t_is_canceled()) { exit; } if ( t_check_status("401|407") ) { return; # next DST - only for 5xx or local timeout } else if ( t_check_status("5[0-9][0-9]") || (t_branch_timeout() && !t_branch_replied()) ) { if ( ds_next_dst() ) { t_on_branch("PROVIDER_BRANCH"); t_on_branch_failure("DISPATCH_PROVIDER_FAILOVER"); t_on_failure("DISPATCH_PROVIDER_FAILURE"); route(RELAY); exit; } else { xlog("L_NOTICE", "--- SCRIPT_DISPATCH_PROVIDER_FAILOVER: Failed to route request to PROVIDER! Giving Up. Negative response will be sent upstream.\n"); } } } One will rightfully wonder why use both branch-failure routes and failure_route to handle things. Well, it's not obvious from this because the relevant parts are removed for brevity, but I'm generally using branch failure event routes to try other destinations in the same destination set, and failure routes to additionally try other downstream "providers" in case all entries is the current destination set have failed. I don't think these actions/routes are relevant because the problem manifests without any of these failover mechanisms (of switching over to the next provider) engaging. Regarding the line "$du = "sip:" + $T_rpl($si) + ":" + $T_rpl($sp);" in the failure_route, which might seem a little odd, this was added because without it, uac_auth() will send the auth-carrying INVITE always to the first destination in the destination set even if ds_next_dst has been called from the branch-failure event route. Unfortunately I haven't been able to determine if this actually helped, because I have only received a 5xx error once from the downstream peer before the "fix" and haven't been able to reproduce this since then. But that's another issue unrelated to the one I'm asking about here. So what happens with track_cseq_updates enabled is, after uac_auth() successfully sends the authenticated INVITE and the peer starts sending provisional responses, kamailio doesn't seem to acknowledge them. Instead, it will retransmit the auth INVITE as if there was a firewall preventing the 1xx responses from being admitted. Then, the timeout will engage and the configuration script will behave as if t_branch_timeout returns true (it will do ds_next_dst). So I was wondering if this is to be expected with this configuration, or if this should be reported as a bug. Thanks! Best regards, George

2 2

Kamailio-5.3.3. NAT and RTP not working
by Pavithra Mohanraja 24 Jun '20

24 Jun '20

Hi , Could anybody please give me the approach how to configure NAT in kamailio IMS. I have installed kamailio IMS in openstack environment which has private ips for all four vms (each component in one vm). pcscf has one floating IP attached. I have enabled "WITH_NAT" and configured rtpproxy . when i make a call from outside of openstack using 2 zoiper clients with outbound proxy keeping floating ip of pcscf, I am getting 200 ok with all SIP packets correctly flowing but there is no RTP Packet. When i try the same within openstack environment without using public ip, I am getting RTP Packets. Kindly tell me the procedure of NAT Enablement or do i need to do anything other than this. Kindly help because i am stuck with this for long time. Thanks.

4 26

multiple domains with microsoft team
by Leanderson Correia 24 Jun '20

24 Jun '20

I'm testing here to use multiple domains with microsoft teams and I'm having trouble making and receiving calls. It seems that when connecting a call, kamailio keeps the tls connection active and reuses it, making the domain parameters used the wrong ones for new call. Can someone help me, or give a tip? -- Att, Lean

3 2

Presence and call queueing - Kamailio and Asterisk
by Pover A.V. 24 Jun '20

24 Jun '20

Summary: Using Kamailio to load balance between Asterisk servers. Trouble getting call queuing working because can't get device state hints to work reliably. - Hello I'd be grateful for any advice that can be offered for this problem to do with presence and call queueing. I have 3 Kamailio (v5.3.4) servers that load-balance incoming traffic from various sources: a legacy PBX, soft phone clients, and a SIP trunking service. Sitting behind the 3 Kamailio servers are 2 Asterisk (v16 LTS) servers. Kamailio acts as a registrar and user location service. Asterisk is routing calls and (should be) providing device state information for each extension. An external script builds a list of PJSIP endpoints based on the content of the Kamailio subscriber table, and deploys it to the Asterisk servers whenever a change is detected. The AOR for each record is configured to point to Kamailio. No authentication details are attached to these endpoints, as Kamailio is doing authentication before dispatching a request to Asterisk. Whenever Kamailio receives an INVITE, it passes this to an Asterisk server using the dispatcher module (round-robin algorithm '4'); asterisk then processes the request. At the moment, everything is working except for presence/device states and call queueing. I've tried to set up 'presence', 'pua', 'dialog' and '(presence/pua)_dialoginfo' modules on Kamailio with limited success. The presence state when a user toggles 'do not disturb' on their soft phone is passed to other users who are subscribed (watching), however when a user makes a call, no presence updates are seen by subscribed users (the extensions show as 'Idle' or 'Available' instead of 'In Use' or 'Busy'). Asterisk is able to report that an extension is 'Ringing' or 'InUse' when a call is made to that extension, because an INVITE to an internal extension matches a pattern in the dialplan and Asterisk routes it to a PJSIP endpoint with the same name. However, when somebody makes a call, Asterisk is unable to track the state of the extension making the call because the call is going via Kamailio so Asterisk is not seeing the caller as an internal extension. We need a way to make Asterisk think the call is coming from an internal device if the caller ID matches a PJSIP endpoint. Both Asterisk servers are configured to share device state updates with each other, and this is working. Without being able to see device state for an internal calling party, we will be unable to reliably set up call queues in Asterisk. Using the 'mohqueue' module on Kamailio is not something we want to do because of its reliance on a database, which is creating a single point of failure - if the database server is down for any reason, Kamailio refuses to start. Ideally, we would achieve some form of distributed call queueing, however the simplest option seems to be setting up a queue on one Asterisk server and having an active-passive high availability configuration, should a server go down. I'm looking for advice on how best to keep track of what is happening in a call, and for Asterisk to know the states of each extension and make these available in a dialplan hint. Or any other suggestions that might help to achieve call queueing in this setup. Thank you

2 1

Dialog Profiles
by Ilie Soltanici 24 Jun '20

24 Jun '20

Hello, Is there any way in Kamailio to get the profile values the current dialog is part of? For example, I have one dialog with the following profiles: profiles: { { test: abcd } test: 1234 } I need in the reply route to get the values for the test dialog profile. (abcd, 1234). I know I can use get_profile_size, but that will give the number of profiles of the current dialog, but I need the values to loop through. Thanks.

2 2

Kamailio I-CSCF not sending SIP:200 OK messages to Asterisk (to tag)
by Martin W Woscek 24 Jun '20

24 Jun '20

Hello all, SIP/UE (boghe or imsdroid) client registered to Kamailio makes call to an Asterisk registered SIP phone is successful: [UE]-Kamailio--->INVITE--->Asterisk16-[phone] [UE]-Kamailio--->200 OK with to_tag--->Asterisk16-[phone] But in reverse direction for a call, Kamailio does not return the SIP OK so no to_tag is sent so call fails to ring and complete: [phone]-Asterisk16 --->INVITE--->Kamailio Where the UE device never receives the INVITE, Asterisk never gets and 200 OK message with the to_tag from the I-CSCF, the call flow itself gets lost in Kamailio, where the P-CSCF sends final INVITE to I-CSCF and and ultimately a 604 HSS user unknown message is sent back to Asterisk from the I-CSCF. Basically Im using the default "sample" configs for both the P and the I-CSCF. Our sauce is in the S-CSCF for out going calls that originate by a registered UE. Any insight or sample Kamailio configuration that Im lacking? Has anyone done this and could share the asterisk and Kamailio script snippets that make it possible. Thanks, _Martin

2 1

Kamailio ims video calls
by Pavithra Mohanraja 24 Jun '20

24 Jun '20

Hi, Could anyone please give me some idea on how video call works in IMS. Does it need rtp proxy to work? Or is it the same way how voice call works ? It would be good , if anybody share some links on video calls as well voice conferencing. Thanks, pavithra

2 1

kamailio 5.2.2 enabling hashing_type
by Martin W Woscek 24 Jun '20

24 Jun '20

Hello, I'm revisiting some initial issues I had with P-CSCF. I could not set the hashing type in my PCSCF (kamalio 5.2.2) so I simply commented it out and moved on. But now, I may need it as the P-CSCF is not handling INVITES coming from I-CSCF that originated from external SIP server, where the endpoint (UE) is registered to the IMS but the P-CSCF sends the INVITE back to the S rather that the registered UE and fails. kamailio.cfg: modparam("ims_usrloc_pcscf", "hashing_type", 1) So with that, when I set the hashing_type to 1,1 or 2 I get the following starting the pcscf: Starting the P-CSCF: Jun 17 08:25:30 ubuntu kamailio_pcscf[5753]: * Not starting Kamailio IMS P-CSCF: invalid configuration file! Jun 17 08:25:30 ubuntu kamailio_pcscf[5753]: * Jun 17 08:25:30 ubuntu kamailio_pcscf[5753]: * 0(5766) ERROR: <core> [core/modparam.c:140]: set_mod_param_regex(): parameter <hashing_type> of type <2> not found in module <ims_usrloc_pcscf> Jun 17 08:25:30 ubuntu kamailio_pcscf[5753]: 0(5766) CRITICAL: <core> [core/cfg.y:3510]: yyerror_at(): parse error in config file /etc/kamailio/pcscf/kamailio.cfg, line 364, column 47: Can't set module parameter Am I missing some module that I did not include at compile time? According to the docs, it has no additional dependencies other than the ims_usrloc_pcscf.so https://kamailio.org/docs/modules/5.0.x/modules/ims_usrloc_pcscf.html#idp35… Any help is appreciated. Thanks, _Martin

2 1

kamailio5.2.2 calling t_relay(host, port)
by Martin W Woscek 24 Jun '20

24 Jun '20

Hello all, Im trying to call t_relay() with host and port as documented in: https://www.kamailio.org/docs/modules/5.2.x/modules/tm.html However, I get a subst_str(): no match on P-CSCF start and therefore fails. Is t_relay(host, port) supported as documented? Am I missing something? Thanks, _Martin

2 1

Expired transaction
by Marat Gareev 24 Jun '20

24 Jun '20

Hello! I'm trying to manually reproduce expired transactions which couldn't be cleared. I had one idea, but it was unsuccessful. I tried to send SIGSTOP to a group of Kamailio processes while the transaction is live. The idea is to freeze processes for a while, then continue to work, so that by this time the transaction deletion time was missed. But after a few seconds, the transaction is still cleared. Do you have any ideas for reproduction? P.S. I'm sorry If you received my incoherent mails. These were attempts to reply to the thread, seems to be my mail client or sr-users mailing list doesn't support In-Reply-To header.

2 1

ERROR: pua [send_publish.c:304]: publ_cback_func(): No Expires header found
by Denys Pozniak 23 Jun '20

23 Jun '20

Hello! Could someone point me to the possible source of the problem? Jun 16 09:51:06 proxy-edge /usr/sbin/kamailio[9645]: ERROR: pua [send_publish.c:304]: publ_cback_func(): No Expires header found Jun 16 09:51:07 proxy-edge /usr/sbin/kamailio[9640]: ERROR: pua [send_publish.c:304]: publ_cback_func(): No Expires header found Jun 16 09:51:07 proxy-edge /usr/sbin/kamailio[9600]: ERROR: pua [send_publish.c:304]: publ_cback_func(): No Expires header found Jun 16 09:51:07 proxy-edge /usr/sbin/kamailio[9660]: ERROR: pua [send_publish.c:304]: publ_cback_func(): No Expires header found Modules settings: *loadmodule "pua.so"* *loadmodule "pua_dialoginfo"* *loadmodule "dialog.so" * *modparam("pua", "default_expires", 1200)* *modparam("pua", "outbound_proxy", LOCAL_IP_ADDR)* *modparam("pua_dialoginfo", "include_callid", 1)* *modparam("pua_dialoginfo", "include_localremote", 1)* *modparam("pua_dialoginfo", "include_tags", 1)* *modparam("pua_dialoginfo", "send_publish_flag", 8)* *modparam("dialog", "dlg_flag", 4)modparam("dialog", "db_mode", 0)modparam("dialog", "enable_stats", 1)modparam("dialog", "dlg_match_mode", 1)modparam("dialog", "default_timeout", 7200) modparam("dialog", "hash_size", 8192)* *request_route{* *...* *setflag(8);* *...* *}* Generated SIP PUBLISH: *PUBLISH sip:extension@sip.company.com <sip%3Aextension(a)sip.company.com> SIP/2.0* *Via: SIP/2.0/UDP 8.8.8.8;branch=z9hG4bK3e55.afdce977000000000000000000000000.0* *To: <sip:extension@sip.company.com <sip%3Aextension(a)sip.company.com>>* *From: <sip:extension@sip.company.com <sip%3Aextension(a)sip.company.com>>;tag=e98d9fc94bada9746d67946ab287a76f-ba77* *CSeq: 10 PUBLISH* *Call-ID: 72ef85f374c8e703-9659(a)10.10.10.10 <72ef85f374c8e703-9659(a)10.10.10.10>* *Content-Length: 578* *User-Agent: proxy* *Max-Forwards: 70* *Event: dialog* *Expires: 7201* *Content-Type: application/dialog-info+xml* *<?xml version="1.0"?>* *<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0" state="full" entity="sip:extension@sip.company.com <sip%3Aextension(a)sip.company.com>">* * <dialog id="162092997-5060-464(a)BJC.BGI.CA.BBI" call-id="162092997-5060-464(a)BJC.BGI.CA.BBI" direction="initiator">* * <state>Trying</state>* * <remote>* * <identity>sip:1234567890@sip.company.com <sip%3A1234567890(a)sip.company.com></identity>* * <target uri="sip:1234567890@sip.company.com <sip%3A1234567890(a)sip.company.com>"/>* * </remote>* * <local>* * <identity>sip:extension@sip.company.com <sip%3Aextension(a)sip.company.com></identity>* * <target uri="sip:extension@sip.company.com <sip%3Aextension(a)sip.company.com>"/>* * </local>* * </dialog>* *</dialog-info>* -- BR, Denys Pozniak

2 5

icscf start error
by Pavithra Mohanraja 22 Jun '20

22 Jun '20

Hi, I have installed kamailio - 5.3.4 version where i am facing below issues in icscf while starting kamailio. db_mysql is loaded but .. what could be the issue? icscf : Aliases: *: ims.sip.example.com:* 0(789) INFO: <core> [core/tcp_main.c:5042]: init_tcp(): using epoll_lt as the io watch method (auto detected) 0(791) INFO: pike [pike.c:97]: pike_init(): PIKE - initializing 0(791) INFO: pike [ip_tree.c:81]: init_lock_set(): probing 256 set size *0(791) ERROR: <core> [db.c:204]: db_bind_mod(): Module db_con1=>mysql not found. Missing loadmodule? 0(791) ERROR: ims_icscf [db.c:83]: ims_icscf_db_bind(): ims_icscf_db_bind: cannot bind to database module! Did you forget to load a database module ? 0(791) ERROR: ims_icscf [ims_icscf_mod.c:270]: mod_init(): Failed to initialise ims_icscf module 0(791) ERROR: <core> [core/sr_module.c:849]: init_mod(): Error while initializing module ims_icscf (/usr/lib/x86_64-linux-gnu/kamailio/modules/ims_icscf.so)* ERROR: error while initializing modules 0(791) INFO: cdp [cdp_mod.c:255]: cdp_exit(): CDiameterPeer child stopping ... 0(791) INFO: cdp [diameter_peer.c:367]: diameter_peer_destroy(): destroy_diameter_peer(): Terminating all children... 0(791) INFO: cdp [diameter_peer.c:386]: diameter_peer_destroy(): destroy_diameter_peer(): All processes terminated. Cleaning up. 0(791) INFO: cdp [worker.c:139]: worker_destroy(): Unlocking workers waiting on empty queue... 0(791) INFO: cdp [worker.c:142]: worker_destroy(): Unlocking workers waiting on full queue... 0(791) CRITICAL: cdp [diameter_peer.c:426]: diameter_peer_destroy(): destroy_diameter_peer(): Bye Bye from C Diameter Peer test 0(791) INFO: cdp [cdp_mod.c:257]: cdp_exit(): ... CDiameterPeer child stopped 0(791) INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized

2 3

Kamailio v5.3.5 Released
by Daniel-Constantin Mierla 22 Jun '20

22 Jun '20

Hello, Kamailio SIP Server v5.3.5 stable release is out. This is a maintenance release of the latest stable branch, 5.3, that includes fixes since the release of v5.3.4. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.3.x. Deployments running previous v5.3.x versions are strongly recommended to be upgraded to v5.3.5. For more details about version 5.3.5 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2020/06/kamailio-v5-3-5-released/ RPM, Debian/Ubuntu packages will be available soon as well. 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

1 0

reason=404 (Not Found - destination user not found on this S-CSCF)
by Pavithra Mohanraja 22 Jun '20

22 Jun '20

Hi, I am using kamailio ims-5.3.4 where i am facing this kind of problem during route(term). call registration is happening successfully. In call Establishment , during route(term) , I am facing destination user not found . what could be the problem?. can anyone please help me. Thanks, Pavithra

2 2

Releasing Kamailio v5.3.5
by Daniel-Constantin Mierla 22 Jun '20

22 Jun '20

Hello, I am considering to release next week (likely on Monday, June 22) the next minor version out of branch 5.3, respectively 5.3.5. Should anyone be aware of issues not reported to the bug tracker or commits for fixes in master not backported to 5.3 branch, let us know. For new issues, the best is to report them on the tracker. For commits, you can provide links/hashids or make pull requests. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 1

Re: [SR-Users] 403 domain not served error in scscf
by David Villasmil 20 Jun '20

20 Jun '20

you shouldn't write to me directly. You can write to the list and address me. Anyway, kamailio thinks the message received is NOT directed to it. You should add an "alias" with the IP you're sending to (or the FQDN) Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 On Sat, Jun 20, 2020 at 9:01 PM Pavithra Mohanraja <pavimohan3004(a)gmail.com> wrote: > Hi David, > Could you please help me in this issue > > ---------- Forwarded message --------- > From: Pavithra Mohanraja <pavimohan3004(a)gmail.com> > Date: Fri, Jun 19, 2020, 1:36 PM > Subject: 403 domain not served error in scscf > To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> > > > Hi, > I am using kamailio IMS - 5.3.4 > when I am making a call between two clients using zoiper , Call > establishment is done successfully . > > But during BYE packet sent from caller to callee , it is reaching till > S-CSCF and giving 403 Domain not served .. Kindly help me in this regard. > > > I have sent lot of mails regarding this error and no replies .. Kindly > help .. > > Error in scscf : > Jun 19 13:29:17 ubuntu-VirtualBox kamailio[21891]: DEBUG: <core> > [core/mem/q_malloc.c:526]: qm_free(): freeing frag. 0x7fa1e6945ca8 alloc'ed > from core: core/rvalue.c: rval_new_empty(224) > Jun 19 13:29:17 ubuntu-VirtualBox kamailio[21891]: DEBUG: <core> > [core/socket_info.c:628]: grep_sock_info(): checking if host==us: 10==11 && > [*10.45.5.22] == [10.45.4.181]* > Jun 19 13:29:17 ubuntu-VirtualBox kamailio[21891]: DEBUG: <core> > [core/socket_info.c:635]: grep_sock_info(): *checking if port 4080 > (advertise 0) matches port 40629* > Jun 19 13:29:17 ubuntu-VirtualBox kamailio[21891]: DEBUG: <core> > [core/forward.c:412]: *check_self(): host != me* > Jun 19 13:29:17 ubuntu-VirtualBox kamailio[21891]: ERROR: <script>: > sip:alice at sip.example.com > Jun 19 13:29:17 ubuntu-VirtualBox kamailio[21891]: ERROR: <script>:* > Domain not served 403* > Jun 19 13:29:17 ubuntu-VirtualBox kamailio[21891]: ERROR: <script>: BYE ( > sip:bob@sip.example.com (10.45.4.177:4060) to sip:alice@sip.example.com;transport=TCP, > y-B071ZDjFS-z4NLxREyVw..) > > > Kindly help .. > 10.45.5.22 - sip client alice(a)sip.example.com > 10.45.4.181 - scscf > > > please help.. >

2 1

How do I test the Kamailio service ？
by zhaosl＠shinetechchina.com 20 Jun '20

20 Jun '20

Hi : Now I set up a Kamailio Service on the server, it works now. But I didn`t know how to test the server by trying the voice tool.Can you give me some advices , tools or methods ? Thanks. zhaosl(a)shinetechchina.com

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Re: [SR-Users] R: R: R: R: R: acc time mode 2 wrong timestamp
by mohsen khashei 20 Jun '20

20 Jun '20

Hi can you share your kamaili config. I want to add some extra columns to cdr table like what you did. Thanks. Sent from Mail<https://go.microsoft.com/fwlink/?LinkId=550986> for Windows 10

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Re: [SR-Users] R: R: R: R: R: acc time mode 2 wrong timestamp
by mohsen khashei 20 Jun '20

20 Jun '20

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403 domain not served error in scscf
by Pavithra Mohanraja 19 Jun '20

19 Jun '20

Hi, I am using kamailio IMS - 5.3.4 when I am making a call between two clients using zoiper , Call establishment is done successfully . But during BYE packet sent from caller to callee , it is reaching till S-CSCF and giving 403 Domain not served .. Kindly help me in this regard. I have sent lot of mails regarding this error and no replies .. Kindly help .. Error in scscf : Jun 19 13:29:17 ubuntu-VirtualBox kamailio[21891]: DEBUG: <core> [core/mem/q_malloc.c:526]: qm_free(): freeing frag. 0x7fa1e6945ca8 alloc'ed from core: core/rvalue.c: rval_new_empty(224) Jun 19 13:29:17 ubuntu-VirtualBox kamailio[21891]: DEBUG: <core> [core/socket_info.c:628]: grep_sock_info(): checking if host==us: 10==11 && [*10.45.5.22] == [10.45.4.181]* Jun 19 13:29:17 ubuntu-VirtualBox kamailio[21891]: DEBUG: <core> [core/socket_info.c:635]: grep_sock_info(): *checking if port 4080 (advertise 0) matches port 40629* Jun 19 13:29:17 ubuntu-VirtualBox kamailio[21891]: DEBUG: <core> [core/forward.c:412]: *check_self(): host != me* Jun 19 13:29:17 ubuntu-VirtualBox kamailio[21891]: ERROR: <script>: sip:alice at sip.example.com Jun 19 13:29:17 ubuntu-VirtualBox kamailio[21891]: ERROR: <script>:* Domain not served 403* Jun 19 13:29:17 ubuntu-VirtualBox kamailio[21891]: ERROR: <script>: BYE ( sip:bob@sip.example.com (10.45.4.177:4060) to sip:alice@sip.example.com;transport=TCP, y-B071ZDjFS-z4NLxREyVw..) Kindly help .. 10.45.5.22 - sip client alice(a)sip.example.com 10.45.4.181 - scscf please help..

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Development for v5.4.x is frozen
by Daniel-Constantin Mierla 19 Jun '20

19 Jun '20

Hello, the formal notification that the development for the next major version 5.4.0 is now frozen. The focus has to be on testing the master branch. Also, the master branch should not get commits with new features till the branch 5.4 is created, expected to happen in 2-4 weeks, a matter of how testing goes on. Meanwhile, the commits with new features in the C code can be pushed to personal branches, pull requests can still be done, but they will be merged after branching 5.4. Can still be done commits with documentation improvements, enhancements to related tool (e.g., kamctl, kamcmd), exporting missing KEMI functions and completing the functionality of the new modules added in 5.4. Once the branch 5.4 is created, new features can be pushed again to master branch as usual. From that moment, the v5.3.0 should be out very soon, time used for further testing but also preparing the release of packages. If someone is not sure if a commit brings a new feature, just make a pull request and it can be discussed there on github portal or via sr-dev mailing list. A summary of what is new in upcoming 5.4 is going to be built at: * https://www.kamailio.org/wiki/features/new-in-5.4.x Upgrade guidelines will be collected at: * https://www.kamailio.org/wiki/install/upgrade/5.3.x-to-5.4.0 Everyone is more than welcome to contribute to the above wiki pages, especially to the upgrade guidelines, to help everyone else during the migration process from v5.3.x to 5.4.x. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

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MD5 and SHA-256 instead of MD5 or SHA-256...
by Aymeric Moizard 18 Jun '20

18 Jun '20

Hi All, I'd like to improve my setup by switching to SHA-256. However, as a first step, I would like to offer both MD5 and SHA-256 in 2 different WWW-Authenticate header. If I'm correct, this is not doable with the latest auth module? Is this a planned feature? As an alternative, I would like to decide the algorithm in the script instead of a module parameter. It looks to me this is also not doable? Again, is this a planned feature? Thanks to all, Regards Aymeric -- Antisip - http://www.antisip.com

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Freezing development for Kamailio v5.4.0
by Daniel-Constantin Mierla 18 Jun '20

18 Jun '20

Hello, based on the discussions during the last online devel meeting, the plan is to release the next major version (Kamailio v5.4.0) around the mid of this summer. Therefore I propose to freeze the development on Thursday, June 18, 2020. Then we will have about 4-6 weeks of testing before the release is out. Master will be unfrozen when the branch 5.4 is created (usually that was done after 3-4 weeks in the testing phase). If you are planning to introduce new features in the 5.4 release series, be sure you push them to master branch or create a pull request till the freezing date. If anyone else wants a different time line for 5.4.0 release, let us know and we can discuss to match the best for everyone. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

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