- sr-users - kamailio.org

by Patrick Karton

hello, as we are all aware kamailio create a lot of processes for many tasks. im using kamailio in kubernetes and i would like to monitor only sip processes with top command. i only need a way to get only kamailio sip processes id (udp receiver that handle sip request) . but i dont knwo how to get it. top -c -p id1,id2,id3,id4 Thanks

2 years, 7 months

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SEMS license with kamailio and rtpengine

by Patrick Karton

Hello im using kamailio and rtpengine to build a custom SBC. i was using freeswitch as my b2bua but i noticed a lot of issues when cps and simultaneous increased so i tried sems and i have better results. But i do not really understand sems license so i want to know if i can use it. i read from sems github "It is licensed under dual license terms, the GPL (v2+) and proprietary license. This program is released under the GPL with the additional exemption that compiling, linking, and/or using OpenSSL is allowed. For a license to use SEMS under non-GPL terms, please contact FRAFOS GmbH at info(a)frafos.com" so my question is if i want to use sems as b2bua with kamailio and rtpengine and make profit of it do i need to purchase a license each time i want to sell it to a customer ? Thanks.

2 years, 7 months

7
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Kamailio server behind NAT

by Tihomir Culjaga

Hi all I'm wondering what is best practice to configure kamailio for a setup shown below. Carriers only support interconnection over a private network so I need to terminate every separate network on kamailio machine (and on RTPEngine) . That will turn very quickly into an administration nightmare as we add more and more carriers. My plan is to terminate carrier private networks on a router and NAT the traffic to kamailio where it leads me to a kamailio behind NAT setup. Well, in theory set_advertised_address and set_advertised_port per message can do the trick... at least I hope. Also, some carriers require a SIP registration to activate SIP trunks. I do that using UAC module (remote registration) but here again I got a problem with contact address as you can set only one contact address (modparam("uac", "reg_contact_addr", "10.1.1.11:5060")). /--RTP 10000-19999-- RTPEngine1 -----RTP------+-- IPPBX1 carrier1 --- NAT ---| | | | | | carrier2 --- NAT ---+-----SIP 5060------ kamailio ----SIP---- Dispatcher | | | carrier3 --- NAT ---| | | \--RTP 20000-29999-- RTPEngine2 -----RTP------+-- IPPBX2 So bottom line, I'm wondering what would be the best practice for such a scenario ? That would spare me from hitting many pitfalls. Would really appreciate a hint. Regards, Tihomir.

2 years, 7 months

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Vo5G with open5gs

by Hornbuckle, Mark A

Hello, I have been trying to find documentation on Kamailio IMS and Vo5G support for open5gs. I wanted to ask you directly if Kamailio IMS does support Vo5G with open5gs? Thank you

2 years, 7 months

1
0
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resending same message from UE after receiving RP-ACK

by Eugene B

Hi, I trying to use smsops module as SMSC. When I sending message from UE I see re-sending same message after each RP-ACK(~10 attempt). What could be the problem? Thanks! pcap in attach. kamailio 5.3.2 -- *Yours faithfully,* *Eugene B.*

2 years, 7 months

2
1
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kamilio open5gs 5G SA compatible?

by Sandland, Seth P

Were looking to test a kamilio on Open5gs 5g stand alone ( non lte, non NSA) can you conform or provide documents or install instructions on how this is accomplished. I can not find much of any details if Kamilio support Open5Gs 5g SA and what features it does support on a 5g SA core. Seth Sandland Wireless Communications IT Engineer Pacific Northwest National Laboratory Tel: (509) 375-6842 Email: seth.sandland(a)pnnl.gov<mailto:seth.sandland@pnnl.gov> www.pnnl.gov<http://www.pnnl.gov/> [A close up of a logo Description automatically generated]

2 years, 7 months

1
0
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Publish XML and xpath

by Igor Olhovskiy

Hello, I'm trying to get state of endpoint from PUBLISH message in a most simple way using xmlops module Trying to make this $xml(body=>doc) = $rb; log("L_ERR", "$rU is now in state $xml(body=>xpath:/dialog-info/dialog/state)\n"); XML in PUBLISH itself <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0" state="full" entity="sip:88881@dev.company"> <dialog id="v6qmEYf5yI" direction="initiator"> <state>confirmed</state> </dialog> </dialog-info> Actual text in log is ... <script>: 88881 is now in state Means it's empty. xmlops module is loaded with no parameters. Tried also string $xml(body=>xpath:/dialog-info/dialog/state/text()) -> same result What am I missing here? kamailio 5.4.6 -- Thanks in advance, Igor

2 years, 7 months

2
5
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KEMI KSR.async Python3

by Ahmed Moubayed

Hi, I'm trying to use the async module from a python3 Kemi script but it conflicts with the keyword async. Any suggestions on how it can be used? Is there an alternative to KSR.async.route()? Thanks, Ahmed

2 years, 7 months

1
0
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CDR preserve original call direction

by Ahmed Fouad

Hi, I've enabled ACC and CDR logging and it's working fine for the logging but I noticed something wrong in the CDR table , the call source and destinations are depending on the BYE message not the initial invite dialog ACC table +-----+--------+--------------------------------------+---------------------+---------------+---------------------------+--------------+--------------+--------------------------------------+ | id | method | callid | time | src_ip | src_domain | src_user | dst_ouser | dst_domain | +-----+--------+--------------------------------------+---------------------+---------------+---------------------------+--------------+--------------+--------------------------------------+ | 216 | BYE | 81fa79a0-3deb-47b6-80ab-831a78a5d46e | 2022-02-09 06:57:51 | x.x.x.x | domain.com | DST_CLID | SRC_CLID | domain.com | | 215 | INVITE | 81fa79a0-3deb-47b6-80ab-831a78a5d46e | 2022-02-09 06:55:17 | x.x.x.x | domain.com | SRC_CLID | DST_CLID | dst_domain.com | The INVITE is the correct one with SRCIP x.x.x.x and domain.com but the BYE reversed the order since the the DST_CLID who sent the BYE so it's now logged as the source and the final CDR log looks like +--------+--------------+---------------------------+--------------+--------------------------------------+---------------------+---------------------+----------+-----+-------+--------+-------+---------+ | cdr_id | src_username | src_domain | dst_username | dst_domain | call_start_time | call_end_time +--------+--------------+---------------------------+--------------+--------------------------------------+---------------------+---------------------+----------+-----+-------+--------+-------+---------+ | 195 | DST_CLID | domain.com | SRC_CLID | domain.com | 2022-02-09 06:55:12 | 2022-02-09 06:57:51 | the CDR as shown is reversed regarding the source/dst username I tried to enable the direction ACC direction flag but didn't help and here is my current paramters modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) modparam("acc", "detect_direction", 0) modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "cdr_enable", 1) modparam("acc", "cdrs_table", "cdrs") Can you please advise how to configure ACC/CDR to log the CDR with the correct initial INVITE call flow regardless of the BYE direction ?

2 years, 7 months

3
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REGISTER with http_async_client ?

by Cyril Ramière

Hi there, I have a weird issue with kamailio (latest docker image kamailio-ci:5.5.2-alpine) and http_async_client. Before posting a lot of logs, let me describe what I want to achieve. I have a Kamailio and a SIP Phone. The SIP phone sends a REGISTER to kamailio, then in my routing block, I check if I have an Authorization header. Since I don't have an Authorization (first message), I use "www_challenge()". This replies to the SIP phone, and then the SIP phone sends a new REGISTER with the correct Authorization header. So far so good. Now, when I get the REGISTER with Authorization header, I want to ask an HTTP endpoint if this user is allowed to connect and check the password using http_async_query(). The problem is that when the transaction resumes, the tmx module is unhappy and throws this error : 30(36) CRITICAL: tmx [t_var.c:546]: pv_get_tm_reply_code(): no picked branch (-1) for a final response in MODE_ONFAILURE And a 500 error is sent back to the sip phone. The AUTH_REPLY route is still called and I can use the $http* values. Do you see something that I am doing wrong or missing in my logic? Is pausing/resuming to use the async http client is allowed if I'm handling a REGISTER transaction? Here's a simplified version of my routing block (not far from reality): ##### SNIP request_route{ route(AUTH); route[AUTH]{ if (is_method("REGISTER"){ if(no_auth_header){ www_challenge("$td","1"); exit; } else{ t_newtran(); http_async_query("http://xxx.xxx.xxx.xxx:9000/auth?foo=bar", "AUTH_REPLY"); } } } route[AUTH_REPLY]{ xlog("L_INFO", "route[HTTP_REPLY]: status $http_rs\n"); } } ##### END SNIP Best regards!

2 years, 7 months

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Kamailio doc request

by Mustafa Rahman

2 years, 7 months

2
1
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Failure route with t_relay()

by Max Kemel

Hey, I am trying to make a re-transmission mechanism from the failure route, I've tried to use $dp and $du as a relay, but variables are not accepted in t_relay(), When using ip and port straight forward t_relay("144.0.0.0", "1111"), I am getting: [t_fwd.c:1752]: t_forward_nonack(): no branches for forwarding [tm.c:1753]: _w_t_relay_to(): t_forward_noack failed Is there a way to re-transmit the call to the same carrier again? Thanks in advance, and have a great day, Max Kemel

2 years, 8 months

2
1
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Failure route mechanism with counter

by Max Kemel

Hey, I would like to use a failure route with a retry mechanism, which provides a MAX_NUM_OF_RETRIES (max number of times reaching the failure route), Which can be achieved by a simple global counter, from what I've seen, the Counters module doesn't provide a GET function in script, What is the easiest solution for this kind of behavior? Thanks a lot, Max Kemel

2 years, 8 months

2
1
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Timers firing too soon

by Alberto Izquierdo

Hi all, I'm using Kamailio timers to execute a series of calls with a certain delay. I have found out that from time to time, the timers are triggered too soon, sometimes they are triggered 2/3 seconds before they should. I have read in the documentation that it is normal to have a bit of delay in some of the timers, but it doesn't mention anything about this behavior. Is this normal? Thank you.

2 years, 8 months

1
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Debian kamailio.init vs. kamailio.service

by Juha Heinanen

By comparing pkg/kamailio/deb/bullseye dir kamailio.init and kamailio.service files, it appears to me that they are not equal alternatives. Is this correct interpretation? -- Juha

2 years, 8 months

1
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Homer7 and Kamailio 5.3

by Angelo Sipper

Hi to all, We have a Kamailio 5.3 that sends all traffic to a homer7 using siptrace module with parameters force_send_sock and hep_mode_on,1 and hep_version,3 and trace_mode,1. Everything works fine getting all traffic to Homer7 but, in case Homer7 is not reachable, Kamailio then no longer accepts traffic or registrations and it is like it crashes. When we revive Homer7 and is it again reachable accepting traffic, Kamailio then works again fine without any restart or actions. Any ideas why this may be happening? Is there anything we need to add to the config to avoid this behavior as having Homer7 now is a single point of failure for Kamailio? Thank you! Kind Regards, Angelo

2 years, 8 months

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uac problem with 5.3.8

by Richard Robson

we have been running 5.3.4 with the uac_auth_mode() function ans it has been working we have upgraded to 5.3.8 and with the same script we are getting an error with the uac module saying it was unable to find the cmd auc_auth_mode I have tried this with uac_auth as well with the same result. it looks like the uac module is not loaded or is not loading but running a core.modules reports the uac module can anyone shed any light on this problem? is it a version pecific problem?

2 years, 8 months

3
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Updating $du for uac REGISTERs in event_route[tm:local-request]

by Raffaella Seuß

Hello all, I tried to update $du for REGISTERs sent by the uac module in event_route[tm:local-request]. (I want to overwrite the auth_proxy under some conditions.) However, the requests are still sent to the auth_proxy defined in uacreg. Am I missing something here, why doesn't it work? It's basically just: event_route[tm:local-request] { if (is_method("REGISTER")) { # ... $du = "sip:PROXY_IP:PROXY_PORT"; } } My Kamailio version is 5.5.2. With kind regards, Raffaella Seuß foncloud GmbH & Co KG Hahlweg 2a 36093 Künzell Tel: / Fax: +49 661 968990-99 Email: raffaella.seuss(a)foncloud.net Web: www.foncloud.net P.S.: Wussten Sie schon? Unter https://www.foncloud.net/wissen finden Sie zahlreiche Informationen und hilfreiche Artikel rund um unsere Produkte und Services. Registergericht: Amtsgericht Fulda, Persönlich haftende Gesellschafterin der foncloud GmbH&Co.KG: Global Brain Network GmbHGeschäftsführer der Global Brain Network GmbH: Peter Krug Sitz der Gesellschaft: Künzell. Diese E-Mail enthält vertrauliche und/oder rechtlich geschützte Informationen. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail irrtümlich erhalten haben, informieren Sie bitte sofort den Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser Mail ist nicht gestattet. This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorised copying, disclosure or distribution of the material in this e-mail is strictly forbidden.

2 years, 8 months

1
0
0 0

Siremis configuration

by Tim Mastarone

Hi, I am using Siremis v 5.3 with kamailio. My users can log into the Siremis site just fine but the field to enter their email address is not appearing. My sql queries seem to be working besides that. Does this feature need to be enabled? I would like to use email to confirm the user login with a confirmation code. (mysql Ver 15.1 Distrib 10.1.48-MariaDB, for debian-linux-gnu (x86_64) using readline 5.2) Thank you, [cid:image001.png@01D81830.52C21EA0]<http://www.vikingelectronics.com/> TIM MASTARONE Associate Engineer 715-386-8861 x208 vikingelectronics.com<http://www.vikingelectronics.com/>

2 years, 8 months

1
0
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two SIP proxies

by Uwe Sigleur

2 years, 8 months

2
1
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tel2sip is giving unpredictable result

by Jyoti Bansal

hello, we are using kamailio 5.3 . while establishing a call the tel uri is getting converted into sip uri sometimes incorrectly . In the sip uri some extra characters are getting appended in the invite message. Can somebody please provide some guidance on this, attaching the pcap for reference(packet no. 8). Thanks, *Jyoti Bansal* Software Engineer . Great Software Laboratory | www.gslab.com -- Confidentiality Notice and Disclaimer: This email (including any attachments) contains information that may be confidential, privileged and/or copyrighted. If you are not the intended recipient, please notify the sender immediately and destroy this email. Any unauthorized use of the contents of this email in any manner whatsoever, is strictly prohibited. If improper activity is suspected, all available information may be used by the sender for possible disciplinary action, prosecution, civil claim or any remedy or lawful purpose. Email transmission cannot be guaranteed to be secure or error-free, as information could be intercepted, lost, arrive late, or contain viruses. The sender is not liable whatsoever for damage resulting from the opening of this message and/or the use of the information contained in this message and/or attachments. Expressions in this email cannot be treated as opined by the sender company management – they are solely expressed by the sender unless authorized.

2 years, 8 months

2
1
0 0

kamailio log dialog correlation

by marek

hi, is it possible create log of SIP dialog like https://wiki.asterisk.org/wiki/display/AST/Unique+Call-ID+Logging ? it looks like its possible like this xlog("L_INFO", "D$dlg(h_id) something to log \n"); but i must copy $dlg(h_id) in every xlog tried this modparam("xlog", "prefix", "D$dlg(h_id)") but it not works with variables any other tips/ideas? goal is to correlate logs for one dialog (call) thanks Marek

2 years, 8 months

7
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REGISTER with http_async_client?

by Cyril Ramière

Hi there, I have a weird issue with kamailio (latest docker image kamailio-ci:5.5.2-alpine) and http_async_client. Before posting a lot of logs, let me describe what I want to achieve. I have a Kamailio and a SIP Phone. The SIP phone sends a REGISTER to kamailio, then in my routing block, I check if I have an Authorization header. Since I don't have an Authorization (first message), I use "www_challenge()". This replies to the SIP phone, and then the SIP phone sends a new REGISTER with the correct Authorization header. So far so good. Now, when I get the REGISTER with Authorization header, I want to ask an HTTP endpoint if this user is allowed to connect and check the password using http_async_query(). The problem is that when the transaction resumes, the tmx module is unhappy and throws this error : 30(36) CRITICAL: tmx [t_var.c:546]: pv_get_tm_reply_code(): no picked branch (-1) for a final response in MODE_ONFAILURE And a 500 error is sent back to the sip phone. The AUTH_REPLY route is still called and I can use the $http* values. It seems that my AUTH_REPLY route is called but considered as a "failure_route" 19(27) exec: *** cfgtrace:dbg_cfg_trace(): failure_route=[AUTH_REPLY] c=[/etc/kamailio/kamailio.cfg] l=665 a=16 n=if 17(23) NOTICE: <script>: [AUTH_REPLY] FAILURE ROUTE If I remove the http_async_client and try with route(AUTH_REPLY) then it is considered as request_route (but this is not a solution since I need to do this http request) Do you see something that I am doing wrong or missing in my logic? Is pausing/resuming to use the async http client is allowed if I'm handling a REGISTER transaction? Here's a simplified version of my routing block (not far from reality): ##### SNIP request_route{ route(AUTH); route[AUTH]{ if (is_method("REGISTER"){ if(no_auth_header){ www_challenge("$td","1"); exit; } else{ t_newtran(); http_async_query("http://xxx.xxx.xxx.xxx:9000/auth?foo=bar", "AUTH_REPLY"); } } } route[AUTH_REPLY]{ xlog("L_INFO", "route[HTTP_REPLY]: status $http_rs\n"); if(t_is_failure_route()) { xlog("L_NOTICE", "[AUTH_REPLY] FAILURE ROUTE \n\n"); } } } ##### END SNIP Best regards!

2 years, 8 months

1
1
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Use specific expires per User-Agent

by mayamatakeshi

Hi, I have a requirement to specify a REGISTER max_expires based on the User-Agent. I checked the docs and it seems there is no pseudo-variable or something like that would permit this. I would like to be able to do something like this: if($ua ~= "Some-UA") { $max_expires=100; } else { $max_expires=150; } save("location"); Is there some way to achieve this?

2 years, 8 months

2
2
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failed to find string from message body

by Juha Heinanen

I tried to check if message body contains a string using function call str_find("$rb", "string")) but it failed like this: Jan 29 17:37:03 siika /usr/bin/sip-proxy[2577003]: ERROR: <core> [core/pvapi.c:1491]: pv_printf_mode(): no more space for spec value - printed:0 token:6880 buffer:4096 Jan 29 17:37:03 siika /usr/bin/sip-proxy[2577003]: ERROR: <core> [core/pvapi.c:1503]: pv_printf_mode(): buffer overflow -- increase the buffer size... Jan 29 17:37:03 siika /usr/bin/sip-proxy[2577003]: ERROR: <core> [core/sr_module.c:1698]: get_str_fparam(): Could not convert the PV-formated string to str Jan 29 17:37:03 siika /usr/bin/sip-proxy[2577003]: ERROR: textops [textops.c:4457]: str_find_f(): cannot get first parameter Any suggestions on how to get it done properly? -- Juha

2 years, 8 months

1
0
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Question about siptrace destination/source port capture in HEPv3 mode

by Karsten Horsmann

Hello List, I observe an interesting behavior. We use siptrace as an active capture agent for QXIP HOMER/HEPIC installations. The destination port and the source port in the HEPv3 Header are not correct if you use transport tcp for example. My Question: is that intended or a bug and i should fill up an issue? On udp, it's filled with 5060 or whatever your listening directive says. I tested that with kamailio 5.5 and kamailio master. I attached an HEPv3 capture from two kamailios (master) speaking with each other. You need the https://github.com/sipcapture/hep-wireshark lua dissector to read the capture. 10.0.2.15 (kamailio with uacreg and sipcapture/HEPv3 mode) 192.168.50.4 (kamailio as registrar) Frame 1 shows an tcp transport register Request from 10.0.2.15 to 192.168.50.4 in HEPv3. The Source Port in the HEP3 Protocol is not correct (its an highport / 45419). Frame 4 shows the 401 answer and the Destination port is 0. Frame 7 also had this behavior. ----- params (nearly at the end): modparam("siptrace", "duplicate_uri", "MY_HOMER_CAPTURE") modparam("siptrace", "hep_mode_on", 1) modparam("siptrace", "hep_version", 3) modparam("siptrace", "hep_capture_id", MY_HOMER_CAPTURE_ID) # capture agent id Limitation: 32-bit for HEPv3. modparam("siptrace", "trace_to_database", 0) modparam("siptrace", "trace_on", 1) modparam("siptrace", "trace_mode", 1) # default 0, if 1 then you dont need call siptrace flag or sip_trace() -- Mit freundlichen Grüßen / Kind Regards *Karsten Horsmann*

2 years, 8 months

3
2
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tel2sip is not working correctly

by Jyoti Bansal

Hello all, I am using kamailio code to establish calls with the real ue's. Our calls are getting established with an issue , whenever invite is send from A party to B party with teluri that tel uri is converted to sip uri. While converting to sip uri some extra bytes are getting added after user=phone and the call ends. Can someone help us on this issue of getting some extra bytes added in the sip uri? for example :- sip:405874112224411@ims.mnc123.mcc456.3gppnetwork.org;user=phone�� Thanks, *Jyoti Bansal* Software Engineer . Great Software Laboratory | www.gslab.com -- Confidentiality Notice and Disclaimer: This email (including any attachments) contains information that may be confidential, privileged and/or copyrighted. If you are not the intended recipient, please notify the sender immediately and destroy this email. Any unauthorized use of the contents of this email in any manner whatsoever, is strictly prohibited. If improper activity is suspected, all available information may be used by the sender for possible disciplinary action, prosecution, civil claim or any remedy or lawful purpose. Email transmission cannot be guaranteed to be secure or error-free, as information could be intercepted, lost, arrive late, or contain viruses. The sender is not liable whatsoever for damage resulting from the opening of this message and/or the use of the information contained in this message and/or attachments. Expressions in this email cannot be treated as opined by the sender company management – they are solely expressed by the sender unless authorized.

2 years, 8 months

1
0
0 0

Comunication between devices RTP/SRTP and viceversa

by Social Boh

I'm trying to configure Kamailio to permit communication between devices: some use TLS and SRTP some UDP and RTP with RTPEngine in the middle. A priori I don't know if the device support SRTP or RTP so in the routing I have to detect on the fly how to act. The problem is this: Call from RTP to SRTP device: 488 Not Acceptable here Call from SRTP to RTP device: 415 Unsupported Media Type Any hint? Thank you Regards -- --- I'm SoCIaL, MayBe

2 years, 8 months

4
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200 OK re transmission only on re-SUBSCRIBE

by Lewis Hutchinson

First of all I should start by saying this is my first post so go easy on me!! : ) and thank for everyone's efforts to share knowledge it has been invaluable for my kamailio learning. Build Details Version: kamailio 5.5.3 (x86_64/linux) Kamailio kemi python Scenario Relevant config def ksr_route_api_query(self, msg): KSR.log("info", "SUBSCRIBE API Query Starts") # Create new Transaction KSR.tm.t_newtran() async def main(): url = 'https://api.hidingtheurl.com/subscribe' payload = { "ruri": KSR.pv.get("$ru"), "from": KSR.pv.get("$fu"), "from_tag": KSR.pv.gete("$ft"), "to": KSR.pv.get("$tu"), "callid": KSR.pv.get("$ci"), "cseq": KSR.pv.get("$cs"), "contact": KSR.pv.gete("$ct"), "supported": KSR.pv.get("$hdr(Supported)"), "event": KSR.pv.get("$hdr(Event)"), "expires": KSR.pv.get("$hdr(Expires)"), "diversion": KSR.pv.get("$di"), "body": KSR.pv.get("$rb") } async with aiohttp.ClientSession() as session: async with session.post(url, json=payload) as response: if response.status == 0: KSR.xlog.xinfo(f"SUB Manager Timeout: {response.status}") KSR.tm.t_reply(408, "SUB Manager Timeout") elif response.status >= 500: KSR.xlog.xinfo(f"SUB Manager Down: {response.status}") KSR.tm.t_reply(500, "SUB Manager Down") elif response.status >= 400: KSR.xlog.xinfo(f"SUB Manager Error: {response.status}") KSR.tm.t_reply(400, "SUB Manager Error") else: if response.status == 200: KSR.xlog.xinfo(f"SUB Manager Success: {response.status}") KSR.tm.t_reply(200, "OK") KSR.log("info", "SUBSCRIBE API Query Ends") asyncio.run(main()) # API Query Ends Scenario Initial SUBSCRIBE >>> hits Kamailio >>> send payload in JSON to API get receive 200OK response from API and send a 200OK on to Device SUBSCRIBE --------> <------------ 200 OK After Subscription Expiry re-SUBSCRIBE >>> hits Kamailio >>> send payload in JSON to API get 200OK response from API and send a 200OK on to Device but also another 200OK a fraction of a second later... SUBSCRIBE --------> <------------ 200 OK <<<--------- 200 OK This happens for all following re-SUBSCRIBE's The 200 OK are identical, and it is a re transmission, but I can't work out why.. Relevant DEBUG is below Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: INFO: <script>: SUB Manager Success: 200 Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: DEBUG: app_python3 [apy_kemi.c:232]: sr_apy_kemi_exec_func_ex(): execution of method: t_reply Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: DEBUG: app_python3 [apy_kemi.c:287]: sr_apy_kemi_exec_func_ex(): params[2] for: t_reply are int-str: [200] [OK] Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: DEBUG: tm [t_lookup.c:1034]: t_check_msg(): msg (0x7eff0cd4e828) id=2/11234 global id=2/11234 T start=0x7eff0918d7d8 Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: DEBUG: tm [t_lookup.c:1108]: t_check_msg(): T (0x7eff0918d7d8) already found for msg (0x7eff0cd4e828)! Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: DEBUG: <core> [core/msg_translator.c:162]: check_via_address(): (80.111.111.111, 80.111.111.111, 0) Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: DEBUG: tm [t_reply.c:1763]: cleanup_uac_timers(): RETR/FR timers reset Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: DEBUG: tm [t_reply.c:637]: _reply_light(): reply sent out - buf=0x7eff0cd50938: SIP/2.0 200 OK#015#012Via:... shmem=0x7eff09190ca8: SIP/2.0 200 OK#015#012Via: Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: DEBUG: tm [t_reply.c:648]: _reply_light(): finished Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: DEBUG: app_python3 [apy_kemi.c:232]: sr_apy_kemi_exec_func_ex(): execution of method: t_precheck_trans Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: DEBUG: app_python3 [apy_kemi.c:232]: sr_apy_kemi_exec_func_ex(): execution of method: t_check_trans Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: DEBUG: tm [t_lookup.c:1034]: t_check_msg(): msg (0x7eff0cd4e828) id=2/11234 global id=2/11234 T start=0x7eff0918d7d8 Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: DEBUG: tm [t_lookup.c:1108]: t_check_msg(): T (0x7eff0918d7d8) already found for msg (0x7eff0cd4e828)! Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: DEBUG: tm [t_reply.c:1703]: t_retransmit_reply(): reply retransmitted. buf=0x7eff0cb18220: SIP/2.0 2..., shmem=0x7eff09190ca8: SIP/2.0 2 Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: DEBUG: app_python3 [apy_kemi.c:112]: sr_kemi_config_engine_python(): execution of route type 1 with no name returned 1 Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: DEBUG: <core> [core/receive.c:514]: receive_msg(): request-route executed in: 38258 usec Everything works well as is but I want to clean up this issue or at least understand what is causing it. Lewis

2 years, 8 months

2
6
0 0

Some problems with SIP TRACE

by Arnaud

Hi, I'm trying to use siptrace to getting copy of my SIP transactions, especially in TLS. Here's my configuration : *loadmodule "siptrace.so" modparam("siptrace", "trace_mode", 1) modparam("siptrace", "trace_to_database", 0) modparam("siptrace", "trace_on", 1) modparam("siptrace", "duplicate_uri", "sip:MY_IP:5065") modparam("siptrace", "trace_local_ip", "MY_IP:5061") modparam("siptrace", "hep_mode_on", 1) modparam("siptrace", "hep_version", 2) modparam("siptrace", "hep_capture_id", 1) modparam("siptrace", "trace_flag",22); * And in my Routing Logic : *request_route {* *sip_trace();setflag(22);* *(...)* But I didn't receive anything on my listening port 5065 according my TCPDUMP. Status confirm me siptrace is working : *root@debian:/# kamcmd siptrace.status checkEnabled* I think I made a mistake in the configuration, but I don't know where. My second question is when I try to use sip_trace_mode("t"); I getting : failed to find command sip_trace_mode. Is it in an other module ? Many thanks for your help. Regards, Arnaud

2 years, 8 months

2
1
0 0

Re: [SR-Users] sr-users Digest, Vol 200, Issue 25

by surendra tiwari

Hi Aditya, There are 2 options to achieve this: Option-1 use below param in auth_db module and while doing authentication change table name to your table user_column (string) <https://www.kamailio.org/docs/modules/devel/modules/auth_db.html#auth_db.p.…> domain_column (string) <https://www.kamailio.org/docs/modules/devel/modules/auth_db.html#auth_db.p.…> password_column (string) <https://www.kamailio.org/docs/modules/devel/modules/auth_db.html#auth_db.p.…> Option-2 create a new http service that will connect to kamailio via evapi and will be responsible for authentication and location service. On Wed, 26 Jan 2022 at 16:31, <sr-users-request(a)lists.kamailio.org> wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.kamailio.org > > To subscribe or unsubscribe via the World Wide Web, visit > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.kamailio.org > > You can reach the person managing the list at > sr-users-owner(a)lists.kamailio.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. Kamailio as a SIP registrar (Aditya Uppuluri) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 26 Jan 2022 16:21:33 +0530 > From: Aditya Uppuluri <uppuluriaditya(a)gmail.com> > To: sr-users(a)lists.kamailio.org > Subject: [SR-Users] Kamailio as a SIP registrar > Message-ID: > <CAKWpYb= > 9EABvpmbPhvCyt-roQaOkiJnZSh17vVu6Wk26PiBbeg(a)mail.gmail.com> > Content-Type: text/plain; charset="utf-8" > > Hello Team, > Thank you for the amazing product. I am a newbie to kamailio and currently > trying to use kamailio as a SIP Registrar (for a multidomain). Here is the > requirement > > 1. We already have a mysql database (name: VOIP) with a table named > VoipSubscribers which contains the userid, password and domain information > of all the users. > 2. We want the ability in kamailio to use the same VoipSubscribers table > for user registrations and also store the location information in the same > database (VOIP) > > May I know how to achieve this without altering the table schema? > Thank you in advance. > > Regards, > Aditya Uppuluri >

2 years, 8 months

1
0
0 0

Kamailio as a SIP registrar

by Aditya Uppuluri

Hello Team, Thank you for the amazing product. I am a newbie to kamailio and currently trying to use kamailio as a SIP Registrar (for a multidomain). Here is the requirement 1. We already have a mysql database (name: VOIP) with a table named VoipSubscribers which contains the userid, password and domain information of all the users. 2. We want the ability in kamailio to use the same VoipSubscribers table for user registrations and also store the location information in the same database (VOIP) May I know how to achieve this without altering the table schema? Thank you in advance. Regards, Aditya Uppuluri

2 years, 8 months

2
1
0 0

Append HF don't work in event_route

by Arnaud

Hello, I'm trying to connect kamailio to Microsoft Teams. Dispatch is correctly configured and sended. Following this tutorial : https://skalatan.de/en/blog/kamailio-sbc-teams, I added an Event_Route to add contact header to SIP OPTION from dispath to MS : event_route[tm:local-request] { xlog("*** tm:local-request *** "); if(is_method("OPTIONS") && $ru =~ "pstnhub.microsoft.com") { xlog("L_INFO", "Append new contact !\n"); append_hf("Contact: <sip:MY_SBC_NAME.COM:5061 ;transport=tls>\r\n"); } xlog("L_INFO", "Sent out tm request: $mb\n"); } When I check the log to see my sended frame, I don't see a contact header field : INFO: <script>: Append new contact ! INFO: <script>: Sent out tm request : OPTIONS sip:sip.pstnhub.microsoft.com;transport=tls SIP/2.0 Via: SIP/2.0/TLS MY_SBC_NAME.COM:5061 ;branch=z9hG4bK4497.f3e5e754000000000000000000000000.0 To: <sip:sip.pstnhub.microsoft.com;transport=tls> From: <sip:MY_SBC_NAME.COM.fr>;tag=a76e006ace3802951ed860e08df964af-7971 CSeq: 10 OPTIONS Call-ID: 2e1278ea77796160-9837@MY_SBC_IP Max-Forwards: 70 Content-Length: 0 User-Agent: kamailio (5.2.1 (x86_64/linux)) According the documentation about textop, append_hf(txt[, hdr]) : This function can be used from REQUEST_ROUTE, ONREPLY_ROUTE, FAILURE_ROUTE, BRANCH_ROUTE. So my question is append HF can be used in Event_route ? Is there any other way to add contact header ? Regards, Arnaud

2 years, 8 months

1
0
0 0

g_hash_table_insert_internal: assertion 'hash_table != NULL' failed

by Muhammad Danish Moosa

Hi, I am running stress tests on kamalilio , at times i see this on screen (process:30959): GLib-CRITICAL **: 21:43:56.961: g_hash_table_insert_internal: assertion 'hash_table != NULL' failed version: kamailio 5.5.3 Kamailio does not crash and keeps accepting calls though. Is it anything alarming. -- Muhammad Danish Moosa " The core of mans' spirit comes from new experiences. "___ Christopher McCandless

2 years, 8 months

1
0
0 0

dispatcher lb

by Gokan Atmaca

Hello I'm load balancing with Dispatcher. The call comes in but hangs up without picking up the phone. What could be the problem? The settings are as follows. dispatcher.list 1 sip:18.X.X.X.X:5060 0 0 maxload=20 1 sip:67.X.X.X.X:5060 0 0 maxload=20 # Dispatch requests route[DISPATCH] { # round robin dispatching on gateways group '1' if(!ds_select_dst("1", "4")) { send_reply("404", "No destination"); exit; } xlog("L_DBG", "--- SCRIPT: going to <$ru> via <$du>\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } { NRSETS: 1 RECORDS: { SET: { ID: 1 TARGETS: { DEST: { URI: sip:X.X.X.X:5060 FLAGS: AP PRIORITY: 0 ATTRS: { BODY: maxload=20 DUID: MAXLOAD: 20 WEIGHT: 0 RWEIGHT: 0 SOCKET: } LATENCY: { AVG: 4.130000 STD: 0.398000 EST: 4.006000 MAX: 6 TIMEOUT: 0 } } DEST: { URI: sip:X.X.X.X:5060 FLAGS: AP PRIORITY: 0 ATTRS: { BODY: maxload=20 DUID: MAXLOAD: 20 WEIGHT: 0 RWEIGHT: 0 SOCKET: } LATENCY: { AVG: 22.282000 STD: 2.345000 EST: 23.075000 MAX: 42 TIMEOUT: 0 } } } } } } -- ⢀⣴⠾⠻⢶⣦⠀ ⣾⠁⢠⠒⠀⣿⡁ Debian - The universal operating system ⢿⡄⠘⠷⠚⠋⠀ https://www.debian.org ⠈⠳⣄⠀⠀⠀⠀

2 years, 8 months

2
3
0 0

Effect of GRUU on the behavior of loose_route()

by Denys Pozniak

Hello! I ask for help in explaining the different behavior of loose_route() function in two similar cases (depending on GRUU parameter on the end device). The call path goes through a series of Proxies: Proxy Registrar (use Path) -> Proxy Edge -> PhonerLite (GRUU on/off). If the GRUU parameter is enabled on the PhonerLite, then the in-dialog SIP ACK (on 200 OK/SDP) message is routed on the Proxy Edge successfully (loose_route() deletes Route and handle_ruri_alias() sets $du). When GRUU is disabled - loose_route() for some reason replaces RURI (but lr=on ?) and sets $du. The configuration of the [WITHINDLG] route section is similar to the default script. IP: 1.1.1.1 is the Proxy Registrar IP: 2.2.2.2 is the Proxy Edge IP: 3.3.3.3 is the Phone (Phoner Lite) Successful case (GRUU enabled): 2022/01/21 20:08:20.745859 1.1.1.1:5060 -> 2.2.2.2:5060 ACK sip:userid@2.2.2.2:5060;gr=009356C7-5A79-EC11-8468-538E88103B96;alias=3.3.3.3~5065~1 SIP/2.0 Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bK0a34.0bcc52c86238bd3a73bd88712c562d10.0 Via: SIP/2.0/UDP 10.60.27.126;rport=5060;branch=z9hG4bK0a34.1bb19d987d7566fe74d300766284d73a.0 Via: SIP/2.0/UDP 10.60.27.84:5080 ;received=10.60.27.84;branch=z9hG4bKkr2inaF~;rport=5080 From: "Denys VOIP" <sip:5009@1.1.1.1>;tag=41A6E617-61EB049F000008A2-86EEE700 To: <sip:userid@10.60.27.84>;tag=00fd16295b79ec118469538e88103b96 CSeq: 10 ACK Call-ID: 537f0686-f590-123a-ffb5-001dd8b71cb2_userid Route: <sip:2.2.2.2;lr=on;nat=yes;did=28c.4fa> Max-Forwards: 63 Contact: <sip:10.60.27.84:5080;transport=udp> Content-Length: 0 2022/01/21 20:08:20.746952 2.2.2.2:5060 -> 3.3.3.3:5065 ACK sip:userid@2.2.2.2:5060;gr=009356C7-5A79-EC11-8468-538E88103B96 SIP/2.0 Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK0a34.2b5076a5473cc42d883bf38969aa7dcf.0 Via: SIP/2.0/UDP 1.1.1.1;rport=5060;branch=z9hG4bK0a34.0bcc52c86238bd3a73bd88712c562d10.0 Via: SIP/2.0/UDP 10.60.27.126;rport=5060;branch=z9hG4bK0a34.1bb19d987d7566fe74d300766284d73a.0 Via: SIP/2.0/UDP 10.60.27.84:5080 ;received=10.60.27.84;branch=z9hG4bKkr2inaF~;rport=5080 From: "Denys VOIP" <sip:5009@1.1.1.1>;tag=41A6E617-61EB049F000008A2-86EEE700 To: <sip:userid@10.60.27.84>;tag=00fd16295b79ec118469538e88103b96 CSeq: 10 ACK Call-ID: 537f0686-f590-123a-ffb5-001dd8b71cb2_userid Max-Forwards: 62 Contact: <sip:10.60.27.84:5080;transport=udp> Content-Length: 0 DEBUG: 3(11755) DEBUG: rr [loose.c:85]: is_preloaded(): is_preloaded: No 3(11755) DEBUG: <core> [core/socket_info.c:646]: grep_sock_info(): checking if host==us: 13==13 && [2.2.2.2] == [2.2.2.2] 3(11755) DEBUG: <core> [core/socket_info.c:649]: grep_sock_info(): checking if port 5060 (advertise 0) matches port 5060 3(11755) DEBUG: <core> [core/forward.c:429]: check_self(): host (0: 2.2.2.2:5060) == me 3(11755) DEBUG: <core> [core/socket_info.c:646]: grep_sock_info(): checking if host==us: 13==13 && [2.2.2.2] == [2.2.2.2] 3(11755) DEBUG: <core> [core/socket_info.c:649]: grep_sock_info(): checking if port 5060 (advertise 0) matches port 5060 3(11755) DEBUG: <core> [core/forward.c:429]: check_self(): host (0: 2.2.2.2:5060) == me 3(11755) DEBUG: rr [loose.c:856]: after_loose(): Topmost route URI: 'sip:2.2.2.2;lr=on;nat=yes;did=28c.4fa' is me 3(11755) DEBUG: <core> [core/parser/msg_parser.c:91]: get_hdr_field(): found end of header 3(11755) DEBUG: rr [loose.c:181]: find_next_route(): No next Route HF found 3(11755) DEBUG: rr [loose.c:886]: after_loose(): No next URI found 3(11755) DEBUG: rr [rr_cb.c:95]: run_rr_callbacks(): callback id 1 entered with <lr=on;nat=yes;did=28c.4fa> 3(11755) DEBUG: path [path.c:397]: path_rr_callback(): no received parameter in route header 3(11755) DEBUG: rr [rr_cb.c:95]: run_rr_callbacks(): callback id 0 entered with <lr=on;nat=yes;did=28c.4fa> 3(11755) DEBUG: nathelper [nathelper.c:1202]: handle_ruri_alias(): setting dst_uri to <sip:3.3.3.3:5065> 3(11755) DEBUG: nathelper [nathelper.c:1224]: handle_ruri_alias(): rewriting r-uri to <sip:userid@2.2.2.2:5060 ;gr=009356C7-5A79-EC11-8468-538E88103B96> Unsuccessful case (GRUU disabled): 2022/01/21 20:08:47.851170 1.1.1.1:5060 -> 2.2.2.2:5060 ACK sip:userid@2.2.2.2:5060;alias=3.3.3.3~5065~1 SIP/2.0 Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bK110e.a47bb0e8f6d6d99a10ae36eb25d2f665.0 Via: SIP/2.0/UDP 10.60.27.126;rport=5060;branch=z9hG4bK110e.4d82c337912ca1187474aa2474a7d563.0 Via: SIP/2.0/UDP 10.60.27.84:5080 ;received=10.60.27.84;branch=z9hG4bK4bKmGaAy;rport=5080 From: "Denys VOIP" <sip:5009@1.1.1.1>;tag=0FBBBFFE-61EB04BB000A3B4F-86EEE700 To: <sip:userid@10.60.27.84>;tag=80dc2e395b79ec11846d538e88103b96 CSeq: 10 ACK Call-ID: 64957c76-f590-123a-ffb5-001dd8b71cb2_userid Route: <sip:2.2.2.2;lr=on;nat=yes;did=3f8.6a83> Max-Forwards: 63 Contact: <sip:10.60.27.84:5080;transport=udp> Content-Length: 0 2022/01/21 20:08:47.851963 2.2.2.2:5060 -> 2.2.2.2:5060 ACK sip:2.2.2.2;lr=on;nat=yes;did=3f8.6a83 SIP/2.0 Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK110e.af605388ece54d48387c1de53615d894.0 Via: SIP/2.0/UDP 1.1.1.1;rport=5060;branch=z9hG4bK110e.a47bb0e8f6d6d99a10ae36eb25d2f665.0 Via: SIP/2.0/UDP 10.60.27.126;rport=5060;branch=z9hG4bK110e.4d82c337912ca1187474aa2474a7d563.0 Via: SIP/2.0/UDP 10.60.27.84:5080 ;received=10.60.27.84;branch=z9hG4bK4bKmGaAy;rport=5080 From: "Denys VOIP" <sip:5009@1.1.1.1>;tag=0FBBBFFE-61EB04BB000A3B4F-86EEE700 To: <sip:userid@10.60.27.84>;tag=80dc2e395b79ec11846d538e88103b96 CSeq: 10 ACK Call-ID: 64957c76-f590-123a-ffb5-001dd8b71cb2_userid Max-Forwards: 62 Contact: <sip:10.60.27.84:5080;transport=udp> Content-Length: 0 DEBUG: 1(11753) DEBUG: rr [loose.c:85]: is_preloaded(): is_preloaded: No 1(11753) DEBUG: <core> [core/socket_info.c:646]: grep_sock_info(): checking if host==us: 13==13 && [2.2.2.2] == [2.2.2.2] 1(11753) DEBUG: <core> [core/socket_info.c:649]: grep_sock_info(): checking if port 5060 (advertise 0) matches port 5060 1(11753) DEBUG: <core> [core/forward.c:429]: check_self(): host (0: 2.2.2.2:5060) == me 1(11753) DEBUG: rr [loose.c:676]: after_strict(): Next hop: 'sip:2.2.2.2;lr=on;nat=yes;did=3f8.6a83' is loose router 1(11753) DEBUG: rr [loose.c:726]: after_strict(): The last route URI: 'sip:2.2.2.2;lr=on;nat=yes;did=3f8.6a83' 1(11753) DEBUG: rr [rr_cb.c:95]: run_rr_callbacks(): callback id 1 entered with <alias=3.3.3.3~5065~1> 1(11753) DEBUG: path [path.c:397]: path_rr_callback(): no received parameter in route header 1(11753) DEBUG: rr [rr_cb.c:95]: run_rr_callbacks(): callback id 0 entered with <alias=3.3.3.3~5065~1> 1(11753) DEBUG: rr [record.c:977]: add_rr_param(): rr_param_buf=<;did=3f8.6a83> 1(11753) DEBUG: siputils [checks.c:123]: has_totag(): totag found 1(11753) DEBUG: siputils [checks.c:123]: has_totag(): totag found 1(11753) DEBUG: tm [t_lookup.c:1329]: t_newtran(): msg (0x7efe343f4998) id=11/11753 global id=10/11753 T start=0xffffffffffffffff 1(11753) DEBUG: tm [t_lookup.c:498]: t_lookup_request(): start searching: hash=57361, isACK=1 1(11753) DEBUG: tm [t_lookup.c:456]: matching_3261(): RFC3261 transaction matching failed - via branch [z9hG4bK110e.a47bb0e8f6d6d99a10ae36eb25d2f665.0] 1(11753) DEBUG: tm [t_lookup.c:675]: t_lookup_request(): no transaction found 1(11753) DEBUG: tm [t_funcs.c:286]: t_relay_to(): forwarding ACK statelessly -- BR, Denys Pozniak

2 years, 8 months

1
0
0 0

Experiences with Kamailio on Debian 11 Bullseye

by Daniel-Constantin Mierla

Hello, has anyone experienced issues with Kamailio running on Debian 11 Bullseye? It's now more than 2 months since Debian 11 release and I wonder if people started to migrate to it for production systems. So far I did only basic tests on it, although I have a Debian Testing that I used for compilation and simple tests during the past several months before Debian 11 release. If someone discovers issues or changes in behaviour with various libraries or applications/database servers, report them to the tracker to try to address them as soon as possible. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Online Nov 08-11, 2021 (Europe Timezone) - Nov 22-25, 2021 (America Timezone) * https://www.asipto.com/sw/kamailio-advanced-training-online/

2 years, 8 months

8
14
0 0

Handling special characters - KEMI lua redis_ndb

by Muhammad Danish Moosa

Hi Guys, I am trying to do following code KSR.ndb_redis.redis_cmd("srvN", "SET var " .. testvar, "r"); it works fine but if testvar has special character like SPACE or % , it does not work. For example when testvar="asdsada%Dasda"; it starts considering %D as place holder and does not add key to redis (does not raise error as well) With Space I tried to use delimeters like "" or '' but none of them work. For now I have got it working through , search and replace but thats not a clean solution. Any suggestions? -- Muhammad Danish Moosa " The core of mans' spirit comes from new experiences. "___ Christopher McCandless

2 years, 8 months

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Does cfgutils usleep block the process?

by mayamatakeshi

Hi, it seems execution of usleep eventually reaches this code: https://github.com/kamailio/kamailio/blob/master/src/core/ut.h#L508 so it seems it would block the process and so it might cause delay of packet processing if many processes do this at the same time. Can someone confirm this?

2 years, 8 months

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use $var or $avp in ds_list_exists & ds_select dispatcher function

by Patrick Karton

Hello im using kamailio 5.5.3 and im trying to use pseudo vars in ds_list_exists and ds_select function but it does not work. ds_list_exists return -2 with ds_list_exists("6") it works $var(dset)=6 or $avp(dset)=6 it does not work ds_list_exists("$var(dset)") --> return -2 ds_list_exists("$avp(dset)") --> return -2 Thanks.

2 years, 8 months

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Fwd: Possible memory leak on 5.5.x (new)?

by George Diamantopoulos

Forwarding my reply to the list, using gmail's reply button set Henning as the sole recipient :-\ ---------- Forwarded message --------- From: George Diamantopoulos <georgediam(a)gmail.com> Date: Sat, 26 Jun 2021 at 02:25 Subject: Re: [SR-Users] Possible memory leak on 5.5.x (new)? To: Henning Westerholt <hw(a)skalatan.de> Hello Henning, Thanks for your reply. Here's what has come up after a few hours: shm55: https://pastebin.com/h9JCePmc shm54: https://pastebin.com/Nx5xEEnA It seems to me htable is the culprit? Are you seeing anything different? 54 has been running for 77020 seconds, 55 for 28521 (significantly less). I'm going to turn it off until we figure something out... BR, George On Fri, 25 Jun 2021 at 18:17, Henning Westerholt <hw(a)skalatan.de> wrote: > Hello, > > > > Good observation. Please run the memory statistics CLI commands to get > more hints about the module that might cause it (as per below link). Then > please report more details. If you can point to a particular module, you > can also open an issue on our tracker. > > > > https://www.kamailio.org/wiki/tutorials/troubleshooting/memory > > > > Cheers, > > > > Henning > > > > *From:* sr-users <sr-users-bounces(a)lists.kamailio.org> *On Behalf Of *George > Diamantopoulos > *Sent:* Friday, June 25, 2021 4:53 PM > *To:* Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> > *Subject:* [SR-Users] Possible memory leak on 5.5.x (new)? > > > > Hello all, > > > > I'm still investigating the (most likely non-kamailio-related) memory leak > of my previous message to the list, there have been no developments so far. > I'll update if anything changes. > > > > This concerns a new finding which seems to affect kamailio 5.5.x. I have > two kamailio instances receiving the same traffic via round-robin. I > upgraded only one of them to 5.5.1 and left the other to 5.4.6 as I feared > of any issues arising. I was lucky to do so, because with identical > configuration, 5.5.x seems to run out of SHM very quickly. Here are links > to graphs produced by our monitoring system: > > > > Old kamailio (no memory leak): https://pasteboard.co/K8fVBiD.png > > New kamailio (possible leak): https://pasteboard.co/K8fVS9N.png > > > > The configuration uses mtree, htable, vars and vns extensively. Has anyone > come across anything similar? Let me know if I can provide any further > information to help disect this. Thanks! > > > > BR, > > George >

2 years, 8 months

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Issues running mongodb command from kamailio

by Edo

Hi All, Please help! I am getting an error when trying to start kamailio after making some changes to the configuration file. Error is : *rse error in config file /etc/kamailio/kamailio.cfg, line 878, column 47-57: '('')' expected (function call)* *rse error in config file /etc/kamailio/kamailio.cfg, line 878, column 102: bad expression* *rse error in config file /etc/kamailio/kamailio.cfg, line 878, column 102: bad command* *atus=255/EXCEPTION* *rse error in config file /etc/kamailio/kamailio.cfg, line 878, column 103: bad command* *rse error in config file /etc/kamailio/kamailio.cfg, line 878, column 105: bad command* * used inside params of another function: xlog* *rse error in config file /etc/kamailio/kamailio.cfg, line 879, column 28: use of function execution inside params not allowed* *Server.* Changes made is : # Routing to foreign domains route[SIPOUT] { if (uri==myself) return; #Start of change if (mongodb_find_one("mongodbsrv1", “telephone", “cXXX", " { \'did\': \'2XXXXXXXXXXXXXXXXX\' } ", "mgr1")) { xlog("response from mongodb is [[$mongodb(mgr1=>value)]]\n"); } #End of change append_hf("P-hint: outbound\r\n"); route(RELAY); exit; } Please can someone tell me what I am doing wrong. Error says syntax but I have tried many permutations of the suggested syntax with no luck. Please help Thanks -- ----- Ekunwe EDO Network Services, Inc. Tel: 601.497.3932

2 years, 8 months

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REGISTER for MS Teams

by Lucian Romi

Hi, can register send from ms teams or I have to use Kamailio to send to my PBX.

2 years, 8 months

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JS Kemi TypeError: undefined not callable (property 'ifdef' of [object Object])

by Huseyin Kaan Koyukan

Hello Everyone, First of all, this is my first ever e-mail to the group so I'd like to thank all the developers who built this great product and thank everyone for the kind and helpful community. I've been using Kamailio for the past 8 months, I've had lots of fun with it. My issue is I'm trying to set-up JS Kemi interpreter for my Kamailio-Asterisk box. I translated the config file to JS kemi, which wasn't too bad but I'm getting a type error for the "KSR.kx.ifdef("WITH_ASTERISK")" statement that I used. Jan 17 15:11:26 anayurt kamailio[8261]: 0(8261) DEBUG: app_jsdt [app_jsdt_api.c:1478]: dukopen_KSR(): module 'KSR' has been initialized Jan 17 15:11:26 anayurt kamailio[8261]: 0(8261) DEBUG: app_jsdt [app_jsdt_api.c:473]: jsdt_sr_init_child(): loading js script file: /usr/local/etc/kamailio/kamailio-basic-kemi-jsdt.js Jan 17 15:11:26 anayurt kamailio[8261]: 0(8261) ERROR: app_jsdt [app_jsdt_api.c:447]: jsdt_kemi_load_script(): failed running: TypeError: undefined not callable (property 'ifdef' of [object Object]) Jan 17 15:11:26 anayurt kamailio[8261]: 0(8261) ERROR: <core> [core/sr_module.c:865]: init_mod_child(): error while initializing module app_jsdt (/usr/local/lib64/kamailio/modules/app_jsdt.so) (idx: 0 rank: -127 desc: [main]) Jan 17 15:11:26 anayurt kamailio[8261]: 0(8261) ERROR: <core> [main.c:1671]: main_loop(): error in init_child(PROC_INT) -- exiting In syslog I can see that kex.so module is loaded successfully and 'KSR' has been initialized. I don't know what I'm doing wrong. A few sample JS Kemi code that I could find did not use the kx.ifdef function. Was this a new addition to the interpreter? Perhaps my Kamailio version (kamailio 5.5.0 x86_64/linux) does not support it? Thank you all in advance. Kaan.

2 years, 8 months

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Help with rewriting headers for NAT manually

by Chad

We are looking for some help (possibly a paid consultant) to help us with our Kamailio setup. To keep this as short as possible: we use Kamailio as a NAT proxy to bridge our external IP and our private IP asterisk servers (via dispatcher). However both the external IP and the internal IP that the Kamailio server uses are virtual IPs created by keepalived. Because of that neither mhomed nor fix_nated_contact work, and we use force_send_socket to direct the traffic. We run linux Debian 10 for the OS. Also we do not use a DB at all, everything is done with local config files. The problem is that when traffic goes out the Contact header has a private IP in it, like: Contact: <sip:##########@10.10.10.###]:5060> There are 2 possible solutions to this: 1. Make changes to linux, keepalived and/or Kamailio so that Kamailio recognize the virtual IPs so that mhomed and fix_nated_contact work as usual. 2. Create a manual header rewrite system. If solution #2: What we need to do is create a way to rewrite the contact header to the external IP on the way out, and on the way back rewrite it back to the internal server that the call is already connected to. Not sure if we will need to store those paths on the server or if we can do some kind of cheat with another persistant header like P-Preferred-Identity or P-Asserted-Identity (i.e. store the internal IP in the name field or something). If anyone out there know of a way to do this or wants to give it a try please reach out to me. Thank you all for your time. -- ^C Chad

2 years, 8 months

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Alternate to FoHSS for Kamailio IMS?

by Simon,Benjamin T

Hello, We are attempting to setup an instance of Kamailio to serve as an IMS for a test setup of 4G/5G NSA with Open5GS. The instructions on the Kamailio wiki and provided by Open5GS currently specify to use FoHSS as the HSS solution for Kamailio IMS. As this HSS solution has not seen significant development in a few years we are reticent to deploy it on our implementation. Is there an alternative open source HSS which can be deployed to serve as the HSS for Kamailio instead of FoHSS, and would there need to be any adjustments to Kamailio to reflect the deployment of this HSS instead of FoHSS? Thank you, Benjamin Simon Research Assistant Florida Institute for Cybersecurity Research, University of Florida Email: bsimon4(a)ufl.edu Disclaimer: Florida has a very broad public records law. As a result, any written communication created or received by the University of Florida will be made available to the public and media, upon request, unless otherwise exempt. Under Florida law, e-mail addresses are public records. If you do not want your e-mail address released in response to a public records request, do not send electronic mail to this address. Instead, contact by phone or in writing.

2 years, 8 months

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RR balance kamailio

by Gokan Atmaca

Hello I installed kamailio on Ubuntu. The version information is as follows. (0) I added two dispatcher sip servers. (1) When I enter the parameter as below in the configuration file , I get an error as follows. (2) I am getting RR module not available error. Packages installed are "kamailio-berkeley-modules, kamailio-tls-modules, kamailio-presence-modules" How can I solve the problem? (0) version: version: kamailio 5.5.3 (x86_64/linux) flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, 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_BLOCKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled with gcc 9.3.0 (1) Dispatcher list: { NRSETS: 1 RECORDS: { SET: { ID: 1 TARGETS: { DEST: { URI: sip:X.X.X.X FLAGS: AP PRIORITY: 0 LATENCY: { AVG: 0.000000 STD: 0.000000 EST: 0.000000 MAX: 0 TIMEOUT: 0 } } DEST: { URI: sip:Y.Y.Y.Y FLAGS: AP PRIORITY: 0 LATENCY: { AVG: 0.000000 STD: 0.000000 EST: 0.000000 MAX: 0 TIMEOUT: 0 } } } } } } (2) if ( method=="INVITE" ) { ds_select_dst("1","4"); sl_send_reply("100","Trying"); forward(); exit(); } Err_log: Jan 12 12:15:11 kamailio /usr/sbin/kamailio[21241]: INFO: rr [../outbound/api.h:52]: ob_load_api(): unable to import bind_ob - maybe module is not loaded Jan 12 12:15:11 kamailio /usr/sbin/kamailio[21241]: INFO: rr [rr_mod.c:188]: mod_init(): outbound module not available -- ⢀⣴⠾⠻⢶⣦⠀ ⣾⠁⢠⠒⠀⣿⡁ Debian - The universal operating system ⢿⡄⠘⠷⠚⠋⠀ https://www.debian.org ⠈⠳⣄⠀⠀⠀⠀

2 years, 8 months

2
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Mid-dialog failover without virtual IP

by Muhammad Danish Moosa

Hi, I have a pair of kamailio stateful proxies and am trying to achieve Active-Passive failover model with DMQ dialog replication. Primary Proxy is working in the middle of 2 SBCs and all nodes are within the same network so NATing is NOT involved ;secondary proxy is accessible through VPN. It's pretty simple configuration, proxy is just adding and removing headers in the middle. So if I stop the kamailio service on primary IP , SBCs blacklist that and start sending new invites to secondary IP. That's all good. But if I reproduce this failover during a connected session , SBCs do not send mid-dialog requests/responses to failover IP. I can see they still try to send messages to primary SBC which obviously end up with timeout. I understand the most natural solution is HA with Virual IP but it's not possible because secondary IP is accessible through VPN. I tried to use DNS SRV record with priorities but that also does not work with mid-dialog messages, I also have tried to change record-route/via header to fqdn through advertise_address. Its not changing record-route but via only but I am still not getting expected results. Any guidance of possible solution(s) here will be appreciated. -- Muhammad Danish Moosa " The core of mans' spirit comes from new experiences. "___ Christopher McCandless

2 years, 8 months

3
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Massive uploading of aliases through a dbtext

by Denys Pozniak

Hello! I need help uploading a list of aliases through a dbtext to use them via myself. It is likely that there is some configuration problem: *loadmodule "db_text.so"* *modparam("db_text", "db_mode", 1)modparam("db_text", "emptystring", 1)* *loadmodule "alias_db.so"modparam("alias_db", "db_url", "text:///etc/kamailio/dbtext")* *[root@kamailio]$ cat dbtext/dbaliasesid(int,auto) alias_username(string) alias_domain(string) username(string) domain(string)1::sip.test.local::* *[root@kamailio]$ kamcmd core.aliases_list{ myself_callbacks: yes aliases: { }}* If done through a script (alias=sip.test.local), then everything works fine: *[root@kamailio]$ kamcmd core.aliases_list{ myself_callbacks: yes aliases: { alias: { proto: * address: sip.test.local port: * } }}* -- BR, Denys Pozniak

2 years, 8 months

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Retrieve last via IP from SIP MEssage

by Adarsh Chauhan

Hi All, I have tried couple of methods to retrieve last via IP of SIP message but failed to retrieve it, trying to retrieve the source - @hf_value.via[-1].host, it simply prints the exact text back @hf_value.via[-1].host - $sel((a)hf_value.via[-1]) it doest give the IP but the entire via value SIP/2.0/TCP 172.31.30.144;branch=z9hG4bKf8bd.753341d2000000000000000000000000.0.host @via[-1].host it also simply prints the exact text back @via[-1].host Regards, Adarsh Chauha

2 years, 8 months

2
1
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