- sr-users - kamailio.org

by Benoît Panizzon

Heya I try to use mutli tree mtree to define multiple 'named' prefixes. id(int,auto) tname(string) tprefix(string) tvalue(string) 0:tech:+1:1 0:tech:+9:1 0:cust:+1:1 0:cust:+9:0 Kamailio is not complaining while loading this file, but mtree.summary shows no entries. Any hint on what I may be do wrong? -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

2 years, 1 month

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Kamailio Configuration Help

by P. White

Hi All, Really sorry to bother everyone, but if possible, can I ask for a little help? I've been attempting to get Kamailio configured and working for some considerable time now (read: Years of occasional tinkering, and weeks of recent dedicated evening work) -- all without success. I have a sip trunk from sipgate, with two PSTN numbers assigned. My local network has a static external IP, with an internal private network. Currently I have Kamailio running within the intenal network along with two physical sip telephones (gigaset and an old grandstream), although I do have the option to put kamailio within a DMZ if necessary. Would anyone have a working config file that they could share please, permitting incoming and outgoing calls? With grateful thanks, Phil

2 years, 1 month

2
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Where to find DMQ protocol specification

by Илья Болмосов

Hello, we use DMQ module and we need to handle some events: 1) when peer joins cluster 2) when peer leaves cluster It will be helpful to know which content will be in KDMQ message in different cases. Tried to find any specification of KDMQ protocol, but didn't find it. Can you, please, help me with this?

2 years, 1 month

2
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Change reply code reason other than using change_reply_status

by Patrick Karton

hello is there a way to change reply code reason other than using () function. because the first parameterc of change_reply_status needs to be a constant so i can not use pseudo var for first argument of that function. so i use change_reply_status_code to change to code with a pseudo. but i would also to change the code reason.

2 years, 1 month

3
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remove_hf() does not remove the complete line

by Bernd Krueger-Knauber

Hi, I tried to remove a Via header line with: remove_hf("Via"); Original I have this 2 lines: Via: SIP/2.0/TCP xxx.xxx.xxx.xxx;branch=z9hG4bK370b.21720bb33c4f8264fde4e7994d288a57.0;i Via: SIP/2.0/TCP yyy.yyy.yyy.yyy:53967;received=yyy.yyy.yyy.yyy;rport=53967;branch=z9hG4bKPjxb0bdb2c9624076815f12937cc9e51d;alias If I insert remove_hf("Via"); in the cfg, I get: Via: SIP/2.0/TCP xxx.xxx.xxx.xxx;branch=z9hG4bK370b.21720bb33c4f8264fde4e7994d288a57.0;i ;received=yyy.yyy.yyy.yyy;rport=53967;branch=z9hG4bKPjxb0bdb2c9624076815f12937cc9e51d;alias Doesn't mattter if the command make sense or not, the result, in my opinion, is wrong. The INVITE packet has now an invalid header line. Any ideas? version: kamailio 5.6.1 (x86_64/linux) Best regards, Bernd

2 years, 1 month

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Kamailio has unconfined processes

by HimaBindu G

Hi, Problem Description: Customer security scan returned unconfined services on Kamailio. Unconfined processes run in unconfined domains Rationale: For unconfined processes, SELinux policy rules are applied, but policy rules exist that allow processes running in unconfined domains almost all access. Processes running in unconfined domains fall back to using DAC rules exclusively. If an unconfined process is compromised, SELinux does not prevent an attacker from gaining access to system resources and data, but of course, DAC rules are still used. SELinux is a security enhancement on top of DAC rules - it does not replace them Solution Investigate any unconfined processes found during the audit action. They may need to have an existing security context assigned to them or a policy built for them. Notes: Occasionally certain daemons such as backup or centralized management software may require running unconfined. Any such software should be carefully analyzed and documented before such an exception is made. See Also https://workbench.cisecurity.org/files/2485 For Kamailio ====== The command returned : 00 kamailio 00 kamailio 00 kamailio 00 kamailio 00 kamailio 10 kamailio 10 kamailio 10 kamailio 10 kamailio 00 kamailio 00 kamailio 00 kamailio 00 kamailio 33 kamailio 33 kamailio 33 kamailio 32 kamailio 17 kamailio 16 kamailio 33 kamailio 00 kamailio 00 kamailio 03 kamailio 05 kamailio 18 kamailio 17 kamailio 18 kamailio 18 kamailio 07 kamailio 00 sleep is any security context available to assign kamailio processes ? theses services can be run as confined services ? Please suggest us with resolution, thanks in advance. Thanks & Regards, Hima Bindu.

2 years, 1 month

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Kamailio World Conference - Sep 7-8, 2022 - Online

by Daniel-Constantin Mierla

For the 3rd year, we are planning an online version of Kamailio World Conference! At the beginning of 2022, the situation was not clear in terms of whether pandemic would decrease enough in intensity and we would be allowed to organize an on-site event late spring/beginner of summer, when the usual Kamailio Conference took place in Berlin, therefore we chose the safety path. But we hope that next year we will return in-person to the beautiful Berlin city center! The event this year, to happen in 4 weeks from now (Sep 7-8, 2022), follows the structure from previous edition, being two afternoons with presentations about Kamailio, sharing technical solutions and open discussions, allowing the community to interact and reconnect. The website of the event: - https://kamailioworld.com/k10-online/ There is no registration required, the event will happen on live video conferencing, also streamed to youtube. A text chatting channel will be made available as well. If you want to present, just send me the title and a short description -- we encourage community members to step forward and share their knowledge. It has to be related to Kamailio and fit within a 30min slot (including the questions). The agenda will be released in the near future. The event will be used also to celebrate 21 years of Kamailio project development, first commit being done on Sep 3, 2001. Two other major celebrations for 2022 to make the party bigger: - 20 years of open source for the project -- after one year of internal development, GPL was added on Sep 19, 2002 and source code was released to public - the 10th edition of Kamailio World Conference Book the dates in your agenda! I am looking forward to a very interesting event and (re-)connecting with many of you! Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

2 years, 1 month

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set source port of TCP socket in Contact header

by Maarten Ureel

Hello I want to set the correct source port in an outgoing message over TCP. Is it possible? I can't find a pseudo-variable for this. Currently we have no port number in the contact header, and the other party is sending their BYE to port 5060 instead of to the source port from which the outgoing TCP connection was made. Regards Maarten

2 years, 1 month

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Syncing Registration Database across Kamailio Pairs

by harneet singh

Hi Experts, We have been using Kamailio in an Active/Standby Pair(with Keepalived under the rugs moving the single Virtual IP to access the Active Kamailio) for sometime now. *Kamailio also acts as a Registrar for our webrtc endpoints. * It has been serving the purpose pretty well and now we have a requirement where we need to be syncing the Registration DB between Two Pairs of Kamailios. Kamailio-Active(*Pair 1*) -------- Kamailio-Standby(*Pair 1*) || Kamailio-Active(*Pair 2*) -------- Kamailio-Standby(*Pair 2*) We generally keep the counterpart in the same Pair as a notifier( *modparam("dmq", "notification_address", "sip:PEER1_IP:5060")* ) so as to sync the dialogs and the userloc data too( *modparam("dmq_usrloc", "enable", 1) * ). In order to achieve the said requirement with the other Pair, we added another "*notification_address*" in the kamailio cfg. At this point, we ran into weird issues. *1. *With Kamailio ver *5.3.2, *the subsequent *notification_address *line in the cfg file, seemed to be overriding the previous one. Hence we see only the latter peer in the dmq list nodes. Example: modparam("dmq", "notification_address", "sip:*172.27.45.77* :5090") modparam("dmq", "notification_address", "sip:*172.27.45.200* :5090") In this case, the "*kamcmd dmq.list_nodes*" would show the local Machine and 172.27.45.200 as the only nodes in the output, ie *172.27.45.77 is not showing up at all,* which is problematic, since the local machine(172.27.45.243 in our case) would not been able to send any dmq sync info to its peer(172.27.45.77) in the same Pair. To see if the above issue might have been addressed in later release, we upgraded to the latest Kamailio ver *5.6.0* *To our respite, the above issue no longer exists in the new version*(though not sure which immediate release after v 5.3.2 it would have been initially fixed.) This is where we have a new issue explained below: *2. *The registration data does get synced to the peer Kamailio in the same Pair, and also to the Kamailio instances in the other Pair. However, the *Socket* Parameter in "*kamctl ul show*" output shows *[not set] *even on the side where the websocket connection actually exists. [root@localhost ~]# *kamctl ul show * { "jsonrpc": "2.0", "result": { "Domains": [{ "Domain": { "Domain": "location", "Size": 1024, "AoRs": [{ "Info": { "AoR": "9008077221", "HashID": 1952082106, "Contacts": [{ "Contact": { "Address": "sip:Harneet_qifir@172.24.58.210", "Expires": 159, "Q": -1, "Call-ID": "vfli2uv8du3ppda73q5ppe", "CSeq": 106, "User-Agent": "EngageDigital", "Received": "sip:172.27.44.252:60070;transport=ws", "Path": "[not set]", "State": "CS_NEW", "Flags": 0, "CFlags": 0, * "Socket": "[not set]", <<<<<<<<<<<<<<<<<<<<<<* "Methods": 7071, "Ruid": "uloc-62f26e3b-2677-1", "Instance": "<urn:uuid:4fbd3e96-b4de-497b-886d-1ca8ffa016a4>", "Reg-Id": 1, "Server-Id": 0, "Tcpconn-Id": -1, "Keepalive": 0, "Last-Keepalive": 1660063892, "KA-Roundtrip": 0, "Last-Modified": 1660063892 } }] } } ], In order to confirm that the socket actually exists on this Kamailio instance, I am pasting the below outputs from the same machine, where ws dump and even the native netstat confirms that. [root@localhost ~]# *netstat -tunelap | grep 60070* tcp 0 0 172.27.45.199:8080 172.27.44.252:*60070* *ESTABLISHED* 994 17322355 16230/kamailio [root@localhost ~]# *kamcmd ws.dump* { connections: { 1: ws:172.27.44.252:*60070* -> ws:172.27.45.199:8080 (state: *OPEN*, last used 24s ago, sub-protocol: sip) } info: { wscounter: 1 truncated: no } } We do need to distinguish the actual Kamailio instance where the websocket connection actually exists, so as to route the call ahead from the same instance, or if it does not exist(and it's merely a sync'ed registration data received over DMQ Channel), then the Kamailio should route the call ahead to the Kamailio instance in the other Pair, which can then route it ahead to the Registered webrtc endpoint. We were hoping to use the Socket Parameter output, but for the said problem, unable to use the same as an indicator. *So what would be the best way to identify which Kamailio has the websocket connection with the Actual endpoint*? Should we rely on the output of netstat or ws.dump to infer that? I mean this needs to be done in kamailio.cfg for each call, so want to know the best way, or if there is completely different approach that can be suggested? Apologies for the long email, but any pointers will be much helpful. Thanks & Regards, Harneet Singh -- "Once you eliminate the impossible, whatever remains, no matter how improbable, must be the truth" - Sir Arthur Conan Doyle

2 years, 1 month

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recompute distribution list

by David Villasmil

Hello guys, Is it possible to have dispatcher recompute the distribution list? i.e. one server goes down, all traffic for that server goes to the next one, doubling traffic on that one. Is there a way of, when a host becomes unreachable, recompute the whole list? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 years, 1 month

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Contact header modification - Kamailio

by Ankit Jayswal

Hello Team, I am in need to modify the contact header in Kamailio before sending an INVITE request to the SIP trunk provider. Basically the IP address I need to modify in the contact header. Please suggest a better way to do this. Below is an example. *OLD (Existing):* Contact: <sip:009169240xxx@*45.118.162.x:5080*;alias=10.52.26.102~5080~1> *NEW (Needed):* Contact: <sip:00912269240xxx@*10.52.26.98:5060 <http://10.52.26.98:5060>*> I tried various functions from the textops module but no hope. I am looking for a function that can replace the contact header's value. Please help me with this. -- Thanks & Regards, *Ankit Jayswal* | Specialist - Software Development

2 years, 2 months

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Support of ECDHE cipher suites for tls connection in kamailio

by Илья Болмосов

Hello. I have a question about support of ECDHE cipher suites in kamailio-5.6.0 in centos7 with installed OpenSSL 1.0.2k-fips. We received kamailio with its modules from https://rpm.kamailio.org/. Our client can use only cipher suites: TLS_AES_256_GCM_SHA384 (0x1302) TLS_CHACHA20_POLY1305_SHA256 (0x1303) TLS_AES_128_GCM_SHA256 (0x1301) TLS_ECDHE_ECDSA_WITH_AES_256_CBC_SHA (0xc00a) TLS_ECDHE_RSA_WITH_AES_256_CBC_SHA (0xc014) TLS_ECDHE_ECDSA_WITH_AES_128_CBC_SHA (0xc009) TLS_ECDHE_RSA_WITH_AES_128_CBC_SHA (0xc013) TLS_EMPTY_RENEGOTIATION_INFO_SCSV (0x00ff) And some of them are supported by our openssl: $ openssl cipher -V ... 0xC0,0x14 - ECDHE-RSA-AES256-SHA SSLv3 Kx=ECDH Au=RSA Enc=AES(256) Mac=SHA1 0xC0,0x0A - ECDHE-ECDSA-AES256-SHA SSLv3 Kx=ECDH Au=ECDSA Enc=AES(256) Mac=SHA1 ... 0xC0,0x13 - ECDHE-RSA-AES128-SHA SSLv3 Kx=ECDH Au=RSA Enc=AES(128) Mac=SHA1 0xC0,0x09 - ECDHE-ECDSA-AES128-SHA SSLv3 Kx=ECDH Au=ECDSA Enc=AES(128) Mac=SHA1 ... But when trying to connect with, for example, cipher suite ECDHE-RSA-AES256-SHA (the same with other 3 cipher suites), we receive, that it is not allowed: $ openssl s_client -connect ${kamailio-serper-ip}:${kamailio-server-port} -cipher ECDHE-RSA-AES256-SHA ... SSL handshake has read 7 bytes and written 121 bytes --- New, (NONE), Cipher is (NONE) Secure Renegotiation IS NOT supported Compression: NONE Expansion: NONE No ALPN negotiated SSL-Session: Protocol : TLSv1.2 Cipher : 0000 ... To exclude the influence of other factors, I installed nginx on the same machine with usage of the same tls certificate and it can use cipher suites ECDHE-RSA-AES256-SHA and ECDHE-RSA-AES128-SHA. $ openssl s_client -connect ${nginx-serper-ip}:${nginx-server-port} -cipher ECDHE-RSA-AES256-SHA ... SSL handshake has read 3271 bytes and written 406 bytes Verification: OK --- New, TLSv1.0, Cipher is ECDHE-RSA-AES256-SHA Server public key is 2048 bit Secure Renegotiation IS supported Compression: NONE Expansion: NONE No ALPN negotiated SSL-Session: Protocol : TLSv1.2 Cipher : ECDHE-RSA-AES256-SHA ... So, the reason of failed handshake is, probably, kamailio. Tried to add cipher_list modparam: modparam("tls", "cipher_list", "ECDHE-RSA-AES256-SHA") but result is the same: $ openssl s_client -connect ${kamailio-serper-ip}:${kamailio-server-port} -cipher ECDHE-RSA-AES256-SHA ... SSL handshake has read 7 bytes and written 121 bytes --- New, (NONE), Cipher is (NONE) Secure Renegotiation IS NOT supported Compression: NONE Expansion: NONE No ALPN negotiated SSL-Session: Protocol : TLSv1.2 Cipher : 0000 ... Can you, please, help me to add support of cipher suites ECDHE-RSA-AES256-SHA and ECDHE-RSA-AES128-SHA to kamailio?

2 years, 2 months

2
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Newbie - Installation Issue - ERROR: could not load the script in /usr/lib/kamailio//kamctl/kamdbctl.mysql for database engine MYSQL

by p.t.crossley＠gmail.com

I suspect this must have been addressed before but I couldn't find a way of searching Archives. I'm using Alpine Linux. Used "apk add kamailio" to install Kamailio 5.5.4-r1 Following the Instructions on https://kamailio.org/docs/tutorials/5.5.x/kamailio-install-guide-git/ At the section for "Creating MySQL database" I modify the file and then run kamdbctl create ERROR: could not load the script in /usr/lib/kamailio//kamctl/kamdbctl.mysql for database engine MYSQL ERROR: database engine not loaded - tried 'MYSQL' I checked the /usr/lib/kamailio//kamctl/ directory and the file wasn't there I suspect I'm missing a module but there aren't any other packages listed for Kamailio at pkgs.alpinelinux.org Thanks for your help P Crossley

2 years, 2 months

3
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Limit cps per destination gw

by Jon Bonilla (Manwe)

Hi I'm thinking on having a cps preference for each pstn gw that I can check. I've started thinking in the pike module but it won't allow me to set custom cps per peer. seems that the threshold is global. Ratelimit module needs the queues to be set as modparam, not dynamic AFAICS. Any hints? cheers, Jon -- PekePBX, the multitenant PBX solution https://pekepbx.com

2 years, 2 months

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Sharing Database between three kamailios

by Waqar 40

I have three Kamailios in three different servers which are dispatching calls to 9 asterisks in three different servers using load balancing. All these Kamailios are using databases installed on that servers. As the entries of the dispatcher table are the same for all the Kamailio, so I want that these Kamailio use a single database installed on one of the servers. All these Kamailios instances should use the same database which is common between them. And also I want these Kamailios to write into the database by creating a table that stores all the active calls in three Kamailios. Is it possible? How can I do it? Please guide me with somewhat explanation as I am not this much expert in Kamailio. Waiting for a quick response. Regards VoIP Engineer Vicky

2 years, 2 months

3
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Kamailio behind NAT

by Алексей Якимкин

Hello, I have kamailio in local network behind NAT. Kamailio have one interface eth1 with ip 10.130.0.23 UserAgent - (internet) - (178.0.0.169)router_with_NAT - (LAN) - kamailio+rtpproxy - pbx I made two listeners: listen=udp:10.130.0.23:5070 #to pbx in lan (I don`t want to put via header with 178.0.0.169) listen=udp:10.130.0.23:5060 advertise 178.0.0.169:5060 #to internet (UserAgent need to get via header with 178.0.0.169 ) mhomed = 1 # I think It not useful for my case UserAgent send Register and Invite to 178.0.0.169:5060, Kamailio get it on udp:10.130.0.23:5060 Kamailio Relay it to udp:10.130.0.23:5070 My pbx get requests from udp:10.130.0.23:5070 with via header 10.130.0.23 Responses to UserAgent go from udp:10.130.0.23:5060 with via 178.0.0.169 In RELAY block: if ($Ru == «udp:10.130.0.23:5070») { #set_send_socket(«udp:10.130.0.23:5060»); $fs = «udp:10.130.0.23:5060»; #force_send_socket(«udp:10.130.0.23:5060»); } else { #set_send_socket(«udp:10.130.0.23:5070»); $fs = «udp:10.130.0.23:5070»; } #$fs = «udp:10.130.0.23:5070»; if (!t_relay()) { sl_reply_error(); } route[REGISTRAR] { if ($rd == "mydomain.ru" || $rd == "mydomain2.ru") { set_send_socket("udp:10.130.0.23:5070"); } } It works for requests from UserAgent to PBX. But INVITE from my pbx to UserAgent goes only through udp:10.130.0.23:5070 , Kamailio realy it to internet from udp:10.130.0.23:5070 I get error in syslog: (pv [pv_branch.c:62]: pv_get_branchx_helper(): error accessing branch [0]) I tried set_send_socket(«udp:10.130.0.23:5070»); and $fs = «udp:10.130.0.23:5070»; Why I can`t relay outgoing requests to internet through another socket? -- Best regards, Alex

2 years, 2 months

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[REGISTRAR] Registered. How to match $sut to usrloc received ?

by Richard Edmands

Hey Kamailio Users, So the scenario is we get the registers. Load them, slightly oddly as we also fill in the received section, not something I usually do. Then when they send an invite, the To and From headers are real phone numbers and for some reason the contact header isn't their username. So to compensate for that I'm trying to match their connection ($sut) using registered, but after a couple of hours I haven't been able to get it to match up. Not sure what I'm missing. ul.dump gets me Received: sip:sipclientpublicip:2353;transport=udp so we know that part is loading $sut correctly into the user location data. modparam("usrloc", "matching_mode", 2) modparam("registrar", "xavp_rcd", "ulrcd") modparam("registrar", "received_avp", "$avp(i:42)") $xavp(regcfg=>match_received) = $sut; if (registered("location","$sut", 2, 1)) { After that block of code $sut is sip: sipclientpublicip:2353;transport=udp which is what's in usrloc for received, but it's not match. There's probably a much smarter way of doing this that I've completely missed, if so then that'd be excellent. But if I'm on the right path somewhat then it would be great to get this working.

2 years, 2 months

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[UAC] How to get the outbound registration status of uac.reg_active?

by Denys Pozniak

Hello! I have several Kamailio sharing a common uacreg table and use the RPC interface to enable and disable outgoing registration on nodes depending on the cluster status: *kamcmd uac.reg_active 0kamcmd uac.reg_active 1* https://kamailio.org/docs/modules/stable/modules/uac.html#uac.r.uac.reg_act… How can I get the current status of uac outbound registration (to get 0 or 1)? -- BR, Denys Pozniak

2 years, 2 months

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Re: [SR-Users] TLS connection failure. Not sure if system behavior is correct.

by Ahmed Fouad

Hi, You are currently using TLSv1/SSLv3 ECDHE-RSA-AES256-GCM-SHA384 256 Teams stopped TLSv1/ TLSv1.1 and only allowing TLS1.2+ So change your TLS settings to enforce it only method = TLSv1.2+ after restarting kamailio your teams connections should be good.

2 years, 2 months

1
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ACC CDR accounting

by Oleg Podguyko

HI I’m trying to get CDR function. I would like to have one CDR for whole call at the syslog. I use: …… loadmodule "db_text.so" loadmodule "dialog.so" loadmodule "acc.so" …….. # -----Dialog module ----- modparam("dialog", "db_mode", 0) # -----ACCounting module ----- modparam("acc", "early_media", 0) modparam("acc", "report_cancels", 1) modparam("acc", "log_facility", "LOG_LOCAL1") modparam("acc", "detect_direction", 0) modparam("acc", "log_level", ACC_LOG_LEVEL) modparam("acc", "log_flag", FLAG_ACC_LOG) modparam("acc", "log_missed_flag", FLAG_ACC_LOG_MISSED) modparam("acc", "failed_transaction_flag", FLAG_ACC_LOG_FAILED) modparam("acc", "log_extra", "ip_src=$si;ip_dst=$avp(ip_dst);sip_from=$fU;sip_to=$tU;ruri_user=$rU;spx_rs=$avp(spx_http_rs)") ## TODO modparam("acc", "cdr_enable", 1) modparam("acc", "cdr_enable_log", 1) modparam("acc", "cdr_facility", "LOG_LOCAL1") request_route { if (is_method("INVITE")) { setflag(FLAG_ACC_LOG); setflag(FLAG_ACC_LOG_MISSED); setflag(FLAG_ACC_LOG_FAILED); } At the kamailio.log I see only transaction for INVITE, but I wait for some CDR for all dialog. Whats wrong in my config? -- -- Олег Подгуйко

2 years, 2 months

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Cannot load usrloc table at startup using DMQ_USRLOC sync feature

by Benedito Marques

I have an scenario with two kamailios, and no databases. At the first, I authenticate one subscriber successfully and save in memory usrloc: /etc/kamailio # kamailio --version version: kamailio 5.4.4 (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_BLACKLIST, 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 on 16:13:18 Feb 15 2021 with gcc 9.3.0 /etc/kamailio # /etc/kamailio # --- /etc/kamailio # cat /etc/os-release NAME="Alpine Linux" ID=alpine VERSION_ID=3.12.3 PRETTY_NAME="Alpine Linux v3.12" HOME_URL="https://alpinelinux.org/" BUG_REPORT_URL="https://bugs.alpinelinux.org/" --- # Handle SIP registrations route[REGISTRAR] { if (!is_method("REGISTER")) return; if (!save("location","0x05")) { sl_reply_error(); } exit; } -- /etc/kamailio # kamcmd ul.dump { Domains: { Domain: { Domain: location Size: 1024 AoRs: { Info: { AoR: 1014(a)my.domain HashID: -1989102610 Contacts: { Contact: { Address: sip:1014@10.0.0.2:7071;ob Expires: 299 Q: -1.000000 Call-ID: d5cf6d5f293647a59ba64ba996a793b0 CSeq: 43689 User-Agent: MicroSIP/3.20.7 Received: sip: 10.0.0.2:7071 Path: [not set] State: CS_NEW Flags: 1 CFlags: 0 Socket: [not set] Methods: 8159 Ruid: uloc-62d95ca2-3c-1 Instance: [not set] Reg-Id: 0 Server-Id: 0 Tcpconn-Id: -1 Keepalive: 0 Last-Keepalive: 1658412252 KA-Roundtrip: 0 Last-Modified: 1658412252 } } } } Stats: { Records: 1 Max-Slots: 1 } } } } /etc/kamailio # *The problem is*: when I start the second kamailio server, the current usrloc users are not replicated to it, so in the second server the usrloc is empty: /etc/kamailio # kamcmd ul.dump { Domains: { Domain: { Domain: location Size: 1024 AoRs: { } Stats: { Records: 0 Max-Slots: 0 } } } } *The modules loaded:* - dmq.so - dmq_usrloc.so *Params (sync enabled):* modparam("dmq_usrloc", "enable", 1) modparam("dmq_usrloc", "sync", 1) modparam("dmq_usrloc", "usrloc_domain", "location") modparam("dmq_usrloc", "usrloc_delete", 1) modparam("dmq_usrloc", "replicate_socket_info", 0) *Obs*: The DMQ replication is working perfectly. So, if I register another user in server 1, server 2 receives the KDMQ message and the new user appears on usrloc. The real problem is to load pre registered users at startup. *Log messages from second server at startup:* 0(51) DEBUG: <core> [core/sr_module.c:873]: init_mod(): dmq_usrloc 0(51) INFO: dmq_usrloc [dmq_usrloc.c:81]: mod_init(): dmq usrloc replication mode = 1 0(51) DEBUG: <core> [core/sr_module.c:652]: find_mod_export_record(): found export of <ul_bind_usrloc> in module usrloc [/usr/lib/kamailio/modules/usrloc.so] 0(51) DEBUG: <core> [core/sr_module.c:652]: find_mod_export_record(): found export of <bind_dmq> in module dmq [/usr/lib/kamailio/modules/dmq.so] 0(51) DEBUG: dmq_usrloc [usrloc_sync.c:286]: usrloc_dmq_initialize(): loaded dmq api 0(51) DEBUG: dmq_usrloc [usrloc_sync.c:299]: usrloc_dmq_initialize(): dmq peer registered 0(51) DEBUG: dmq_usrloc [dmq_usrloc.c:111]: mod_init(): dmq_usrloc initialized 0(51) DEBUG: <core> [core/sr_module.c:804]: init_mod_child(): idx 0 rank -127: dmq_usrloc [main] 0(51) DEBUG: dmq_usrloc [dmq_usrloc.c:127]: child_init(): child_init PROC_INIT 7(62) DEBUG: dmq_usrloc [usrloc_sync.c:552]: usrloc_dmq_request_sync(): requesting sync from dmq peers 0(51) DEBUG: <core> [core/sr_module.c:804]: init_mod_child(): idx 0 rank 0: dialog [main] 7(62) DEBUG: dmq_usrloc [usrloc_sync.c:568]: usrloc_dmq_request_sync(): sending serialized data {"action":3} 7(62) DEBUG: dmq_usrloc [usrloc_sync.c:317]: usrloc_dmq_send(): sending dmq broadcast... 7(62) DEBUG: dmq [dmq_funcs.c:166]: bcast_dmq_message1(): skipping node sip:10.0.2.38:5090 7(62) DEBUG: dmq [dmq_funcs.c:166]: bcast_dmq_message1(): skipping node sip:10.0.2.30:5090 7(62) DEBUG: dmq [dmq_funcs.c:166]: bcast_dmq_message1(): skipping node sip:0.0.0.0:5090 *List nodes at SERVER 1:* /etc/kamailio # kamcmd dmq.list_nodes { host: 10.0.1.36 port: 5090 resolved_ip: 10.0.1.36 status: active last_notification: 0 local: 0 } { host: 10.0.1.35 port: 5090 resolved_ip: 10.0.1.35 status: active last_notification: 0 local: 0 } { host: 0.0.0.0 port: 5090 resolved_ip: 0.0.0.0 status: active last_notification: 0 local: 1 } *List nodes at SERVER 2:* /etc/kamailio # kamcmd dmq.list_nodes { host: 10.0.1.35 port: 5090 resolved_ip: 10.0.1.35 status: active last_notification: 0 local: 0 } { host: 10.0.1.36 port: 5090 resolved_ip: 10.0.1.36 status: active last_notification: 0 local: 0 } { host: 0.0.0.0 port: 5090 resolved_ip: 0.0.0.0 status: active last_notification: 0 local: 1 } -- Att, Benedito Marques

2 years, 2 months

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Job oportunitie

by Jaime Gajardo

Hello, We are seeking Engineers with some experience in open source VoIP solutions like Kamilio, Asterisk, etc. The opportunity is remote in Latin america. We are a Chilean company dedicated to giving different solutions for customer service on cloud. is you are interested please send us an email to alejandro(a)elipse.ai Thank you!!

2 years, 2 months

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db.c can't find dp_postgressql

by Phillip Thern

so, I don't actually know if my previous message found you, or if you replied, since my syslog suddenly grew to 26 GB and my server couldn't receive any mail because of it... Description I loaded module "db_postgres.so". I expected kamilio to start. It didn't Troubleshooting Reproduction config file |#!KAMAILIO #!define WITH_POSTGRESSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_PRESENCE #!define WITH_MSGREBUILD #!define WITH_NAT #!define WITH_NATSIPPING . . . #!ifdef WITH_POSTGRESSQL loadmodule "db_postgres.so" #!endif | Log Messages |Jul 22 10:31:16 opensips /usr/sbin/kamailio[8975]: ERROR: <core> [db.c:204]: db_bind_mod(): Module db_postgresql not found. Missing loadmodule? Jul 22 10:31:16 opensips /usr/sbin/kamailio[8975]: ERROR: usrloc [usrloc_mod.c:361]: mod_init(): failed to bind database module Jul 22 10:31:16 opensips /usr/sbin/kamailio[8975]: ERROR: <core> [core/sr_module.c:942]: init_mod(): Error while initializing module usrloc (/usr/lib/x86_64-linux-gnu/kamailio/modules/usrloc.so) | output from ls of modules directory: |acc.so call_obj.so dialog.so ipops.so msrp.so presence_conference.so qos.so sipcapture.so statsd.so uid_avp_db.so acc_diameter.so carrierroute.so dialplan.so jsonrpcs.so mtree.so presence_dialoginfo.so ratelimit.so sipdump.so stun.so uid_domain.so alias_db.so cfg_db.so dispatcher.so keepalive.so nat_traversal.so presence_mwi.so regex.so sipjson.so tcpops.so uid_gflags.so app_jsdt.so cfg_rpc.so diversion.so kemix.so nathelper.so presence_profile.so registrar.so sipt.so textops.so uid_uri_db.so app_sqlang.so cfgt.so dmq.so kex.so nosip.so presence_reginfo.so rls.so siptrace.so textopsx.so uri_db.so async.so cfgutils.so dmq_usrloc.so lcr.so outbound.so presence_xml.so rr.so siputils.so timer.so userblocklist.so auth.so corex.so domain.so log_custom.so p_usrloc.so pua.so rtimer.so sl.so tm.so usrloc.so auth_db.so counters.so domainpolicy.so lrkproxy.so path.so pua_bla.so rtjson.so sms.so tmrec.so xcap_client.so auth_diameter.so ctl.so drouting.so mangler.so pdb.so pua_dialoginfo.so rtpengine.so smsops.so tmx.so xcap_server.so auth_xkeys.so db2_ops.so enum.so matrix.so pdt.so pua_reginfo.so rtpproxy.so speeddial.so topoh.so xhttp.so avp.so db_cluster.so evrexec.so maxfwd.so permissions.so pua_rpc.so sanity.so sqlops.so topos.so xhttp_prom.so avpops.so db_flatstore.so exec.so mediaproxy.so pike.so pua_usrloc.so sca.so ss7ops.so tsilo.so xhttp_rpc.so benchmark.so db_postgres.so group.so mohqueue.so pipelimit.so pua_xmpp.so sdpops.so sst.so uac.so xlog.so blst.so db_text.so htable.so mqueue.so prefix_route.so pv.so seas.so statistics.so uac_redirect.so xprint.so call_control.so debugger.so imc.so msilo.so presence.so pv_headers.so secfilter.so statsc.so uid_auth_db.so | Possible Solutions tried renaming db_postgres.so to db_postgresql.so, but that didn't work, so I reversed it Additional Information * *Kamailio Version* - output of |kamailio -v| |version: kamailio 5.5.4 (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 10.2.1 | * *Operating System*: |Distributor ID: Debian Description: Debian GNU/Linux 11 (bullseye) Release: 11 Codename: bullseye Linux opensips 5.10.0-16-amd64 #1 SMP Debian 5.10.127-1 (2022-06-30) x86_64 GNU/Linux |

2 years, 2 months

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uac replace PRACK trouble

by Алексей Якимкин

Hello, pbx -> kamailio -> provider kamailio registers (uacreg) on provider. When I call through my provider I need to change headers To From and PAI. I tried with uac_replace_from and uac_replace_to. $tU for example 0987656789 if($fU=~"1234567") { uac_replace_from("sip:provider_login@provider.com"); uac_replace_to("sip:$tU@provider.com"); } My provider answers on Invite with 183 progress, so pbx answers with PRACK. Kamailio change To and from like this: To: <sip:0987656789@provider.comsip:0987656789@provider.com>;tag=..... To: <sip:0987656789@provider.comsip:0987656789@provider.com> From: <sip:provider_login@provider.comsip:provider_login@provider.com >;tag=..... From: <sip:provider_login@provider.comsip:provider_login@provider.com> Is it a bug? I tried to disable this modification for PRACK u REQINIT ($csb != "2 PRACK") It haven't worked for me. I solved this issue with this code. remove_hf("From"); if ($ft == $null) { insert_hf("From: <sip:provider_login@provider.com>\r\n", "To"); } else { insert_hf("From: <sip:provider_login@provider.com>;tag=$ft\r\n", "To"); } version: kamailio 5.7.0-dev0 (x86_64/linux) Ubuntu 18 -- Regards, Alex

2 years, 2 months

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Module db_postgresql not found

by Phillip Thern

Description I loaded module "db_postgres.so". I expected kamilio to start. It didn't Troubleshooting Reproduction config file |#!KAMAILIO #!define WITH_POSTGRESSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_PRESENCE #!define WITH_MSGREBUILD #!define WITH_NAT #!define WITH_NATSIPPING . . . #!ifdef WITH_POSTGRESSQL loadmodule "db_postgres.so" #!endif | Log Messages |Jul 22 10:31:16 opensips /usr/sbin/kamailio[8975]: ERROR: <core> [db.c:204]: db_bind_mod(): Module db_postgresql not found. Missing loadmodule? Jul 22 10:31:16 opensips /usr/sbin/kamailio[8975]: ERROR: usrloc [usrloc_mod.c:361]: mod_init(): failed to bind database module Jul 22 10:31:16 opensips /usr/sbin/kamailio[8975]: ERROR: <core> [core/sr_module.c:942]: init_mod(): Error while initializing module usrloc (/usr/lib/x86_64-linux-gnu/kamailio/modules/usrloc.so) | output from ls of modules directory: |acc.so call_obj.so dialog.so ipops.so msrp.so presence_conference.so qos.so sipcapture.so statsd.so uid_avp_db.so acc_diameter.so carrierroute.so dialplan.so jsonrpcs.so mtree.so presence_dialoginfo.so ratelimit.so sipdump.so stun.so uid_domain.so alias_db.so cfg_db.so dispatcher.so keepalive.so nat_traversal.so presence_mwi.so regex.so sipjson.so tcpops.so uid_gflags.so app_jsdt.so cfg_rpc.so diversion.so kemix.so nathelper.so presence_profile.so registrar.so sipt.so textops.so uid_uri_db.so app_sqlang.so cfgt.so dmq.so kex.so nosip.so presence_reginfo.so rls.so siptrace.so textopsx.so uri_db.so async.so cfgutils.so dmq_usrloc.so lcr.so outbound.so presence_xml.so rr.so siputils.so timer.so userblocklist.so auth.so corex.so domain.so log_custom.so p_usrloc.so pua.so rtimer.so sl.so tm.so usrloc.so auth_db.so counters.so domainpolicy.so lrkproxy.so path.so pua_bla.so rtjson.so sms.so tmrec.so xcap_client.so auth_diameter.so ctl.so drouting.so mangler.so pdb.so pua_dialoginfo.so rtpengine.so smsops.so tmx.so xcap_server.so auth_xkeys.so db2_ops.so enum.so matrix.so pdt.so pua_reginfo.so rtpproxy.so speeddial.so topoh.so xhttp.so avp.so db_cluster.so evrexec.so maxfwd.so permissions.so pua_rpc.so sanity.so sqlops.so topos.so xhttp_prom.so avpops.so db_flatstore.so exec.so mediaproxy.so pike.so pua_usrloc.so sca.so ss7ops.so tsilo.so xhttp_rpc.so benchmark.so db_postgres.so group.so mohqueue.so pipelimit.so pua_xmpp.so sdpops.so sst.so uac.so xlog.so blst.so db_text.so htable.so mqueue.so prefix_route.so pv.so seas.so statistics.so uac_redirect.so xprint.so call_control.so debugger.so imc.so msilo.so presence.so pv_headers.so secfilter.so statsc.so uid_auth_db.so | Possible Solutions tried renaming db_postgres.so to db_postgresql.so, but that didn't work, so I reversed it Additional Information * *Kamailio Version* - output of |kamailio -v| |version: kamailio 5.5.4 (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 10.2.1 | * *Operating System*: |Distributor ID: Debian Description: Debian GNU/Linux 11 (bullseye) Release: 11 Codename: bullseye Linux opensips 5.10.0-16-amd64 #1 SMP Debian 5.10.127-1 (2022-06-30) x86_64 GNU/Linux |

2 years, 2 months

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incomplete values and Wrong Counting for $hfl and $hflc

by Patrick Karton

the counting and values provided by $hfl and $hflc are wrong and incomplete from my understanding. looking at the source code these pseudo vars count the body values of Via, Record-Route and Route headers. in kamalio 5.6.1 with Record-Route: <sip:1.1.1.1;lr=on;nat=yes> Record-Route: <sip:2.2.2.2:5060;lr;transport=udp>,<sip:3.3.3.3:22506;lr;transport=udp> i get $hflc(Record-Route) == 2 instead of 3 and $(hfl(Record-Route)[0]) == 1.1.1.1 $(hfl(Record-Route)[1]) == 2.2.2.2 $(hfl(Record-Route)[3]) does not exists.

2 years, 2 months

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Using Kamailio with Teams

by Javier González Muñoz

Hi, I'm using kamailio with Teams like SBC and asterisk. Works perfectly less one thing. When I make a call asterisk to Teams, in my Teams app I see 3 popup notifications, and 3 calls in my history. If I call Teams to Asterisk, perfect.. When I make an asterisk to Teams call, I get the trace from Asterisk and I can see three 180, I don't know if this is the problem, but are there any way to fix it? Limit to 1 notification? Thanks

2 years, 2 months

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iterate trough all Record-Route values via @hf_value.Record_Route

by Patrick Karton

Hello All, i would to iterate through all ips in Record-Route through @hf_value.Record_Route. if i put static index like @hf_value.Record_Route[0] it works. but $var(i) = 0; @hf_value.Record_Route[$var(i)] does not work. is there a way to get the count of all headers values through @hf_value.Record_Route For example i would like to iterate through all Record-Route Values if i receive it like this Record-Route: <sip:1.1.1.1;lr=on;nat=yes> Record-Route: <sip:2.2.2.2:5060;lr>,<sip:3.3.3.3:22506;lr>

2 years, 2 months

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siremis installation

by Fernando Piana

hello, My name is Fernando Piana from Colombia. I am writing to you because I am testing Kamailio together with the Siremis graphical environment. I was visiting the siremis.org website and also the siremis wiki, but I couldn't find the administration manual. >From this medium I want to ask you if you can send me a copy of the Siremis administration manual. I await a response. Thank you very much. -- Fernando Piana Ing. Comunicaciones y Bases de Datos Cel: +57 311 641 7558

2 years, 2 months

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Using usrloc db_insert_update in a dmq replicated environment with shared db ?

by Matteo Brancaleoni

Hi, I was wondering if it is safe from a cache consistency pow to enable usrloc db_insert_update in a dmq replicated environment, where all registrars share the same db. I'm already using the filter option to avoid loading/deletion by server id, but there's no similar option for writing. I know that it is possible to use db mode 4 on all registrars (which get synced by dmq) except one, which will be responsible for writing, but this creates a little unbalanced setup in addition to performing writes only from one proxy. Any hint? Maybe is time to check how to filter writes, too :) ? Thanks, Matteo

2 years, 2 months

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How to externally trigger htable cache flushing?

by Benoit Panizzon

Hi List We cache stuff like call forwarding information in a htable cache to minimizie SQL lookups and have it available via DMQ on all instances. But of course, if the call forwarding information for a destination changes in the database, we would like to flush either the whole cache, or even better, that single entry. What is the best way to do this? Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

2 years, 2 months

3
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dialog timeout_avp no longer works?

by Alex Balashov

Hi, I discovered recently that, on Kamailio 5.5.4 at least, the dialog ’timeout_avp’ no longer appears to have effect. Here are my (seemingly relevant) dialog parameters: modparam("dialog", "enable_stats", 0) modparam("dialog", "hash_size", 4096) modparam("dialog", "rr_param", "dlgcor") # DLG "correlator" modparam("dialog", "dlg_flag", 2) modparam("dialog", "default_timeout", 28800) # 8 hours modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "table_name", "dialog") modparam("dialog", "send_bye", 1) modparam("dialog", "wait_ack", 0) modparam(“dialog”, “timeout_avp”, “$avp(dlg_timeout)”) The flow is roughly like this: request_route { ... t_check_trans(); if(is_method(“INVITE”)) { ... dlg_manage(); $avp(dlg_timeout) = 10; ... if(!t_relay()) sl_reply_error(); return; } I was able to resolve the problem by switching to dlg_set_timeout() instead, but I wonder if there has been a regression. Last version I have on which this positively works is 4.4.7, though, so not sure when it might have broken. One nuance: $avp(dlg_timeout) is not initialised prior to calling dlg_manage(), and the value is only set if certain conditions are true. Not sure if some stricter check appeared in later releases which could cause this to be a problem. Thank you! — Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 years, 2 months

3
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Nathelper sip ping and Path support ?

by matteo brancaleoni

Hello, I'm trying to setup the following scenario with sip ping: UAC -> Kamailio Edge -> Kamailio Registrar Edge proxy is a multihomed proxy with public/private ips and the registrar is a kamailio on a private network. The edge acts as a nat proxy and loadbalancer. Edge proxy inserts a Path header (and contact alias) which get correctly saved on registrar usrloc. Registration works as expected. Now, since I need to keep nat open towards the uacs and expire them when they're not reachable anymore, I'm trying to send sip pings from the registrar via the edge proxy. I'm using nathelper sipping in the registrar. What I see is that the registrar sends out a sip options putting in the Route header the Path as expected, but the sip packet gets sent directly to the UAC contact and not via the edge, as the Route header specifies. After looking into the nathelper code, it seems that the sip packet is sent via udp directly without any specific routing. So the question: it should work what I'm trying to achieve? Maybe not? Probably I can work around it setting the ;received in the contact on registrar (it will set to the edge proxy) and since the contact is aliased on the edge when the edge receives an option from the registrar it can process the alias and send it to the correct endpoint, but seems a workaround? Or using the keepalive module maybe, even if that means setting specific timers and manually do what the nathelper module already does? Any hint? Thanks, Matteo

2 years, 2 months

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Does pure outbound RFC5626 work without edge proxies?

by Richard Chan

Hi sr-users, Does pure RFC5626 outbound work without an edge proxy->internal proxy design? By "pure" I mean no nathelper to rewrite addresses. I tried setting up the following: Configuration: use standard kamailio configuration without WITH_NAT (Ignore RTP-rtpengine/rtpproxy for the moment as I am just testing the control plane) 1. UA1 UA2 each behind NAT, with bogus Contact: headers 2. kamailio loaded with rr/tm (no outbound/path because I am using single proxy/registrar — without edge proxies) 3. no nathelper: no Contact rewriting, set_contact_alias etc 4. Both UA1 and UA2 establish persistent TLS to kamailio 5. INVITE/200 OK/ACK does work. 6. Contact headers show the NAT bogus addresses (as expected). UA1/UA2 are configured to be "dumb"(i.e. don't self-rewrite Contact:/SDP based on rport/received). Problem: 7. In-dialog requests don't work as kamailio tries to route to the bogus addresses per the Contact header. I.e. kamailio doesn't seem to resolve the UAs(bogus Contact addresses) to the persistent TLS connections. Now in the Route header I don't see any flow-token so probably kamailio is doing "normal" routing instead of RFC5626 routing (i.e. because it has no flow-token to match with an existing flow). This leads me to ask: for pure RFC5626 to work (no nathelper stuff at all) is it a must that there is a "edge-proxy--[1]-->internal-proxy" layout? ...and...the flow-tokens in [1] are what make it possible to skip bogus Contact: header and route to an existing flow? Regards Shih-Ping Chan

2 years, 2 months

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KEMI: getting ksr_request_route not found or is not callable after t_relay()

by Nigel Daniels

``` def ksr_route_relay(self, msg): if KSR.is_method_in("IBSU"): if KSR.tm.t_is_set("branch_route") < 0: KSR.tm.t_on_branch("ksr_branch_manage") if KSR.is_method_in("ISU"): if KSR.tm.t_is_set("onreply_route") < 0: KSR.tm.t_on_reply("ksr_onreply_manage") if KSR.is_INVITE(): KSR.info("try and relay the traffic\n") if KSR.tm.t_is_set("ksr_failure_route") < 0: KSR.tm.t_on_failure("ksr_failure_manage") t_relay_result = KSR.tm.t_relay(["10.0.0.6", "5060"]) KSR.info(f't_relay: {t_relay_result}\n') KSR.info(f'MSG: {msg}') if KSR.tm.t_is_set("ksr_failure_route_one"): KSR.tm.t_on_reply("ksr_on_reply_route_one") ``` ERROR: app_python3 [python_exec.c:91]: apy_exec(): ksr_request_route not found or is not callable Again this only happens after t_replay is called. Any help understanding why this happens would be appreciated. -- - Nigel

2 years, 2 months

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uacreg get error on 2nd 401: error uac_reg_tm_callback(): authentication failed

by Алексей Якимкин

Hello, I try to register on provider`s pbx version: kamailio 5.7.0-dev0 (x86_64/linux) modparam("uac", "reg_db_url", "mysql://kamailio:123@localhost/kamailio") modparam("uac", "reg_contact_addr", "1.1.1.1:5060") modparam("uac", "reg_retry_interval", 120); # for failed registrations modparam("uac", "reg_keep_callid", 1) modparam("uac","restore_mode","auto") uacreg table: +----+---------+------------+-----------+----------------+-----------------+-------+----------------+------------------+----------+---------------------+---------+-------+-----------+--------------+------------------+ | id | l_uuid | l_username | l_domain | r_username | r_domain | realm | auth_username | auth_password | auth_ha1 | auth_proxy | expires | flags | reg_delay | contact_addr | socket | +----+---------+------------+-----------+----------------+-----------------+-------+----------------+------------------+----------+---------------------+---------+-------+-----------+--------------+------------------+ | 1 | 1234567 | pbx | 127.0.0.1 | login | ip.provider.ru | | login | password| | sip:...com | 120 | 0 | 0 | | 10.130.0.23:5060 | I see this two 401 replies [image: изображение.png] 1st REGISTER Via: SIP/2.0/UDP 1.1.1.1:5060 ;branch=z9hG4bK589d.8183f2e6000000000000000000000000.0 To: <sip:login@ip.provider.com> From: <sip:login@ip.provider.com >;tag=f9cf84029bbf24b61836c3d9fef7b7ae-78a2ffe1 CSeq: 10 REGISTER Call-ID: 6818f1b436befce1-21980(a)10.130.0.23 Max-Forwards: 70 Content-Length: 0 User-Agent: server Contact: <sip:1234567@1.1.1.1:5060> Expires: 120 1st 401 WWW-Authenticate: Digest algorithm=MD5,realm=".......org",nonce=" *56806a6f32c49a3afea043c2fa42381f*",domain="sip:......org",qop="auth", *stale=FALSE* 2nd REGISTER Authorization: Digest username="login", realm=".....", nonce=" *56806a6f32c49a3afea043c2fa42381f*", uri="sip:......com", qop=auth, nc=00000001, cnonce="357420659", response="6e300c9f7f4d81095f29b5408f8d5191", algorithm=MD5 2nd 401 WWW-Authenticate: Digest algorithm=MD5,realm=".....org",nonce=" *3b9586f6d31a76e509eb204d8a030415*",domain="sip:.....org",qop="auth", *stale=TRUE* I think kamailio need to sent 3nd REGISTER with a new nonce, because stale=true. But it interrupts register with error: Jul 14 18:41:00 opensips kamailio[21936]: 10(21949) DEBUG: {2 11 REGISTER 6818f1b436befcf7-21980(a)10.130.0.23} uac [uac_reg.c:852]: uac_reg_tm_callback(): completed with status 401 [uuid: 1234567] Jul 14 18:41:00 opensips kamailio[21936]: 10(21949) ERROR: {2 11 REGISTER 6818f1b436befcf7-21980(a)10.130.0.23} uac [uac_reg.c:933]: uac_reg_tm_callback(): authentication failed for <1234567> Can I get kamailio to send REGISTER after the second 401-reply? Can I intercept this REGISTER and 401 in kamailio.conf? Is there any examples? I know about this instruction https://kamailio.org/docs/modules/5.7.x/modules/uac.html :) Itried in request_route xlog("L_NOTICE", "uac 114411 1 remote fU $fU user [$rU] on [$rd] via [$du] $Ri \n"); if (uac_reg_request_to("$fU", 0)) { xlog("L_NOTICE", "uac 114411 Found remote user [$rU] on [$rd] via [$du]\n"); t_on_failure("REMOTE_AUTH"); #t_relay(); } And failure_route failure_route[REMOTE_AUTH] { if ($T_reply_code == 401 or $T_reply_code == 407) { xlog("L_NOTICE", "uac 114411 Remote asked for authentication\n"); uac_auth(); } } but I haven't got logs on register request, 401 reply. -- Best regards, Alex

2 years, 2 months

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usage of {s.urldecode.param} to convert special chars (why is + removed?)

by Benoit Panizzon

Hi List Service Numbers, like to activate CFW probably often start with * and end with a # # is a special character that needs to be urlencoded in an SIP URI. So best would be to use a generic kamailio urldecode transormation. I guess {s.urldecode.param} is what I was looking for. So when I get a call to the 'enable CFW' service I get $rU => "*21+419999999%23" So I want to extract the destination number and remove the urlencoded # sign. So I do $var(cfw-dest) = $var(cfw-dest) = $rU; $var(cfw-dest) = $(var(cfw-dest){s.substr,3,0}); # "+419999999%23" $var(cfw-dest) = $(var(cfw-dest){s.urldecode.param}); But now I end up with 419999999# the + sigh disappeared. Is this expected behavior? How would this be done the correct way? Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

2 years, 2 months

2
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Releasing v5.6.1

by Daniel-Constantin Mierla

Hello, I am considering to release Kamailio v5.6.1 (out of branch 5.6) on Wednesday, July 6, 2022 -- in about 1 week and a half from now. If anyone is aware of issues not yet on the bug tracker, report them there asap in order to have a better chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Online: June 20-23, 2022 * https://www.asipto.com/sw/kamailio-advanced-training-online/

2 years, 2 months

5
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Need Suggestion/Idea to set call limit on 3 kamailios.

by Waqar 40

Please Read the Scenario in full and then guide me. In my scenario, I have three Kamailio (KamA, KamB, KamC) running in production. They do load balancing for the asterisks which are about 9. Kamailios randomly receive calls from the SIP trunk. I have set the limit of 750 calls on each kamailios. I want to provide only 1000 Calls to my client but they are enjoying calls more than this since each kamailio can receive 750 calls (I have set this limit for redundancy purpose). And I cannot set the limit like 750 on KamA and the rest on KamB and KamC for redundancy purposes. If one kamailio goes down then they send calls on the other kamailios so that no call drops. So, I want to know if there is a module or a way that my redundancy is not affected and I limit total calls on the three kamailios to not exceed 1000. Waiting for a prompt response. Let me know if you need any further information. Regards, Vicky

2 years, 2 months

3
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How to disable GRUU?

by Алексей Дырявко

Hello. kamailio -v version: kamailio 5.4.8 (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_BLACKLIST, 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 8.3.0 How can I disable GRUU support completely? Sip client Grandstream NT-XXX sends default registration request with "reg-id" and "instance" REGISTER sip:kamailio3.telecom.ru SIP/2.0 Via: SIP/2.0/UDP 172.17.0.252:5060;branch=z9hG4bK1645839794;rport From: "7777" sip:7777@kamailio3.telecom.ru;tag=53756161 To: sip:7777@kamailio3.telecom.ru Call-ID: 308071885-5060-1 CSeq: 2031 REGISTER Contact: sip:7777@172.17.0.252:5060 ;reg-id=1;+sip.instance="urn:uuid:00000000-0000-1000-8000-000B82566BBB" Authorization: Digest username="7777", realm="kamailio3.telecom.ru", nonce="Ysv842LL+7dABGNcyuYZhl1L2QTPgzFq", uri="sip:kamailio3.telecom.ru", response="0ef43a754a7422dd9d8ed36f91aa546 , algorithm=MD5 Max-Forwards: 70 User-Agent: Grandstream HT-502 V2.0A 1.0.9.1 chip V2.2 Supported: path Expires: 1800 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE Content-Length: 0 SIP/2.0 200 OK Via: SIP/2.0/UDP 172.17.0.252:5060 ;branch=z9hG4bK1645839794;rport=5060;received=172.17.0.252 From: "7777" sip:7777@kamailio3.telecom.ru;tag=53756161 To: sip:7777@kamailio3.telecom.ru ;tag=5bf75eb19869d3e257cd1fde2cbf92c4.7224507f Call-ID: 308071885-5060-1 CSeq: 2031 REGISTER Contact: sip:7777@172.17.0.252:5060;expires=1800;+sip.instance="urn:uuid:00000000-0000-1000-8000-000B82566BBB";reg-id=1, sip:07777@172.17.0.250:5060;expires=101, <sip:7777@172.1 0.250:5061>;expires=101, sip:7777@172.17.0.252:5062 ;expires=607;+sip.instance="urn:uuid:00000000-0000-1000-8000-000B82566BBB";reg-id=2 Server: kamailio (5.4.8 (x86_64/linux)) Content-Length: 0 REGISTER sip:kamailio3.telecom.ru SIP/2.0 Via: SIP/2.0/UDP 172.17.0.252:5062;branch=z9hG4bK212965390;rport From: "7777" sip:7777@kamailio3.telecom.ru;tag=1540711120 To: sip:7777@kamailio3.telecom.ru Call-ID: 1127299103-5062-1 CSeq: 2031 REGISTER Contact: sip:7777@172.17.0.252:5062 ;reg-id=2;+sip.instance="urn:uuid:00000000-0000-1000-8000-000B82566BBB" Authorization: Digest username="7777", realm="kamailio3.telecom.ru", nonce="Ysv862LL+78cGZE/vXJnQfemhCMiBilm", uri="sip:kamailio3.telecom.ru", response="6c3333159d41ae0fcaabfbbd8c5026e , algorithm=MD5 Max-Forwards: 70 User-Agent: Grandstream HT-502 V2.0A 1.0.9.1 chip V2.2 Supported: path Expires: 1800 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE Content-Length: 0 SIP/2.0 200 OK Via: SIP/2.0/UDP 172.17.0.252:5062 ;branch=z9hG4bK212965390;rport=5062;received=172.17.0.252 From: "7777" sip:7777@kamailio3.telecom.ru;tag=1540711120 To: sip:7777@kamailio3.telecom.ru ;tag=5bf75eb19869d3e257cd1fde2cbf92c4.451b5a53 Call-ID: 1127299103-5062-1 CSeq: 2031 REGISTER Contact: sip:7777@172.17.0.252:5060;expires=1792;+sip.instance="urn:uuid:00000000-0000-1000-8000-000B82566BBB";reg-id=1, sip:7777@172.17.0.250:5060;expires=93, <sip:7777@172.17 .250:5061>;expires=93, sip:7777@172.17.0.252:5062 ;expires=1800;+sip.instance="urn:uuid:00000000-0000-1000-8000-000B82566BBB";reg-id=2 Server: kamailio (5.4.8 (x86_64/linux)) Content-Length: 0 Later, when I use branching (with q=-1.0), the kamailio sends an Invite to only one Contact from 7777@172.17.0.252:5062 If I use another Sip client without "instance" , such as Linksys-PAP2T, branching works fine. GRUU is always disabled modparam("registrar", "gruu_enabled", 0)

2 years, 2 months

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(Off-topic) Vienna, meeting

by Henning Westerholt

(Off-topic) Hello, In about two weeks I will be a few days in Vienna (12h - 14th July). If you want to meet for a drink to chat about Kamailio and other topics, just contact me off-list. Thanks, and regards, Henning

2 years, 2 months

1
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Client behind nat without stun and kamailio behind NAT

by Алексей Якимкин

Hello, I hope somebody could help me. This is my scheme. User-agent is behind NAT1. Kamailio and pbx are behind NAT2 [Client ip-phone 192.168.89.213 without stun] - LAN1 - NAT1(46.0.0.30) - (internet) - (51.0.0.60)NAT2 - Local2 - (10.130.0.23:5060)kamailio( 10.130.0.23:5070) - pbx *Questions*: *1*. The VIA header (with 10.130.0.23 and 51.0.0.60) wasn't included in SIP packets. Why? For example 200OK reply. It came from pbx through kamailio. Which setting could break it? Session Initiation Protocol (200) Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 192.168.89.213:5060 ;rport=9570;received=46.0.0.30;branch=z9hG4bK2480172053 Record-Route: <sip:c172.19.19.111.8348.call.cgatepro;lr> Record-Route: <sip:172.19.19.111:5060;lr> Record-Route: <sip:10.130.0.23:5070 ;r2=on;lr;ftag=4268683942;nat=yes>,<sip:51.0.0.60;r2=on;lr;ftag=4268683942;nat=yes> From: "Aleksey" <sip:a.yakimkin@mail.domain.ru:5060>;tag=4268683942 To: <sip:2961@mail.domain.ru:5060 >;tag=A03E2397-404246-FA7543E4_jizmelr-582D Call-ID: 6_1903330087(a)192.168.89.213 [Generated Call-ID: 6_1903330087(a)192.168.89.213] CSeq: 2 INVITE Contact: <sip:signode-404246-FA7543E4_jizmelr-582D@172.19.19.111 ;alias=51.0.0.60~5060~1> Supported: 100rel,timer,replaces,histinfo,precondition Allow: INVITE,BYE,CANCEL,ACK,OPTIONS,INFO,MESSAGE,PRACK,UPDATE,REFER Session-Expires: 1800;refresher=uas Content-Type: application/sdp Content-Length: 1170 Message Body *2*. About Registrar, Path and $du. Phone set Register with headers: Via: SIP/2.0/UDP 192.168.89.213:5060;branch=z9hG4bK2691182696 From: "Aleksey" <sip:a.yakimkin@mail.domain.ru:5060>;tag=3926879477 To: "Aleksey" <sip:a.yakimkin@mail.domain.ru:5060> Contact: <sip:a.yakimkin@192.168.89.213:5060> Kamailio respond Via: SIP/2.0/UDP 192.168.89.213:5060 ;rport=9570;received=46.0.0.30;branch=z9hG4bK617463686 Path: <sip:10.130.0.23:5070 ;lr;received=46.0.0.30~9570~1;r2=on>,<sip:51.0.0.60;lr;received=46.0.0.30~9570~1;r2=on> From: "Aleksey" <sip:a.yakimkin@mail.domain.ru:5060>;tag=3926879477 To: "Aleksey" <sip:a.yakimkin@mail.domain.ru:5060>;tag=194ED16D Contact: <sip:a.yakimkin@192.168.89.213:5060>;expires=360 Contact: <sip:2447@192.168.89.221:5060>;expires=247 Contact: <sip:2447@192.168.9.16:5060>;expires=2116 I try to make a call from pbx to ip phone. pbx inserts in Invite header Route: <sip:10.130.0.23:5070 ;lr;received=46.0.0.30~9570~1;r2=on> But kamailio relayed Invite direct to 192.168.89.213. (There is network connectivity between ip-phone and kamailio through vpn). The code below helps me to solve my issue. I saw mail-list with similar trouble. But no setting could get kamailio to relay Invite to "Route-received" ip. $var(the_route) = $hdr(Route); $var(route0) = $(var(the_route){s.select,0,,}); $var(new_host) = $(var(route0){param.value,received}{s.select,0,~}); $var(new_port) = $(var(route0){param.value,received}{s.select,1,~}); if (!strempty($var(new_host)) && !strempty($var(new_port)) ) { $du = "sip:" + $var(new_host) + ":" + $var(new_port); } *Kamailio settings:* I have such listeners listen=udp:10.130.0.23:5070 # to local network listen=udp:10.130.0.23:5060 advertise 51.0.0.60:5060 # to internet #MODULE SETTING #--- # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) /* set the path to RPC fifo control file */ # modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo") /* set the path to RPC unix socket control file */ # modparam("jsonrpcs", "dgram_socket", "/var/run/kamailio/kamailio_rpc.sock") modparam("path", "use_received", 1) modparam("path", "enable_r2", 1) modparam("path", "received_format", 1) # ----- ctl params ----- /* set the path to RPC unix socket control file */ # modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) modparam("tm", "auto_inv_100_reason", "Trying") # ----- rr params ----- # set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 0) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 1) modparam("rr", "enable_double_rr", 2) modparam("rr", "force_send_socket", 1) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ # modparam("registrar", "max_contacts", 10) /* max value for expires of registrations */ modparam("registrar", "max_expires", 3600) /* set it to 1 to enable GRUU */ modparam("registrar", "gruu_enabled", 0) modparam("registrar", "use_path", 1) modparam("registrar", "path_use_received", 1) modparam("registrar", "path_mode", 0) #--- For register I use this code route[REGISTRAR] { ... add_path_received(); set_send_socket("udp:10.130.0.23:5070"); route(DISPATCH); ... } route[RELAY] { ... if ($Ru eq "sip:10.130.0.23:5070") { $fs = "udp:10.130.0.23:5060"; } else { $fs = "udp:10.130.0.23:5070"; } ... } route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; #send INVITE to IP:PROT from Route:...;received= $var(the_route) = $hdr(Route); $var(route0) = $(var(the_route){s.select,0,,}); $var(new_host) = $(var(route0){param.value,received}{s.select,0,~}); $var(new_port) = $(var(route0){param.value,received}{s.select,1,~}); if (!strempty($var(new_host)) && !strempty($var(new_port)) ) { $du = "sip:" + $var(new_host) + ":" + $var(new_port); } if (client_nat_test("3")) { if(nat_uac_test("18")) { if ($Ru == "sip:10.130.0.23:5070") { rtpproxy_manage("co", "51.0.0.60"); # fix_nated_sdp } else { rtpproxy_manage("co"); } if (is_method("REGISTER")) { #if ($Ru == "sip:10.130.0.23:5070") { # fix_nated_contact(); #} else { # set_contact_alias(); #} set_contact_alias(); } else { if(is_first_hop()) { set_contact_alias(); } else { add_contact_alias("51.0.0.60", "5060", "udp"); #fix_nated_contact(); } } } else { if ($Ru == "sip:10.130.0.23:5070") { rtpproxy_manage("cor", "51.0.0.60"); # fix_nated_sdp } else { rtpproxy_manage("cor"); } } } else { rtpproxy_manage("co"); } if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); #fix_contact(); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); #} else { #fix_contact(); #} } } #!endif return; } Thank you. -- Best regards, Alex

2 years, 3 months

2
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log any rpc command

by Patrick Karton

hello is there a way to log in kamailio any kamcmd entered ? Thanks.

2 years, 3 months

1
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Multiple rtpengine servers

by David Cunningham

Hello, We are looking to implement a system with one Kamailio server and multiple rtpengine servers. I see that the rtpengine module allows the server configuration to be stored in a database table with multiple entries, which is perfect. My question is - how does Kamailio distribute load between multiple rtpengine servers in the same set? If one of the rtpengine servers goes offline then will Kamailio note this and not send calls to that server until it comes back online? If not then presumably calls could be delayed by whatever the timeout for accessing a server is? Thank you very much, -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782

2 years, 3 months

3
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kamailio 5.6.1 minor release

by Patrick Karton

Hello Daniel, you said a couple weeks ago that you plan to release a minor release 5.6.1 for this week. its still on plan ? Thanks.

2 years, 3 months

2
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Kamailio v5.6.1 Released

by Daniel-Constantin Mierla

Hello, Kamailio SIP Server v5.6.1 stable release is out. This is a maintenance release of the latest stable branch, 5.6, that includes fixes since the release of v5.6.0. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.6.x. Deployments running previous v5.6.x versions are strongly recommended to be upgraded to v5.6.1. For more details about version 5.6.1 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2022/07/kamailio-v5-6-1-released/ RPM, Debian/Ubuntu packages will be available soon as well. Adding a short note that the dates of next Kamailio World Conference - Online have been set for September 7-8, 2022. Event website and more details will be published soon! 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

2 years, 3 months

1
0
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branch_route not called on DNS failover

by Patrick Karton

im using dns failover with tm module. i set parameter reparse_on_dns_failover to 0 so that branch_route is called . but when trying next peer of dns failover initial transaction branch_route is not called. do i missing something ? for the first peer branch_route is called and the peer timeout so kamailio tries next peer but branch_route is not called !. do i need again to set branch_route in failure_route ? Thanks for help.

2 years, 3 months

3
4
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Changing the From User in Failure Route

by Mack Hendricks

Hey All, I'm just trying to change the From user before I do a 301 Redirect within a Failure Route. Here's the code snippet: $fu = "+15555555555"; append_to_reply("Contact: sip:+1$var(redirect_did)@$si:$sp\r\n"); t_reply("302", "Redirect"); The From user is not changed and msg_apply_changes() isn't allowed. What am I doing wrong? Thanks

2 years, 3 months

2
1
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Kamailio 5.5 STIR SHAKEN private key buffer size error

by Maharaja Azhagiah

Hi, I am trying STIR/SHAKEN using libstirshaken in Kamailio 5.5. I used a self signed certificate as this is just a test in the local docker environment. However, when I try to add identity with private key (stirshaken_add_identity_with_key), I get "[error_code: 447] Buffer for key from file /tmp/cert/private.pem too short (2000 <= 3247)" I have tried using 2048 and 4096 size root@5907e44bd056:/tmp/cert# openssl rsa -in private.pem -text -noout | grep "Private-Key" RSA Private-Key: (4096 bit, 2 primes) Could you tell me what is wrong with the certificate? Kamailio version: root@5907e44bd056:/usr/local/kamailio/etc/kamailio# kamailio -v version: kamailio 5.5.4 (x86_64/linux) 469465 Error: 0(404) ERROR: {1 30587 INVITE NzIhM1-2YABveZZ1mPvs3m3tw8K7meSq} stirshaken [stirshaken_mod.c:761]: ki_stirshaken_add_identity_with_key(): Failed to load private key 0(404) DEBUG: {1 30587 INVITE NzIhM1-2YABveZZ1mPvs3m3tw8K7meSq} stirshaken [stirshaken_mod.c:117]: stirshaken_print_error_details(): failure details: 0(404) DEBUG: {1 30587 INVITE NzIhM1-2YABveZZ1mPvs3m3tw8K7meSq} stirshaken [stirshaken_mod.c:118]: stirshaken_print_error_details(): failure reason is: src/stir_shaken_ssl.c:2112: [error_code: 447] Buffer for key from file /tmp/cert/private.pem too short (2000 <= 3247) 0(404) DEBUG: {1 30587 INVITE NzIhM1-2YABveZZ1mPvs3m3tw8K7meSq} stirshaken [stirshaken_mod.c:119]: stirshaken_print_error_details(): failure error code is: 447 0(404) ERROR: {1 30587 INVITE NzIhM1-2YABveZZ1mPvs3m3tw8K7meSq} <script>: Failed Regards *Maharaja Azhagiah*

2 years, 3 months

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How to call RPC functions from script

by Marat Gareev

Hello! Is it possible to call RPC functions from script? In particular, I'm interested in the ability to add new groups to the dispatcher. https://kamailio.org/docs/modules/devel/modules/dispatcher.html#dispatcher.…

2 years, 3 months

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