- sr-users - kamailio.org

Re: [SR-Users] Callback for TM?s "fr_inv_timer" ?

by Antonio Couto

I will try this, thanks a lot for your help. regards. On Wed, Jun 15, 2022 at 5:43 PM Alex Balashov <abalashov(a)evaristesys.com> wrote: > Setting t_on_failure("FAILURE_ROUTE") before t_relay(), given a route > block failure_route[FAILURE_ROUTE], as you are doing, should work. > > -- > 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/ > > > -- > 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/ > > > On Wed, Jun 15, 2022 at 8:34 AM Antonio Couto <couto.a(a)gmail.com> wrote: > >> Thanks for your support, I am trying to do this but to be honest I don't >> know >> how to arm the failure route for this case of "fr_inv_timer". >> >> When I received reply 183 I use t_set_fr() function to set the new value >> for "fr_inv_timer", >> do far so good, but I never received failure_route[0] when Kamailio >> terminate the call sending 408, >> even calling like this: >> >> t_on_failure("0"); >> t_relay(); >> >> I can't find on documentation how to do this. >> Thanks a lot >> >> >> >>

2 years, 4 months

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terminate a call on 183 state

by Antonio Couto

Someone know how to terminate a call on TM module from a external application, http, websocket, udp or using other modules when the call on 183 state? I am making a answering machine detection analising audio from RTPEngine recording daemon , but i need to disconnect the call before is anwered byt remote side. Kamailio INVITE >> Destination << 100 trying << 183 session progress I need to comand Kamailio to drop this call from a external application in this point. I searched on each module but I can not find any solution. Thanks in advance

2 years, 4 months

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Re: [SR-Users] Callback for TM?s "fr_inv_timer" ?

by Antonio Couto

Thanks for your support, I am trying to do this but to be honest I don't know how to arm the failure route for this case of "fr_inv_timer". When I received reply 183 I use t_set_fr() function to set the new value for "fr_inv_timer", do far so good, but I never received failure_route[0] when Kamailio terminate the call sending 408, even calling like this: t_on_failure("0"); t_relay(); I can't find on documentation how to do this. Thanks a lot

2 years, 4 months

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Callback for TM´s "fr_inv_timer" ?

by Antonio Couto

There some way to set a callback route for TM´s "fr_inv_timer" ? I want to receive this callback when its fires and before Kamailio send 408 Request Timeout to the caller ? I tryed every callback available but I think I missing something or I will need to implement on TM module. Thanks in advance

2 years, 4 months

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jsonrpc with many methods not found

by Joseph Goettgens

Hi, I think I have not yet understood how jsonrpc needs to be configured. In kamcli I can issue the cmd "ul show" and I get a valid reply. When I print "domain list" I get error: code: 500 message: Method Not Found id: 6897 jsonrpc: '2.0' The same applies to JSONRPC Commands with Siremis. Some commands work, others are not found. Do I need some special configuration?

2 years, 4 months

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Kamailio eats pkg mem, but where?

by Sebastian Damm

Hi, we have some Kamailio instances running (currently latest 5.4 release), and we need to restart it from time to time. We have a grafana graph showing the pkg memory usage of one random tcp listener, and it increases slowly over time. Config is pure python KEMI. A mem dump directly after restarting Kamailio says this: SipSeppBook22:tmp sdamm$ grep alloc pkgmem_before.log | awk '{ print substr( $0, 16, length($0) ) }' | sort | uniq -c | sort -k1n | tail -10 16 sipproxy qm_status(): alloc'd from core: core/re.c: subst_parser(301) 31 sipproxy qm_status(): alloc'd from core: core/sr_module.c: load_module(436) 31 sipproxy qm_status(): alloc'd from core: core/sr_module.c: register_module(236) 31 sipproxy qm_status(): alloc'd from core: core/sr_module.c: register_module(253) 40 sipproxy qm_status(): alloc'd from core: core/pvapi.c: pv_init_buffer(2139) 58 sipproxy qm_status(): alloc'd from core: core/cfg.lex: pp_define(1827) 133 sipproxy qm_status(): alloc'd from core: core/rpc_lookup.c: rpc_hash_add(101) 162 sipproxy qm_status(): alloc'd from core: core/counters.c: cnt_hash_add(339) 211 sipproxy qm_status(): alloc'd from core: core/cfg.lex: addstr(1448) 265 sipproxy qm_status(): alloc'd from core: core/pvapi.c: pv_table_add(236) And after running for some weeks, the same dump looks like this: SipSeppBook22:tmp sdamm$ grep alloc prod_pkgmem.log | awk '{ print substr( $0, 16, length($0) ) }' | sort | uniq -c | sort -k1n | tail -10 31 ifens5 qm_status(): alloc'd from core: core/sr_module.c: register_module(253) 40 ifens5 qm_status(): alloc'd from core: core/pvapi.c: pv_init_buffer(2139) 59 ifens5 qm_status(): alloc'd from core: core/cfg.lex: pp_define(1827) 133 ifens5 qm_status(): alloc'd from core: core/rpc_lookup.c: rpc_hash_add(101) 161 ifens5 qm_status(): alloc'd from core: core/counters.c: cnt_hash_add(339) 203 ifens5 qm_status(): alloc'd from core: core/cfg.lex: addstr(1448) 265 ifens5 qm_status(): alloc'd from core: core/pvapi.c: pv_table_add(236) 686 ifens5 qm_status(): alloc'd from core: core/pvapi.c: pv_parse_format(1173) 694 ifens5 qm_status(): alloc'd from htable: ht_var.c: pv_parse_ht_name(158) 707 ifens5 qm_status(): alloc'd from core: core/pvapi.c: pv_cache_add(349) I know, currently there are a few lines in the code which look like this: self.instance_name = KSR.pv.get("$sht(pbxdata=>ip.%s)" % (ip,)) This has been an issue in the past and I have replaced the code with something like this: KSR.pv.sets("$var(tmpInstanceIp)", ip) self.instance_name = KSR.pv.get("$sht(pbxdata=>ip.$var(tmpInstanceIp))") However, even after changing this, the memory still grows slowly but steadily. The usage scenario is TLS-only on one side (clients) and TCP-only on the other side (pbxes). Does anybody have a hint for me how to debug this? Looks like there's a lot of pv stuff in the memory, but I don't really know where. Thanks for any hints, Sebastian

2 years, 4 months

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Re: [SR-Users] Hangup a call after a certain time from 183

by Antonio Couto

Thanks a lot I will try this options, sorry, I am new on Users Mailing List, I think that´s is not right way to make a response. regards.

2 years, 4 months

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Hangup a call after a certain time from 183

by Antonio Couto

I need to hang up a call after some seconds after received 183 provisional response but before any other SIP message. I am using "rtimer" module to set a callback on time, but how can I search for a call on transaction module in order to terminate the call? I think I am missing something to prepare the transaction data before I can use this data to terminate the call. I am thinking of storing the call-Id of all transactions to use on timeout callback, I don't know if it's possible. Thanks in advance. Antonio

2 years, 4 months

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issue using phonenum module

by David Escartín

Hello all we are using the phonenum_match_cn function to detect valid phone numbers, but seems there are some numbers like 576015145834 which are no detected as valid ones, $phn(ani=>valid) --> 0 $phn(ani=>ccname) --> <null> $phn(ani=>cctel) --> 0 $phn(ani=>error) --> Invalid number but using https://libphonenumber.appspot.com/phonenumberparser?number=576015145834&co… seems it gives a valid result Validation Results Result from isPossibleNumber() true Result from isValidNumber() true Result from isValidNumberForRegion() true Phone Number region CO Result from getNumberType() FIXED_LINE we are using kamailio 5.4 and libphonenumber tag v8.12.30 (also tested v8.9) the command is used with 'ZZ' cnc and passing the number with + maybe we are missing something in the configuration/setup? thanks and regards david -- <http://www.sonoc.io/> David Escartín NOC engineer *www.sonoc.io* <http://www.sonoc.io/> [image: LinkedIn] <https://www.linkedin.com/company/sonoc> [image: Twitter] <https://twitter.com/sonoc_> [image: Facebook] <https://www.facebook.com/SONOCio/> This e-mail is for the exclusive use of its recipients and may contain business secrets or other confidential or privileged information. Any unauthorised use, copying, publication or distribution of this e-mail is strictly prohibited. If you are not the intended recipient, please inform us immediately by replying to this e-mail and delete it, including any attachments or copies on your system. In accordance with the GDPR (EU) 2016/679 and the LOPDGDD 3/2018, we inform you that this e-mail address and/or any other personal data you have provided us with will be treated by SONOC with absolute confidentiality and with the only purpose of providing you with the requested services, due to your condition as a client, supplier or because you have requested information from us at any time. These data will only be kept for as long as required to comply with legal obligations. You can exercise your rights at any time by sending an e-mail to: *dataprotection(a)sonoc.io* <dataprotection(a)sonoc.io>*.* <dataprotection(a)sonoc.io>

2 years, 4 months

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KAZOO module on quorum queues

by David VILLAUME

Hello, Did any of you had any success with Kazoo module with RabbitMq quorum queues ? Regards, David

2 years, 4 months

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Get all contacts by domain

by Jose Fco. Irles Durá

Hi, I'm running a cloud infrastructure with multiple sip domains and I have some kamailio's as registrars with dmq for usrloc replication. The kamailio registrars are configured without db, so the aors are in ram. For some gui's I need to get all aors for one sip domain, If I execute ul.dump I get all aors, but I want only a few. In my situation it would be perfect to get all registrations for one sip domain in one request (memory, performance, cache at gui...) I thought I could save the registers with save("$rd") but ul.dump doesn't allow any parameter that restricts the "location" domain. Is there any solution for this? Best regards -- Jose Fco. Irles Durá

2 years, 4 months

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problem getting the rtp packet source port

by hamed r

hello i have problem on receive source port rtp packet client-ip = 5.202.105.250 server-ip = 194.5.205.33 client2-ip = 188.212.241.46 In the 200 ok packet from client2 ,The value 4002 is set in the audio port [image: invite.jpg] I expect rtp packets to be sent to me from source port 4002, but these packets are sent to me from another source , which is port 55755 in the image below, and I just opened this port(4002) in the firewall. If I turn on my firewall service, all packets will be dropped. I have to have this firewall [image: rtp.jpg] thanks for help me

2 years, 4 months

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Use of interface name or listening socket name as DMQ server_socket

by Ben Kaufman

Kamailio allows the use of an interface name (eth0 for example), as the identifier for the listen directive. Additionally it allows assignment of a name to be used as an identifier. This can be used, as indicated in the documentation: A unique name can be set for sockets to simplify the selection of the socket for sending out. For example, the rr and path modules can use the socket name to advertise it in header URI parameter and use it as a shortcut to select the corresponding socket for routing subsequent requests. I'm wondering if there is a way to get allow either of these parameters to be used to define DMQ's server_socket parameter. This would be useful in containerization situations where the local IP address is assigned dynamically. Here's what I'm attempting: #!KAMAILIO loadmodule "tm" loadmodule "sl" loadmodule "dmq" loadmodule "textops" listen=udp:eth0:5060 name "sip5060" listen=udp:eth0:5090 name "sip5090" modparam("dmq", "server_address", "sip:dmq_quicktest:5090") modparam("dmq", "notification_address", "sip:dmq-bus:5090") modparam("dmq", "server_socket", "sip5090" ) ## Above line doesn't work, neither does the line below: #modparam("dmq", "server_socket", "udp:eth0:5090" ) modparam("dmq", "multi_notify", 1) modparam("dmq", "num_workers", 3) modparam("dmq", "ping_interval", 30) request_route { exit; } I get the following error during startup: 0(1) ERROR: dmq [dmq.c:241]: mod_init(): server_uri is not a socket the proxy is listening on Thoughts? I realize that I can configure and pass static networking with the containers, but I'm looking for a way to have this set on the fly - especially when testing, etc. Ben Kaufman

2 years, 5 months

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SIP trunk challange

by Mihai

Hello ! I have this issue with the telco provider they give the sip trunk on a private conection with private ips, the problem is that i am trying to relocate the sip trunk in remote server and I've tried though basic networking/tunnels/vpn and no go; their sbc is dropping requests and rejects if they are more via headers, they force me to use that conection and the ip allocated. The challenge is how to this is in kamailio, basically i just want to transport the trunk (similiar to a reverse proxy for web) to its new destination and from there handle everything. Any hints on how to achieve this?

2 years, 5 months

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Kamailio vs FreeSWITCH as SBC

by Unai Rodriguez

Dear List, I’m looking at using Kamailio as SBC. This would replace our current FreeSWITCH-based SBCs. What are your thoughts on this move? Would there be any known reasons why Kamailio would be superior to FreeSWITCH as an SBC? Thank you so much Unai Rodriguez

2 years, 5 months

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t_drop_replies can't be used on a route?

by David Villasmil

Hello guys, I have a failure route which calls another route, and on this route I need to t_drop_replies(); But i'm getting: [core/cfg.y:3510]: yyerror_at(): parse error in config file /etc/kamailio/include/relay_routes.cfg, line 143, column 20: Command cannot be used in the block Any way to fix this without me having to use it in the failure_route? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 years, 5 months

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remove header in async processing (e.g. by http_async_client)

by Henning Westerholt

Hello, I'd like to remove headers during async processing e.g. in a route called from http_async_client(..). The usual textops functions are not working here. Adding headers could be done using something like rtjson, but it do not offer a remove header function. I could of course loop the request locally, but this is not that "nice". Any other suggestions for this case? Or just the rtjson module needs some extensions? Thank you, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com<https://gilawa.com/>

2 years, 5 months

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Siremis startup helper service for systemd

by Joseph Goettgens

Hi, I had some difficulties getting Siremis 5.3 to run under Debian 11 Bullseye. It turned out that the problem was the creation of the Siremis socket: <UnixSockLocal name="unixsocklocal" address="/run/siremis/siremis_rpc.sock" timeout="3.0"/> There is a warning in siremisJRCommands.xml that points out that systemd has to be reconfigured as sockets in /tmp are no longer shared (PrivateTmp setting). By the same token, /run/siremis/siremis_rpc.sock cannot be created as the httpd server is not allowed to write to /run, i.e. it can neither create the socket nor the siremis directory. The permanent creation of the siremis directory with suitable access rights is also not possible, since /run is mounted as tmpfs in the default configuration. At least the directory has to be created at boot time, which can be done with a small helper service. This is working for me (siremis-helper.service): [Unit] Description=Siremis Helper Service After=network.target [Service] Type=oneshot ExecStartPre=/usr/bin/mkdir /var/run/siremis ExecStart=/usr/bin/chown www-data:www-data /var/run/siremis [Install] WantedBy=multi-user.target It is enabled the usual way with systemctl "systemctl start --now siremis-helper". Siremis now works with jsonrpc commands and the procedure would mean that installing Siremis on a newer Debian system would definitely require some attention. Does anyone have any objections to the above procedure?

2 years, 5 months

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Re-establishment of a closed TCP connection

by Vitalii Krutsitskyi

Hello General overview of the infrastructure: Client (Cisco Hardware) ---> AWS NLB ---> Kamailio Instances The client has a time limit for the opened TCP connections - 10 minutes. This is a hardcoded value that can't not be changed. Regardless of the traffic that flows through the connection, it is closed after 10 minutes. Problem: in case of long-running calls above 10 minutes it is not possible for Kamailio to send back SIP traffic back to the client using the original TCP connection. The client has been asked to either increase the time limit or send SIP reinvites - none of that is possible due to some client's infrastructure limitations. Question: is it possible to force Kamailio to open a new TCP connection and associate it with the call? Client has a 5060 port always opened which can accept incoming connections.

2 years, 5 months

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ipops dns_query not working using Kemi

by Thomas de Rooij | CM.com

Hi all, We are trying to do a DNS query using Kemi: KSR.ipops.dns_query(domain, "xyz") The result is always -3. This is returned here: https://github.com/kamailio/kamailio/blob/5.5.4/src/modules/ipops/ipops_pv.… It seems weird to me that the error code is returned because the same non-Kemi code works just fine. Could it be related to this issue: https://github.com/kamailio/kamailio/issues/2052 Thanks. Kind regards, Thomas

2 years, 5 months

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remove headers with less than 5 characters with textops

by Patrick Karton

Hello im trying to textops module to remove headers with name length less than 5 characters with regular expression but it seems impossible or the module does not handle $ char remove_hf_re("^.{1,5}$")

2 years, 5 months

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Siremis JSONRPC problem on Bullseye

by Joseph Goettgens

Hi, I seem to have a problem setting up Siremis 5.3 on a Debian 11 box using the Kamailio 5.6.0 release (installed from repo). When I enter something like core.uptime into the JSONRPC Command Panel, I get a "System Internal Error" alert. The Apache error log then says: [Mon Jun 06 10:48:41.459377 2022] [php7:notice] [pid 22322] [client 192.168.xxx.xxx:45030] socket bind failed: No such file or directory, referer: http://deb11srv1-kamailio/siremis/sipadmin/jrcmds The path siremis/sipadmin/jrcmds obviously does not seem to exist and I assume that one needs some internal adjustments for some paths. As far as the socket is concerned, siremis/modules/sipadmin/service/siremisJRCommands.xml has the definition: UnixSockRemote name="unixsockremote" address="/var/run/kamailio/kamailio_rpc.sock" timeout="3.0"/> and the corresponding definition in /etc/kamailio/kamailio.cfg is: modparam("jsonrpcs", "dgram_socket", "/var/run/kamailio/kamailio_rpc.sock") modparam("jsonrpcs", "dgram_mode", 0666) which looks fine to me. I also checked for the existence of the socket and its access rights. Any suggestions?

2 years, 5 months

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t_cancel_callid - doesn't insert Reason header, if called from branch 0

by Pavel Veselovsky

Hallo all, I defined branch_failure routing block via t_on_branch_failure() In the branch_failure route, in case of "408 Request timeout" I am sending CANCEL with reason "t_cancel_callid("$ci", "$cs", "22", "408")". If the t_cancel_callid("$ci", "$cs", "22", "408") is called in branch_failure for branch idx 0, the CANCEL is sent, but Reason header is missing there. Why? (I expect that t_cancel_callid() should insert Reason header regardless of branch it is called from) If the t_cancel_callid is called in branch idx !=0, the CANCEL contains Reason header with cause as expected. Thanks for clarification of this behavior, Pavel. P.S. code snipset: event_route[tm:branch-failure:primary] { if ($T(reply_code) == "408") { if ($avp(branch_idx) == $T_branch_idx) { if (t_cancel_callid("$ci", "$cs", "22", "408")) { xlog("L_INFO", "CANCEL sent ci:$ci, cs:$cs 408"); } else { xlog("L_INFO", "CANCEL not sent ... ci:$ci, cs:$cs 408"); } $var(_palotmp_) = "sip:" + $avp(replaced_cfnry_target_number) + "@" + $dd + ":" + $dp + ";user=phone"; append_branch($var(_palotmp_),"1.0"); t_relay(); return; } } }

2 years, 5 months

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Fatal Python error: failed releasing import lock after fork

by Nigel Daniels

The logs appear as follows: Fatal Python error: failed releasing import lock after fork Current thread 0x 7(13) DEBUG: <core> [core/utils/sruid.c:121]: sruid_init(): root for sruid is [srid-62930e67-d-] (0 / 16) kamailio | 00007fe8fb8a4180 (most recent call first): kamailio | <no Python frame> kamailio | 7(13) DEBUG: <core> [core/sr_module.c:875]: init_mod_child(): idx 7 rank -2: tm [ctl handler] kamailio | 7(13) DEBUG: tm [callid.c:136]: child_init_callid(): callid: '06d932e436730fa0-13(a)10.10.0.4' kamailio | 7(13) DEBUG: <core> [core/sr_module.c:875]: init_mod_child(): idx 7 rank -2: tmx [ctl handler] kamailio | 7(13) DEBUG: tmx [tmx_mod.c:262]: child_init(): rank is (-2) kamailio | 7(13) DEBUG: <core> [core/sr_module.c:875]: init_mod_child(): idx 7 rank -2: sl [ctl handler] kamailio | 7(13) DEBUG: <core> [core/sr_module.c:875]: init_mod_child(): idx 7 rank -2: ctl [ctl handler] kamailio | 7(13) DEBUG: <core> [core/sr_module.c:875]: init_mod_child(): idx 7 rank -2: debugger [ctl handler] kamailio | 7(13) DEBUG: debugger [debugger_mod.c:224]: child_init(): rank is (-2) kamailio | 7(13) DEBUG: <core> [core/sr_module.c:875]: init_mod_child(): idx 7 rank -2: app_python3 [ctl handler] kamailio | 7(13) DEBUG: app_python3 [apy_kemi.c:234]: sr_apy_kemi_exec_func_ex(): execution of method: .info kamailio | 7(13) DEBUG: app_python3 [apy_kemi.c:266]: sr_apy_kemi_exec_func_ex(): param[1] for: info is str: ===== kamailio.child_init(-2) kamailio | kamailio | 7(13) INFO: <core> [core/kemi.c:106]: sr_kemi_core_info(): ===== kamailio.child_init(-2) kamailio | 7(13) DEBUG: ctl [ctl.c:329]: mod_child(): ctl: mod_child(0), fork_process=0, csl=0x556774893180 kamailio | 7(13) DEBUG: ctl [ctl.c:336]: mod_child(): ctl: 0 io_listen_loop(1, 0x556774893180) kamailio | 7(13) INFO: ctl [io_listener.c:213]: io_listen_loop(): io_listen_loop: using epoll_lt io watch method (config) kamailio | 7(13) DEBUG: ctl [io_listener.c:245]: io_listen_loop(): io_listen_loop: adding socket 8, type 2, transport 3 (/var/run/kamailio//kamailio_ctl) kamailio | 7(13) DEBUG: ctl [../../core/io_wait.h:374]: io_watch_add(): processing io_watch_add(0x7fe8f9aef380, 8, 2, 0x556774893180) - fd_no=0 Running python 3.8.10 using the suggested. # ----------------- setting module-specific parameters --------------- # ----- 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) # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) ####### Routing Logic ######## modparam("app_python3", "load", "/opt/kamailio/etc/kamailio.py") cfgengine "python"a import sys import KSR as KSR def dumpObj(obj): for attr in dir(obj): KSR.info("obj.%s = %s\n" % (attr, getattr(obj, attr))) def mod_init(): KSR.info("===== from Python mod init\n") dumpObj(KSR) return kamailio() class kamailio: def __init__(self): KSR.info('===== kamailio.__init__\n') def child_init(self, rank): KSR.info('===== kamailio.child_init(%d)\n' % rank) return 0 def ksr_request_route(self, msg): KSR.info("===== request - from kamailio python script\n") KSR.setdsturi("sip:alice@127.0.0.1:5080") KSR.tm.t_on_branch("ksr_branch_route_one") KSR.tm.t_on_reply("ksr_onreply_route_one") KSR.tm.t_on_failure("ksr_failure_route_one") KSR.sl.send_reply(100, "Trying") if KSR.tm.t_relay() < 0 : KSR.sl.send_reply(500, "Server error") return 1 def ksr_reply_route(self, msg): KSR.info("===== response - from kamailio python script\n") return 1 def ksr_onsend_route(self, msg): KSR.info("===== onsend route - from kamailio python script\n") return 1 def ksr_branch_route_one(self, msg): KSR.info("===== branch route - from kamailio python script\n") return 1 def ksr_onreply_route_one(self, msg): KSR.info("===== onreply route - from kamailio python script\n") return 1 def ksr_failure_route_one(self, msg): KSR.info("===== failure route - from kamailio python script\n") return 1 Any help is appreciated, thanks. -- - Nigel

2 years, 5 months

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SNMP, export concurrent calls count

by ZZ Wave

Hello. Debian 10, Kamailio 5.2.1 I want to monitor concurrent calls in/out per every trunk connected to this proxy (it's done with dialog profiles), and export this to Zabbix. But I can't find in snmpstats documentation how to export user-defined variables. Help me please. Maybe SNMP isn't the best solution for that...

2 years, 5 months

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Centos kamailio.repo file is not updated

by Patrick Wakano

Hello ist, I've noticed some of the latest releases do not have their repo mentioned in the https://rpm.kamailio.org/centos/kamailio.repo The repos are there, it is just the kamailio.repo file missing some of the newer versions like 5.4.7, 4.5.8, 5.5.4, 5.6.0... Cheers, Patrick

2 years, 5 months

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Fedora RPMs

by Tim Anderson

Hi, I can't find any RPMs for Fedora 35 or 36. Have they moved somewhere other than rpm.kamailio.org? Regards, Tim

2 years, 5 months

3
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DNS Failover try next destination on receiving code other than 503

by Patrick Karton

Hello, im using DNS Failover and everything works well except i would like to try other resolved ips when i receive a 408 from the previous ones As far as i know only a received 503 will trigger next destinations and there is no config parameters in tm module to handle that behaviour. so how can i implement it ? set a failure route before relay and just call t_relay() after again ? also is there a way in failure route to know how many destinations are left to exit failure route. Thanks for help.

2 years, 5 months

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KEMI Python Syntax

by Olivier Roy

Hello, I am writing some Kamailio scripts using app_python3. I understand that the KSR module is automatically generated by Kamailio. So when I import KSR from KSR I don't get any syntax from my IDE since it doesn't know where KSR is. My question: is there a way to generate a KSR.py file that can be used for autocompletion and syntax highlighting? Thanks. Olivier

2 years, 5 months

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get and store destination ip address and port that the request will be sent to

by Patrick Karton

Hello is there a way or pseudo var to get ip address & port that the request will be sent to and save it in to another variable. i want a way to get it to store it. im not able to save it with to_ip and to_port of the onsend_route. TThanks.

2 years, 5 months

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Find Values in User part of Diversion URI

by Lewis Hutchinson

Sorry if this is a super basic ask but I can't seem to work it out, I have got a work around but for my own sanity wanted to post the question on here. In Kamailio Native 5.5.2 I want to be able to look into this field $(di{uri.user}) and "find" or see if it "contains" a value i.e Diversion: "TEST" sip:112233445566@xxx.xxx.xxx.xxx;user=phone; find 234 or contains 234 I know how to do this in in KEMI Python but for the life of me cant get my head around it in Native. 2ndly is there an opposite to {s.numeric} Removes all non-numeric parts of string. I want the ability to strip all numeric parts. Keep us the good work. Lewis

2 years, 5 months

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TLS connection failure. Not sure if system behavior is correct.

by Володимир Іванець

Hello all! My test Kamailio setup (now updated to version 5.5.4) that was working with MS Teams some time ago, now stopped :) I was wondering if someone could tell me if the current behavior is correct or if there is a problem? Here is a screenshot of the packet capture: [image: image.png] I thought that after MS Teams side sent Change Cipher Spec request, Kamailio should respond with ACK. But instead it just starts sending data (I believe it sends OPTIONS request. sipdump module captured it). Is this correct? There seems to be no issues in the Kamailio debug log: May 20 15:29:05 server kamailio[28961]: 14(28975) DEBUG: <core> [core/io_wait.h:782]: io_watch_chg(): DBG: io_watch_chg (0xae9560, 24, 0x1, 0xffffffff) fd_no=20 called May 20 15:29:05 server kamailio[28961]: 14(28975) DEBUG: <core> [core/io_wait.h:600]: io_watch_del(): DBG: io_watch_del (0xae9560, 25, -1, 0x0) fd_no=20 called May 20 15:29:05 server kamailio[28961]: 14(28975) DEBUG: <core> [core/tcp_main.c:4471]: handle_tcpconn_ev(): sending to child, events 1 May 20 15:29:05 server kamailio[28961]: 14(28975) DEBUG: <core> [core/tcp_main.c:4144]: send2child(): selected tcp worker idx:1 proc:11 pid:28972 for activity on [tls:172.16.30.206:5062], 0x7fad6a70f6c8 May 20 15:29:05 server kamailio[28961]: 11(28972) DEBUG: <core> [core/tcp_read.c:1737]: handle_io(): received n=8 con=0x7fad6a70f6c8, fd=6 May 20 15:29:05 server kamailio[28961]: 11(28972) DEBUG: tls [tls_domain.c:1208]: tls_lookup_private_key(): Private key lookup for SSL_CTX-0x7fad6a5e1eb8: (nil) May 20 15:29:05 server kamailio[28961]: 11(28972) DEBUG: <core> [core/tcp_main.c:2720]: tcpconn_do_send(): sending... May 20 15:29:05 server kamailio[28961]: 11(28972) DEBUG: <core> [core/tcp_main.c:2753]: tcpconn_do_send(): after real write: c= 0x7fad6a70f6c8 n=2279 fd=6 May 20 15:29:05 server kamailio[28961]: 11(28972) DEBUG: <core> [core/tcp_main.c:2754]: tcpconn_do_send(): buf= May 20 15:29:05 server kamailio[28961]: 7 May 20 15:29:05 server kamailio[28961]: 11(28972) DEBUG: <core> [core/io_wait.h:375]: io_watch_add(): processing io_watch_add(0xb555c0, 6, 2, 0x7fad6a70f6c8) - fd_no=1 May 20 15:29:05 server kamailio[28961]: 11(28972) DEBUG: tls [tls_domain.c:1208]: tls_lookup_private_key(): Private key lookup for SSL_CTX-0x7fad6a5e1eb8: (nil) May 20 15:29:05 server kamailio[28961]: 11(28972) DEBUG: tls [tls_domain.c:790]: sr_ssl_ctx_info_callback(): SSL handshake done May 20 15:29:05 server kamailio[28961]: 11(28972) DEBUG: tls [tls_domain.c:794]: sr_ssl_ctx_info_callback(): SSL disable renegotiation May 20 15:29:05 server kamailio[28961]: 11(28972) DEBUG: tls [tls_server.c:542]: tls_connect(): TLS connect successful May 20 15:29:05 server kamailio[28961]: 11(28972) DEBUG: tls [tls_server.c:549]: tls_connect(): tls_connect: new connection to 52.114.132.46:5061 using TLSv1/SSLv3 ECDHE-RSA-AES256-GCM-SHA384 256 May 20 15:29:05 server kamailio[28961]: 11(28972) DEBUG: tls [tls_server.c:552]: tls_connect(): tls_connect: sending socket: 172.16.30.206:0 May 20 15:29:05 server kamailio[28961]: 11(28972) DEBUG: tls [tls_server.c:418]: tls_dump_cert_info(): tls_connect: server certificate subject:/CN=sip.pstnhub.microsoft.com May 20 15:29:05 server kamailio[28961]: 11(28972) DEBUG: tls [tls_server.c:422]: tls_dump_cert_info(): tls_connect: server certificate issuer:/C=US/O=Microsoft Corporation/CN=Microsoft RSA TLS CA 01 May 20 15:29:05 server kamailio[28961]: 11(28972) DEBUG: <core> [core/tcp_main.c:2720]: tcpconn_do_send(): sending... May 20 15:29:05 server kamailio[28961]: 11(28972) DEBUG: <core> [core/tcp_main.c:2753]: tcpconn_do_send(): after real write: c= 0x7fad6a70f6c8 n=509 fd=6 May 20 15:29:05 server kamailio[28961]: 11(28972) DEBUG: <core> [core/tcp_main.c:2754]: tcpconn_do_send(): buf= May 20 15:29:05 server kamailio[28961]: ▒4▒▒▒▒.▒2T$-▒▒ o▒w)▒(▒▒^▒D▒U{xn9▒' ▒&▒t▒▒)▒▒?▒d▒=g▒▒?▒▒▒4/▒▒▒▒-R+▒tR▒H,▒o▒D2r ▒5Q?m▒▒9▒▒#▒O▒▒▒\eϣ▒݄"▒z=▒U▒? May 20 15:29:05 server kamailio[28961]: 14(28975) DEBUG: <core> [core/io_wait.h:600]: io_watch_del(): DBG: io_watch_del (0xae9560, 24, -1, 0x0) fd_no=19 called May 20 15:29:05 server kamailio[28961]: 14(28975) DEBUG: <core> [core/tcp_main.c:4471]: handle_tcpconn_ev(): sending to child, events 1 May 20 15:29:05 server kamailio[28961]: 14(28975) DEBUG: <core> [core/tcp_main.c:4144]: send2child(): selected tcp worker idx:2 proc:12 pid:28973 for activity on [tls:172.16.30.206:5062], 0x7fad6a6f25a0 May 20 15:29:05 server kamailio[28961]: 12(28973) DEBUG: <core> [core/tcp_read.c:1737]: handle_io(): received n=8 con=0x7fad6a6f25a0, fd=6 May 20 15:29:05 server kamailio[28961]: 12(28973) DEBUG: tls [tls_domain.c:1208]: tls_lookup_private_key(): Private key lookup for SSL_CTX-0x7fad6a5e1eb8: (nil) May 20 15:29:05 server kamailio[28961]: 11(28972) DEBUG: <core> [core/tcp_read.c:304]: tcp_read_data(): EOF on connection 0x7fad6a70f6c8 (state: 0, flags: 18) - FD 6, bytes 0, rd-flags 10000 ([52.114.132.46]:5061 -> [52.114.132.46]:0)11(28972) DEBUG: <coreTCP closed event creation triggered (reason: 0) May 20 15:29:05 server kamailio[28961]: 11(28972) DEBUG: <core> [core/tcp_read.c:192]: tcp_emit_closed_event(): no callback registering for handling TCP closed event May 20 15:29:05 server kamailio[28961]: 11(28972) DEBUG: <core> [core/tcp_read.c:1503]: tcp_read_req(): EOF May 20 15:29:05 server kamailio[28961]: 11(28972) DEBUG: <core> [core/io_wait.h:600]: io_watch_del(): DBG: io_watch_del (0xb555c0, 6, -1, 0x10) fd_no=2 called May 20 15:29:05 server kamailio[28961]: 11(28972) DEBUG: <core> [core/tcp_read.c:1878]: handle_io(): removing from list 0x7fad6a70f6c8 id 2 fd 6, state 2, flags 18, main fd 25, refcnt 2 ([52.114.132.46]:5061 -> [52.114.132.46]:0) May 20 15:29:05 server kamailio[28961]: 11(28972) DEBUG: <core> [core/tcp_read.c:1659]: release_tcpconn(): releasing con 0x7fad6a70f6c8, state -1, fd=6, id=2 ([52.114.132.46]:5061 -> [52.114.132.46]:0) May 20 15:29:05 server kamailio[28961]: 11(28972) DEBUG: <core> [core/tcp_read.c:1660]: release_tcpconn(): extra_data 0x7fad6a70f008 May 20 15:29:05 server kamailio[28961]: 14(28975) DEBUG: <core> [core/tcp_main.c:3574]: handle_tcp_child(): reader response= 7fad6a70f6c8, -1 from 1 May 20 15:29:05 server kamailio[28961]: 14(28975) DEBUG: tls [tls_server.c:729]: tls_h_tcpconn_close_f(): Closing SSL connection 0x7fad6a70f008 Thank you!

2 years, 5 months

1
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Debian: Kamailio: -> FIN - SYN -> on new port

by Kevin Browne

Hi, am looking for some specific Kamailio help i cant seem to find anywhere ( # running Kamailio on Debian/buster ... ## after receipt of tcp FIN, port is closed. when signaling is required again from Kamailio side. Kam is attempting SYN on FINd (closed) port. this is rejected RST. How to change SYN to re-establish tcp on new ephemeral port? ## another description, same problem: FIN received from (external) cisco (call owner) due to port recycling, not to keep port open after x minutes. everything is as expected re ACKs and such. When Kam wants to send REFER to cisco after the FIN, tcp attempts a SYN (as we'd expect) but sends SYN on old closed port. ## Where to change this Kam/OS/other is where i'm attempting to identify, then figure how to change. Thank you.

2 years, 5 months

2
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TOPOS + RE-INVITE problem - Contact IP not updated

by c c

It seems that TOPOS is not working properly on RE-INVITE and stop masking IP and contact on some RE-INVITE. When a RE-INVITE is send with Kamailio IP as Request-URI TOPOS is not operating. Note : we are using the HTABLE trick to rewrite $ru and match the call but we would need to have TOPOS re-evaluating the dialog once done. We use a standard configuration with 2 modifications. 1. in route[DISPATCH] to set the HTABLE if(is_method("INVITE")) { $sht(ct=>$ci::$ft) = $sel(contact.uri); } 1. in route[WITHINDLG] to find the call if R-URI = myself if(has_totag() && uri==myself) { if($sht(ct=>$ci::$ft) != $null && $T_reply_code > 400) $shtex(ct=>$ci::$ft) = 10 ; if($sht(ct=>$ci::$tt) != $null) { $ru = $sht(ct=>$ci::$tt); if (is_method("BYE")) $shtex(ct=>$ci::$tt) = 10 ; route(RELAY); } } As you can see in the PCAP the RE-INVITE goes directly to calling stating with the Contact not updated and the 200 OK is also sent to called party with wrong Contact info too. [ https://github.com/kamailio/kamailio/files/8730489/topos-issue.zip ] This behavior breaks the BYE procedure. Any advice ? Chris

2 years, 5 months

2
1
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Load balancer and consult transfer

by Anh Cao

Hi all, I'm building an simple PBX with 1 Kamailio as load balancer for several Asterisk servers. My issue is if an extension is in-call on Asterisk#1 wants to consult transfer to another number, it makes another call and Kamailio forward this new call to another Asterisk and cause call tranfer will fail. Does anyone here have tips or sugestions to resolve this? Thanks in advance for your help. Best regards

2 years, 5 months

3
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Kamailio v5.6.0 Released - new major version is out

by Daniel-Constantin Mierla

Kamailio v5.6.0 is out – it comes with 7 new modules and a considerable set of improvements touching again more than 90 existing modules. You can read detailed release notes at: * https://www.kamailio.org/w/kamailio-v5-6-0-release-notes/ Many thanks to all developers and community members that made possible this release. v5.6.0 brings more flexibility and optimizations across many components, plenty of new variables and transformations, load balancer enhancements, generic iterators for SIP headers and body lines, alternative http client, NATS PubSub connector, improvements to STIR/SHAKEN, IMS/VoLTE, rtpengine and tls as well as more functions exported to KEMI interface ... just to list only a few here. Enjoy Kamailio v5.5.0! Thank you for flying Kamailio! 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, 5 months

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Releasing Kamailio v5.6.0

by Daniel-Constantin Mierla

Hello, with some effort directed lately to convert the wiki content to markdown files and host them on github, it's time to now to nail down the path to release nest major release for Kamailio, respectively v5.6.0. What's new in this major release as well as upgrade guidelines have to be built, but I hope to be done in time to release Kamailio v5.6.0 either on Monday, June 23, 2022 or on Wednesday, June 25, 2022. Should anyone consider a different release planning, let's discuss it. If anyone wants to help with them, the content has to be added to the github wiki repository: * https://github.com/kamailio/kamailio-wiki If not there, new files have to be created and linked from the main page. The wiki is built with mkdocs into html and made available for browsing at: * https://kamailio.org/wikidocs/ 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, 5 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 using TLSv1/ TLSv1.1 and only allowing TLS1.2+ So change your kamailio TLS settings to enforce only it method = TLSv1.2+ After restarting kamailio your team's connections should be good. -- Best Regards, Ahmed Fouad

2 years, 5 months

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LCR GW

by Antoine DEBARRE

Hi, Is there a way to specify an outbound proxy for just a few lcr gateways added in table lcr_gw ? Thanks in advance ! Antoine

2 years, 5 months

2
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Path MTU issues over UDP/IPv6

by Rick van Rein

Hello, IPv6 routers never fragment packets. Rather, they drop a packet that is too large for a (local) MTU and send back ICMPv6 "Packet Too Big". This seems to cause loss of larger SIP messages when an ISP tunnels their IPv6 at the expense of the MTU. The pmtu_discovery flag sets Don't Fragment in IPv4 traffic; in IPv6 this is an implied property. Does Kamailio learn a lower MTU from any "Packet Too Big" for IPv6 even if pmtu_discovery is not set? Future resends can then be fragemented appropriately. The udp_mtu setting diverts to another protocol, but that would be a setting as low as the worst peer, impacting all. It would be a weird struggle with a telco serving many. PMTU would be better to rely on, but how does it work in Kamailio? Details on https://www.rfc-editor.org/rfc/rfc3542#section-11.3 https://stackoverflow.com/questions/38817837/how-does-mtu-retransmission-wo… Thanks, -Rick

2 years, 5 months

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Forwarding SIP status code to HTTP client

by Tim Anderson

Hi, I have a kamailio that is has an HTTP message server. Is it possible to send the SIP status code back in the HTTP reply? Is it possible to suspend an xhttp request while some other stuff happens and then send the reply later? The relevant bit of the config currently looks like event_route[xhttp:request]{ if($hu == "/sms"){ $uac_req(method)="MESSAGE"; $uac_req(ruri)="sip:1234@example.com"; $uac_req(furi)="sip:4567@example.com"; $uac_req(turi)="sip:1234@example.com"; $uac_req(body)=$rb; uac_req_send(); xhttp_reply("200", "OK", "text/plain", "Sent") } } Regards, Tim

2 years, 5 months

2
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Kamailio Stop Responding Randomly

by M Arqum CH

Kamailio first gives timeout on ctl unix socket as I am exporting data using prometheus exporter, after a few minutes it stops responding to any sip request on default/production udp port binded with multiple IP addresses for sip traffic. The other listing port which is used by keepalived check script for bing-pong keeps responding to option messages. Its very random behaviour occurs after a few days or after a few months, the system having max load 350 concurrent calls with load average 200-300. I am attaching core dump which is collected before restarting the process. Please let me know if more information is required. version: kamailio 5.4.1 (x86_64/linux) 2ca223 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: 2ca223 Thankyou in advance. -- Regards M Arqum

2 years, 5 months

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Relay wss to Asterisk

by Michal Hajek

Hello, can i configure kamailio for relay wss to asterisk/pjsip? WebRTC client ( chrome ) --> WSS -> Kamailio --> WSS -> Asterisk with pjsip Asterisk accept wss://ip_address:wss_port/ws Thanks Michal Hajek

2 years, 5 months

2
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Split audio and video into different calls

by Rüdiger, Christoph

Hi everyone, we are running a call center with really good to use hardphones and audio only up to this point. Because of changing regulation, we soon have to deal with video calls as well, but only for a fraction of the calls (below 1%). Instead of switching everyone to clumsy softphones for the rare case of a video call, I’m exploring the ability to branch an incoming call into a pure audio call to the hardphone and optionally a pure video call to the softphone. Has anyone done this in the past and can share some experiences/ideas on 1. Can it be done purely with Kamailio and rtpengine? I don’t have a clear picture of how to do it purely with Kamailio on the SIP side, because we end up with two independent legs (to the hardphone and the softphone). Or do I need to deploy an Asterisk, Yate, freeswitch, etc. and doing it as a 3-way conference where one party gets both RTP streams (the trunk) and the other two parties only get one RTP stream (either audio or video)? 2. Was it worth the effort or a steady candidate for troubles? I’m looking forward to your thoughts, especially on question b). Have a nice weekend, Christoph --- Christoph Rüdiger Digital Operations Architect DTX OU DACH M +49 152 3828 0376 | christoph.ruediger(a)tkelevator.com<mailto:christoph.ruediger@tkelevator.com> TK Aufzüge GmbH | Münchener Str. 100c | 45145 Essen | Deutschland | www.tkelevator.com/de<http://www.tkelevator.com/de> VORSITZENDE DES AUFSICHTSRATS Inge Delobelle | GESCHÄFTSFÜHRUNG Burkhard Schlenker (Vorsitzender) | Christian Fongern | Katja Schrafft | SITZ DER GESELLSCHAFT Neuhausen a. d. F. | REGISTERGERICHT Stuttgart HRB 731152 This e-mail (including any attachments) may contain confidential and/or privileged information. Any unauthorized use or dissemination of this message in whole or in part is strictly prohibited. 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.

2 years, 5 months

4
5
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kamailio WebSocket not respond sometimes

by dolphin

Hello, kamailio could not respond to the question: This machine is on aws, there is no high load, there is an external ip listening to the WebSocket 443 port, and sometimes there are some unknown detection signals "SSTP_DUPLEX_POST" that will cause port 443 to fail to respond. Need to restart kamailio Then, it stopped responding again after about a month. Attach the detailed log details: root@sip1-aws:~# /home/pkg/kamailio/sbin/kamailio -V version: kamailio 5.3.9 (x86_64/linux) a5fd0a 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 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: a5fd0a compiled on 22:31:18 Mar 31 2022 with gcc 5.4.0 Thanks.

2 years, 5 months

2
2
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RabbitMQ hangs if connection lost

by Avremy Cohen

Hi When I Publish to RabbitMQ and the connection is down, the server just hangs and there is no timeout. I am using KEMI with python3 and have tried different timeout functions to circumvent the hanging but it doesn't seem to work 1. Is there a way to tell if RabbitMQ is connected? 2. using python is there a way to timeout a KSR function? Thanks guys!

2 years, 5 months

1
0
0 0

Kamailio Crashed

by M Arqum CH

Dear Team, Kamailio instance got crashed, I found this error in logs. Please give advice on how I can avoid this in future. *kernel: [52641555.303165] kamailio[123814]: segfault at 30 ip 00007f42f70db770 sp 00007ffd5d95a9f0 error 4 in dialog.so[7f42f7033000+c9000]* version: kamailio 5.4.1 (x86_64/linux) 2ca223 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: 2ca223 compiled on 12:51:24 Sep 21 2020 with gcc 6.3.0 Thank you in advance. Regards M Arqum

2 years, 5 months

2
2
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Failing Audio stream on Kamailio+webrtc with integration to asterisk server

by Kelvin Ezenwaka

Summary: We have a Kamailio server as our sip proxy server, sip firewall with WebSocket and RTP engine configured on it (the Kamailio Server). But we experience one way audio, no audio, hang up after 30s, when we try making calls between internal extensions (eg extension 100 to call extension 105) and external calls (eg extension 100 to call mobile number 09056925668) as well. The call flow is as: Webrtc client<------->Kamailio1+rtpengine<-----Asterisk---->Kamailio2<-------->Telco Provider The asterisk communication between all three boxes is via local IP (All ports open between them). The Kamailio 1 box where we have our sip registration cache, has rtp ports and wss port open on the internet. Asterisk 18.9.0 Kamailio 5.5.3 + RTPengine 10.4.0.0 Debian 11 bullseye Kindly see attached below a diagram depicting the VoIP network flow and also attached are logs files for the webrtc client and server side. Ezenwaka Kelvin Arinze VOIP Engineer Tel: (+234) 9056925668 DISCLAIMER: The message and its attachments are for designated recipient(s) only and may contain privileged, proprietary and private information. If you have received it in error, kindly delete it and notify the sender immediately. PressOne accepts no liability for any loss or damage resulting directly and indirectly from the transmission of this e-mail message.

2 years, 5 months

2
1
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Question on pike_check_ip(ipaddr)

by Joel Serrano

Hello, One of our customers has lots of extensions that register to our platform using all the same IP. Sometimes they get caught in our pike velocity rules but if we increase the values to accommodate this client then all the others could abuse the system. I was wondering if pike_check_ip(ipaddr) needs strictly an IP as parameter or can it be a string? I was thinking of using pike instead of by $si by something like $fU-$si but I don't know if this would work? I need to either handle separately different extensions from the same IP or a way to adjust the pike config values just for one specific IP... I'm not sure either of these is possible with pike? My alternative is just use table and mimic the functionality myself which I don't think is too complicated, but I'd love to know if I'm missing a fast-path here... Thanks! Joel.

2 years, 5 months

2
1
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Accept IP/Domain Register

by Social Boh

Hello, maybe is a very basic question; How Kamailio can accept any kind of register, over IP or domains. Example, accept authenticate REGISTER from ip 1.2.3.4 or sip.domain.org or sip1.domain.org, domains associated with ip 1.2.3.4 Thank you in advance. Regards -- --- I'm SoCIaL, MayBe

2 years, 6 months

3
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sr-users