- sr-users - kamailio.org

secfilter (and more modules) in kamailio kemi
by bazrafshan.m.h＠protonmail.com 28 Jul '24

28 Jul '24

Hi guys, some modules are not shown in Kamailio Kemi's documentation, such as secfilter. Is there any way to load them and use them in Python? a way is to simulate the secfilter but it is not a general solution. (Sorry in advance for the obvious question)

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secfilter (and more modules) in kamailio kemi
by bazrafshan.m.h＠protonmail.com 28 Jul '24

28 Jul '24

Hi guys, some modules are not shown in Kamailio Kemi's documentation, such as secfilter. Is there any way to load them and use them in Python? a way is to simulate the secfilter but it is not a general solution. (Sorry in advance for the obvious question)

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Job: System Engineer VoIP - Germany / HO
by Gerrit Jacobsen 25 Jul '24

25 Jul '24

VIER is looking for a System Engineer VoIP What VIER does VIER is a global contact center technology provider. VIER develops and runs its own technology in several data centers. You will be part of the Real-time Media Processing Team which is responsible for the telephony connections between carriers, applications and call-center agents. Your mission You are responsible for the support of the VoIP systems as well as the associated infrastructure Since you have sovereignty over the VoIP systems, you are also responsible for performing system upgrades and configurations. You have a lot of freedom to design, plan, and implement new VoIP system components for internal and external projects. You manage and implement change requests and are a specialist for special topics in the communication area. You show a high-sense of responsibility and foresight to keep systems running 24/7 You are prepared to work occasionally also in the evening hours to roll out new features Your skills You have several years experience in the day-to-day operation of telecom systems You have worked extensively with opensource VoIP software such as Kamailio / OpenSIPS, FreeSWITCH You can administer Linux systems such as RockyLinux and Debian You know SIP, RTP, RTCP, TLS, WebRTC protocols and can debug problems with Wireshark You have also worked with MySQL, Git You are communicative and like to work with team colleagues Further beneficial skills Experience with Ansible or Puppet for automation and configuration management. CheckMK, Homer, Hepic, Redis, CGRates, CDR Tools, Python Own VoIP software projects German language skills Please apply here: https://vier.jobs.personio.de/job/1638419?language=en&display=en

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RTJSON: No Json for routing
by felipe.nunes＠comuniki.me 24 Jul '24

24 Jul '24

Good afternoon I have the following error in the application: [rtjson_routing.c:701]: rtjson_update_branch(): no json for routing This error occurs regardless of whether there is high demand or not and the application sending the data correctly. Can you help me with this? Felipe Nunes

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ds_select from xhttp:request
by Christian Berger 24 Jul '24

24 Jul '24

Hi, we want to execute ds_select from an xhttp:request event route. The idea is to be able to get all active targets of a dispatcher set so our monitoring can check that via http which would be convenient for us. So are there any reasons why ds_select wouldn't work in an xhttp:request even route? Regards Christian Berger -- Christian Berger - berger(a)sipgate.de Telefon: +49 (0)211-63 55 55-0 Telefax: +49 (0)211-63 55 55-22 sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois Steuernummer: 106/5724/7147, Umsatzsteuer-ID: DE219349391 www.sipgate.de - www.sipgate.co.uk

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Listening on multiple TLS ports
by João Silva 23 Jul '24

23 Jul '24

Hello everyone! Is there a way to define a listen port range for TCP/TLS? Or any macro I can use to create the LISTEN commands? And finally is there a performance penalty for having multiple (2k) listening ports at the same time? I am in a strange situation where I need to register multiple credentials on a single SIP trunk, for each registration I need to use a different local port otherwise the trunk will simply overwrite the previous registrations. Furthermore, when I place a call on that trunk I have to be consistent with the port I used for registration for that specific credential. So I am looking into explicitly defining the port to be used per credential, UACREG allows me to use the contact_addr/socket for the registration part and for placing of outbound calls I can use $fsn to force the correct socket. Best regards, Joao

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IMS PCSCF force source IP address
by Anthony Blandin 23 Jul '24

23 Jul '24

Hello, I am using Kamailio pcscf with ipsec. I have one ipsec listen address which is used by the UE to connect to the PCSCF and one other listen address on port 5060. During the registration until the ipsec is not established, the UE sends the requests to the ipsec listen address and the pcscf is using the other listen address to forward the sip request to the ICSCF. That's is ok for me. But when the ipsec is established after the 200 OK REGISTER, the UE sends a SUBSCRIBE but the pcscf forwards the SUBSCRIBE to the SCSCF using its ipsec listen address instead of the other listen address. I tried to use $fs to force the socket but without success. Do you know how to force the source IP address to be used on the pcscf? Thanks for your support. Regards, Anthony

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mysql error after migration from 5.7.5 to 5.7.6
by Alejandro Guralnik 23 Jul '24

23 Jul '24

Hello, After upgraded from kamailio 5.7.5 to 5.7.6, when did start kamailio process, got the following error: WARNING: db_mysql [km_my_con.c:179]: db_mysql_new_connection(): opt_ssl_ca option not supported by mysql version (value (null)) - ignoring After downgrade to 5.7.5 dind't happened. (we don't use mysql ssl connection).

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Best Practices for Scaling Kamailio in a Large VoIP Deployment
by Maria Jonas 23 Jul '24

23 Jul '24

Hello Kamailio Community, I am currently working on a large scale VoIP deployment and would love to get insights on the best practices for scaling Kamailio. Our current setup handles a moderate volume of calls,,, but we’re expecting a significant increase in traffic soon. Specifically.., I am looking for advice on: -Optimizing Kamailio’s configuration for high availability and performance. -Efficiently managing load balancing and failover. -Recommended hardware specifications for large-scale operations. -Common pitfalls to avoid when scaling up. Any tips or resources you could share would be greatly appreciated. If you have experience with similar projects,,, I’d love to hear about your setups and any challenges you faced. Thank you! Maria https://www.igmguru.com/cloud-computing/aws-developer-certification-trainin…

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Best Practices for Scaling Kamailio in a Large VoIP Deployment
by Maria Jonas 23 Jul '24

23 Jul '24

Hello Kamailio Community, I am currently working on a large scale VoIP deployment and would love to get insights on the best practices for scaling Kamailio. Our current setup handles a moderate volume of calls,,, but we’re expecting a significant increase in traffic soon. Specifically.., I am looking for advice on: -Optimizing Kamailio’s configuration for high availability and performance. -Efficiently managing load balancing and failover. -Recommended hardware specifications for large-scale operations. -Common pitfalls to avoid when scaling up. Any tips or resources you could share would be greatly appreciated. If you have experience with similar projects,,, I’d love to hear about your setups and any challenges you faced. Thank you! Maria <a href="https://www.igmguru.com/cloud-computing/aws-developer-certification-trainin…">aws</a>

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SHM Memory leak?
by Ihor Olkhovskyi 23 Jul '24

23 Jul '24

Hello! I'm trying to get where all the SHM memory gone. For the moment I'm using these settings: -m 8192 -M 1024 which means 8G of SHM meory. Kamailio parameters are the following fork=yes children=8 tcp_children=12 enable_tls=yes enable_sctp=no tls_threads_mode=2 tcp_accept_no_cl=yes tcp_max_connections=63536 tls_max_connections=63536 tcp_accept_aliases=no tcp_async=yes tcp_connect_timeout=10 tcp_conn_wq_max=63536 tcp_crlf_ping=yes tcp_delayed_ack=yes tcp_fd_cache=yes tcp_keepalive=yes tcp_keepcnt=3 tcp_keepidle=30 tcp_keepintvl=10 tcp_linger2=30 tcp_rd_buf_size=80000 tcp_send_timeout=10 tcp_wq_blk_size=2100 tcp_wq_max=10485760 open_files_limit=63536 And having ~1000 TLS/WSS clients I'm getting this after 3 days of running: { "jsonrpc": "2.0", "result": [ "shmem:fragments = 2796", "shmem:free_size = 3737947072", (3.7 G) "shmem:max_used_size = 4857418512", (4.8 G) "shmem:real_used_size = 4851987520", "shmem:total_size = 8589934592", "shmem:used_size = 4838988096" (4.8 G) ], "id": 984479 } Means half of SHM memory is gone When running kamcmd mod.stats all shm I'm getting (output truncated for the largest values) Module: core { ... tcpconn_new(1201): 130669280 (0.13 G) ... Total: 131657632 (0.13 G) } Module: sl { ... Total: 7520 } Module: siptrace { ... Total: 13520 } Module: rr { Total: 0 } Module: dialog { ... Total: 146080 } Module: permissions { ... Total: 62144 } Module: htable { .... Total: 3359552 (3.3 M) } Module: rtpengine { ... Total: 31456 } Module: textopsx { Total: 0 } Module: tsilo { ... Total: 75072 } Module: tm { .... Total: 3459504 (3.4 M) } Module: usrloc { .... Total: 1217616 } Module: pua_dialoginfo { ... Total: 8720 } Module: pua { ... Total: 150848 } Module: pike { ... Total: 9504 } Module: websocket { wsconn_add(198): 54139856 (54 M) ... Total: 54148096 (54 M) } Module: debugger { ... Total: 21872 } Module: tmx { Total: 512 } Module: kex { .... Total: 1888 } Module: tls { ... ser_malloc(364): 87246112 (87 M) ... Total: 87997168 (87 M) } Module: secfilter { ... Total: 768 } Module: exec { ... Total: 16 } Module: dispatcher { ... Total: 2992 } Module: cfgutils { ... Total: 48 } Module: app_python3 { ... Total: 32 } So, I'm guessing, where all the memory gone? Is there any way to get more detailed info on SHM? Kamailio 5.8.2 -- Thanks in advance, Ihor

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Not responding to OPTIONS if existing tcp connection
by Jon Bonilla (Manwe) 22 Jul '24

22 Jul '24

Hi all version 5.7.6 Debian 12 I'm having issues with a kamailio server not sending 200 to some OPTIONS requests. It does send 200 to some subscribers but not 100% of the time to 100% of subs. I have enabled debug=3 in that server and this is what I found: If the log says that it found existing tcp connection and it's reusing it I can't see the 200ok in a sngrep capture. Example: <core> [core/receive.c:126]: sip_check_fline(): first line indicates a SIP reply <core> [core/tcp_main.c:1722]: _tcpconn_find(): found connection by id: 2 <core> [core/tcp_main.c:2614]: tcpconn_send_put(): send from reader (360359 (18)), reusing fd <core> [core/tcp_main.c:2845]: tcpconn_do_send(): sending... <core> [core/tcp_main.c:2881]: tcpconn_do_send(): after real write: c= 0x7fb0faa893d8 n=384 fd=14 <core> [core/tcp_main.c:2882]: tcpconn_do_send(): buf=#012SIP/2.0 200 Keepalive#015#012Via: SIP/2.0/TCP 10.35.190.105:5060;branch=z9hG4bK15ebf764;rport=55894;received=1.1.1.186#015#012From: "Unknown" <sip:contesta01@10.35.190.105>;tag=as5311c3a7#015#012To: <sip: 1.1.1.138>;tag=054524a74bcccf444a0f08f67c5b7657.d07b5401#015#012Call-ID: 15dcd38c46f915576ded132a11d8ff4c@10.35.190.105:5060#015#012CSeq: 102 OPTIONS#015#012Server: aaa SBC#015#012Content-Length: 0#015#012#015#012 <core> [core/receive.c:531]: receive_msg(): request-route executed in: 302 usec On the other side, if kamailio doesn't find an existing tcp connection (at least nothing in the logs about that) I can see the 200 ok in sngrep. Example: <core> [core/receive.c:126]: sip_check_fline(): first line indicates a SIP reply <core> [core/receive.c:531]: receive_msg(): request-route executed in: 295 usec All subs use tcp. And the kamailio.cfg is very simple regarding OPTIONS: Sanity, check maxfwd and sl_send_reply(200). Any hints? cheers, Jon -- PekePBX, the multitenant PBX solution https://pekepbx.com

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Drouting using TLS and domain names
by SAMUEL MOYA TINOCO 19 Jul '24

19 Jul '24

Hi everyone, I'm trying to configure drouting module but i'm not sure how to configure dr_gateways table to use domain names and TLS instead of IPs How should I do it? In dispatcher module is by using sips:domain but when I configure it like this I have problems to send invites (maybe I'm doing something wrong). Does anyone know how to configure it? Thanks in advance for your help Samuel Moya Tinoco Departamento de Sistemas y Redes Móvil: (+34) 606985997 smoya(a)vivelibre.es<mailto:smoya@vivelibre.es> [cid:image001.png@01DAD156.D8CFF170] Soluciones inteligentes para la autonomía personal

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Possible Kamailio memmory leak - version 5.8.2
by Vanderlei Torres Batistela 18 Jul '24

18 Jul '24

Hello all, I am using a very simple scritp with corex module enabled . Normal calls works fine, but a scenario where INVITE does not receive response and the call is released with SIP 408 (request timeout) leads to memmory leak. I verified it making a lot of calls with this scenario and printing the shared memory (using kamcmd core.shmmem) . Its ocurred with modparam("corex", "nio_intercept", 1). If I modify to modparam("corex", "nio_intercept", 0) the memmory leak does not ocurr anymore. The problem also occurs in another scenario where the SIP Cancel message is not responded to. My script has this event route # event route event_route[network:msg] { xlog("L_INFO", "Dentro do Event_Route \n"); if (is_incoming()) { xlog("L_INFO", "Received message '$mb' \n"); $avp(msg) = $mb; xinfo("avp(msg) = $avp(msg)"); } else { xlog("L_INFO", "Sending message '$mb' \n"); $avp(msg) = $mb; }; } Test done wtih 4000 calls: [root@labcom070 kamailio]# kamctl stats shmem { "jsonrpc": "2.0", "result": [ "shmem:fragments = 900", "shmem:free_size = 6180368", "shmem:max_used_size = 66980272", "shmem:real_used_size = 60928496", "shmem:total_size = 67108864", "shmem:used_size = 54884944" ], "id": 393115 } [root@labcom070 kamailio]# kamcmd mod.stats all shm Module: core { create_avp(178): 52257504 counters_prefork_init(211): 53760 cfg_clone_str(132): 112 cfg_shmize(221): 832 main_loop(1381): 16 init_pt(104): 16 init_pt(103): 16 init_pt(102): 6224 cfg_register_ctx(47): 96 init_tcp(5197): 8192 init_tcp(5191): 32768 init_tcp(5182): 16 init_tcp(5175): 16 init_tcp(5167): 16 init_tcp(5161): 16 init_tcp(5149): 16 init_avps(92): 16 init_avps(91): 16 init_dst_blocklist(435): 16384 init_dst_blocklist(427): 16 timer_alloc(494): 96 init_dns_cache(368): 16 init_dns_cache(359): 16384 init_dns_cache(351): 16 init_dns_cache(343): 16 init_timer(264): 16 init_timer(263): 16384 init_timer(262): 16 init_timer(261): 16 init_timer(250): 16 init_timer(218): 16 init_timer(207): 278544 init_timer(206): 16 init_timer(194): 16 cfg_child_cb_new(832): 64 sr_cfg_init(371): 16 sr_cfg_init(364): 16 sr_cfg_init(356): 16 sr_cfg_init(344): 16 sr_cfg_init(332): 16 ksr_shutdown_phase_init(123): 16 rpc_hash_add(118): 16 qm_shm_lock_init(1463): 16 Total: 52687808 } I attached my kamailio script used and a log with one call using the command "kamcmd corex.shm_summary" with modparam("corex", "nio_intercept", 1) and modparam("corex", "nio_intercept", 0) Regards, Vanderlei

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Re: Difference betwwen TM functions in KEMI
by Duarte Rocha 18 Jul '24

18 Jul '24

Hello Alex, You're totally right, that was the issue. Thank you so much for the help. Helpful as always. Cheers, Duarte Rocha

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Difference betwwen TM functions in KEMI
by Duarte Rocha 17 Jul '24

17 Jul '24

Hello all, I'm using Kamailio with native script and KEMI in Javascript. I noticed that functions from the TM module can have different results if I call them on KEMI or Native script. Is this intended behaviour? Don't they share the state? In order to test this i made the following code : KEMI Code : function ksr_request_route() { try{ if ( KSR.is_method("BYE") ) { print.error("BYE Request KEMI BYE") KSR.route("exists_test") if(KSR.tm.t_exists()) { print.error("KEMI : Transaction exists") } else { print.error("KEMI : Transaction does not exist") } Native Code : route[exists_test] { if( t_exists()) { xerr("Native Script : Transaction exists \n"); } else { xerr("Native Script : Transaction does not exist \n"); } } I've also tried to change function call order but every time I get "Transaction does not exist" on native and "Transaction Exists" on KEMI. Is this intended behaviour? Should I only call tm functions in native or Kemi in order to have consistency ? Thanks in advance, Best Regards,

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Dns record not alone?
by Pyry Aaltonen 17 Jul '24

17 Jul '24

Hi, I recently updated my testing env to 5.8.1 and after that I’ve been receiving dns_cache.c warning “record not alone”. I found out that it’s been added to quite recent commit: https://github.com/kamailio/kamailio/commit/d8a35b3b6c837b36779e232b65fce61… And that’s probably the why I haven’t seen it before. As far as I know, I have no other problem with dns or anything else than the warning appearing in the logs 😊 I can’t quite get what the warning is about? Clearly refers to dns queries, and with debug I found out that maybe its somehow related to ip which has several domains to pointing it (and queried by one of those domains by this kamailio). Can it be something with the PTR-record? And this warning comes ones every hour, so probably when some ttl expires, and meanwhile the cache is ok with the situation. And as a background, I haven’t done any specific dns-configurations in my config, so settings should be pretty much in defaults. Any advice what I should fix here to get the warning disappear? -Pyry

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Add a Header to CANCEL request
by sadik.oualla.mohamed＠gmail.com 17 Jul '24

17 Jul '24

Hello all, I am facing an issue with Kamailio when trying to add custom headers to CANCEL requests. My current configuration adds the "Reason" header perfectly: ``` if(!is_present_hf("Reason")){ $var(cause_request) = 31; append_hf("Reason: Q.850;cause=$var(cause_request)\r\n", "CSeq"); msg_apply_changes(); } ``` However, when I attempt to add headers like "X-Reason" or "TestHeader", they are not added: ``` if(!is_present_hf("Reason")){ $var(cause_request) = 31; append_hf("X-Reason: Q.850;cause=$var(cause_request)\r\n", "CSeq"); # or append_hf("TestHeader: Q.850;cause=$var(cause_request)\r\n", "CSeq"); msg_apply_changes(); #nothing changes in the relayed CANCEL request :-/ } ``` I've read in this discussion "https://lists.kamailio.org/mailman3/hyperkitty/list/sr-users@lists.kamailio…" that this might work if sending CANCEL in stateless mode, but I want to keep it stateful. Any suggestions on how to achieve this? Regards, Mohamed.

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Changing ksr_tcp_accept_iplimit from default 1024
by Who AmI 16 Jul '24

16 Jul '24

Hi list, I hope you are all well? I am currently loadtesting some services internally and I am getting an odd error around TCP connections from the same source IP below: "/usr/sbin/kamailio[1806789]: CRITICAL: <core> [core/tcp_main.c:4459]: handle_new_connect(): hit the limit of connections per source IP (my_ip:39043) - rejecting" Its from the core the error so I was looking at the cookbooks and documentation but cannot get around this specific limit or find a configuration to tweak that fixes it. The limit only affects TCP (will affect TLS as well but I haven't got that far) and is 1024 it seems. Kamailio version is: 5.7.6 OS: Ubuntu 22.04 Things I have done so far: I have set limitNOFILE=65535 in the service file Set the tcp/tls_max_connections at 8192 (default is 2048) Increase the tcp_rd_buf_size to 16k+ The service I am using to loadtest is SIPp with flags "-t tn" for TCP multisocket as this service will have much more than 1024 concurrent TCP connections to it from a single source IP. I think I have missed something basic here as the devel and the source code references: "ksr_tcp_accept_iplimit" here at line 541: https://github.com/kamailio/kamailio/blob/master/src/main.c#L541 I cannot find a way to change this with a configuration line it seems, am I being blind or is my only option to build from source and change this? Many Thanks, John.

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Weird telephone-event payload issue - need help
by Tim Bowyer 16 Jul '24

16 Jul '24

Hello Everyone, I've been grappling with this issue for some months, time to raise the white flag. Essentially I need to work out a way to configure Kamailio/RTPengine to normalise telephone-events to 101 earlier in the piece. 1. Call comes in from Carrier with INVITE w/SDP, telephone-event payload of 96. 2. INVITE sent to Asterisk with the same SDP, payload = 96 3. INVITE sent from Asterisk to Client with payload = 101 * 200 OK w/SDP from Client has payload = 101 * 200 OK w/SDP to Kamailio has payload = 96 * 200 OK w/SDP to Carrier has payload = 96 4. Asterisk (with Direct Media enabled) sends INVITE to Kamailio with payload = 96, and new media IP of Client 5. Kamailio passes this onto Carrier with media IP of RTPengine 6. 200 OK w/SDP from Carrier has payload = 96 sent to Kamailio 7. Kamailio passes this 200 OK to Asterisk, which responds with ACK and doesn't do anything further +---------+ +----------+ +---------+ +---------+ | Carrier | <----1-----> | Kamailio | <-----2----> | Asterisk| <----3------>| Client | | | | | | | | | +---------+ +----------+ +---------+ +---------+ ^ | | | | | |3| | | | | | +------5---------> +-----------+ <-----------------+------------4-----------+ | RTPengine | +-----------+ Big issue is, the Client is expecting telephone-events with a payload of 101, whereas RTPengine is passing it 96 from the Carrier. * Inbound DTMF only works at client-side if Asterisk stays in path (far from ideal). * The issue is exacerbated when calls are sent back out to the same or another carrier (forward scenario). * There's nothing I can seem to do in Asterisk to change how it handles DTMF in a direct media/re-invite setting. Any ideas? Cheers, Tim Bowyer

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how to upgrade kamailio minor release from 5.7.5 to 5.7.6
by sadik.oualla.mohamed＠gmail.com 15 Jul '24

15 Jul '24

I am currently using Kamailio version 5.7.5, which I installed from the source code. I would like to upgrade to version 5.7.6 but have been unable to find documentation specifically addressing minor upgrades between these minor releases. I have found only wikis detailing the process for upgrading for major releases (like 5.7.x to 5.8.0). Could you please point me to the documentation for my current need or share with me steps on how to perform this minor upgrade? Thank you. Cheers.

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How to get the rtpengine IP used by the current call among multiple rtpengines
by mm e 15 Jul '24

15 Jul '24

modparam("rtpengine", "rtpengine_sock","udp:192.168.1.3:12221=2 udp:192.168.1.1:12222=1 udp:192.168.1.5:12221=7") When rtpengine_sock is set to multiple rtproxies mode, how to get the id or ip of the rtpengine used in the current call? There is no information returned which rtpengine

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How to get the id or ip in the multiple rtproxies for LB with weights。
by mm e 15 Jul '24

15 Jul '24

modparam("rtpengine", "rtpengine_sock","udp:192.168.1.3:12221=2 udp:192.168.1.1:12222=1 udp:192.168.1.5:12221=7") When rtpengine_sock is set to multiple rtproxies mode, how to get the id or ip of the rtpengine used in the current call? There is no information returned which rtpengine kamailio selected for the call.

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Re: (no subject)
by Sergio Charrua 12 Jul '24

12 Jul '24

Hi Brett! From what I could understand, my client (a Carrier) assigns services and products to its customers identified by trunkgroups (depending on the customer and the service requested, as well as the ingress trunkgroup the call will be forwarded to a specific egress trunkgroup). . The INVITE is sent by customer to the SBC (ingress), where some kind of process (which i have no details) will decide what IP:Port to forward the messages to Kamailio with additional infos (Kamailio is listening on different NICs/IPs/Ports). Then, Kamailio will check on the (external) routing engine the route to take and forwards to the outbound endpoint. Depending on the IP:Port that Kamailio receives the call (SBC may choose one NIC or the other) the request is then received by Kamailio and verified/processed by the routing engine replying back to kamailio with the route to take, and then Kamailio forwards to the outbound endpoint (egress SBC...). It is a bit confusing, and I want to simplify things as much as possible, but apparently the old/current solution is set this way and they pretend to keep the new solution as similar as possible to the old/current solution. This is mostly due to the billing process, which is very (overly?) complex and they do not pretend to change it. IMHO, the routing engine should be much simpler but it is a much more complex process the Carrier currently has...and wishes to keep. (sorry for posting without a subject. I did, however, re-posted with corrected subject) Atenciosamente / Kind Regards / Cordialement, *Sérgio Charrua* On Fri, Jul 12, 2024 at 8:11 PM Brett Nemeroff <brett(a)voicefoxtelephony.com> wrote: > Is there a reason you have to identify the trunk group based on received > IP/Port? I've seen this as a requirement from some more antiquated systems > and it's always a pain. It's always better to use source IP or even a trunk > prefix (dialed number prefix) which isn't really secure, but works when > security isn't an issue. > > If you have to do it that way, I'd probably turn the $Ri/$Rp into some > sort of cache key like $Ri_$Rp and map it to something like a dispatcher > setid. > > There may be a more modern way to do this, but I don't think there is. > > Good luck! > -Brett > > > On Fri, Jul 12, 2024 at 1:40 PM Sergio Charrua via sr-users < > sr-users(a)lists.kamailio.org> wrote: > >> Hi all! >> >> for a project under development, one requirement is to be able to >> forward/relay the call to a specific destination depending on which IP >> Address and Port the call was received by Kamailio. >> This also means that Kamailio will be listening on multiple IP Address >> and Ports >> listen=udp:IP_1:port_1 # trunkgroup 1 >> listen=udp:IP_2:port_2 # trunkgroup 2 >> listen=udp:IP_3:port_3 # trunkgroup 3 >> >> I know the pv $Ri and $Rp can be used to identify from which IP address >> and port the message reached Kamailio and having those details, depending >> on the value, I can define some conditions that allows Kamailio to relay >> the call to different destination endpoints. >> For example: >> - calls from +32 to + 39 received on trunkgroup1 needs to go to Italy SBC >> address A.B.C.D >> - but calls from +32 to +39 received on trunkgroup2 need to go to Germany >> on SBC address 1.2.3.4 (as it is cheaper) >> >> Is there a better way of doing this without using IF/THEN/ELSE or SWITCH >> blocks and "hard code" destination SBC addresses? >> I have read the DRouting module's documentation but not sure if it could >> help and how... >> >> Any suggestions? >> >> >> >> Atenciosamente / Kind Regards / Cordialement, >> >> >> *Sérgio * >> __________________________________________________________ >> Kamailio - Users Mailing List - Non Commercial Discussions >> To unsubscribe send an email to sr-users-leave(a)lists.kamailio.org >> Important: keep the mailing list in the recipients, do not reply only to >> the sender! >> Edit mailing list options or unsubscribe: >> > > > -- > > ======================================================== > Brett Nemeroff > Voice Fox Telephony LLC > Office: 512-670-8369 > Email: brett(a)voicefoxtelephony.com >

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Change the To Header in the failure_route
by sadik.oualla.mohamed＠gmail.com 12 Jul '24

12 Jul '24

Hello, I need help with modifying the "To" header in the "failure_route[ID]{...}". Specifically, I want to add the "to tag" parameter from the previous SIP messages of the current dialog(e.g., 180 Ringing, PRACK or final responses). Here's the scenario: 1. Kamailio receives and relays the SIP messages: INVITE, 180 Ringing, PRACK, 200 OK, 486 Busy Here. 2. In my case, the 180 Ringing includes the "to tag" parameter. 3. When Kamailio receives the 486 response, it triggers "failure_route[ID]" using "t_on_failure(ID)" method. 4. I extract the SIP response code and reason from the final response of the INVITE transcation in this failure_route. 5. After processing and applying some instructions, I send a reply using "t_send_reply" with the code and reason extracted. The problem: - The failure_route[ID] holds the initial INVITE, which has no "to tag" parameter. - When I send a reply using "t_send_reply()", Kamailio generates a random "to tag" different from the previous SIP messages, which confuses my UAC. I think adding the "to tag" parameter manually might be a solution, but I'm not sure if I'm doing it correctly or if there is a better approach. I also tried using uac_replace_to() method, but it doesn't work in the failure_route[ID]. How can I make sure the reply in failure_route[ID] uses the correct "to tag" from the previous SIP messages? Are there any other possibilities to achieve this? Any assistance would be appreciated. Thank you!

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