sr-users July 2024

sr-users@lists.kamailio.org

42 participants
68 discussions

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

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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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Kemi Python3 throws a TypeError
by Michel Pelletier 12 Jul '24

12 Jul '24

Hello, Please help. I am using Kemi with the python3 module. I get random TypeError exceptions thrown within the Kamailio code. For example I have the following code: event = str(KSR.pv.get("$hdr(Event)")) log.logInfo( "KamailioStateful.ksr_request_route_loopback(): " + "Passing the request to the presence " + "publish handler for event: " + event ) rc = KSR.presence.handle_publish() if rc != 1: log.logError( "KamailioStateful.ksr_request_route_loopback(): " + "The presence publish handler returned " + "an error code: " + str(rc) ) This is what I see in the logs: 21(120) ERROR: PY3 {PUBLISH}: app_python3 [python_support.c:167]: python_handle_exception(): apy_exec: ksr_request_route((null)): Unhandled exception in the Python code: TypeError: expected bytes, str found The above exception was the direct cause of the following exception: Traceback (most recent call last): File "/etc/kamailio/kamailio.py", line 91, in ksr_request_route return ksf.ksr_request_route(msg, "Loopback") ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ File "/etc/kamailio/kamailio_stateful.py", line 100, in ksr_request_route return self.ksr_request_route_loopback(msg) ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ File "/etc/kamailio/kamailio_stateful.py", line 240, in ksr_request_route_loopback rc = KSR.presence.handle_publish() ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ SystemError: <built-in function handle_publish> returned a result with an exception set If I put the offending line within a Try/Except it hides it but I would rather not see it not happen at all. Any Kemi Python3 users experiencing something similar? What could be causing this? Regards, Michel Pelletier

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routing call depending on which IP address SIP message reached Kamailio
by Sergio Charrua 12 Jul '24

12 Jul '24

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 *

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sr-users July 2024