[SR-Users] Re: Kamailio not listening to SIP despite setup OK

s / Cordialement / Un saludo, *Sérgio Charrua* On Thu, Sep 19, 2024 at 9:29 AM Sergio Charrua wrote: > Thank you both for your replies. > Client followed some of mine guidelines and has proven that the issue is > with the Fortinet F5 firewall: he made a snapshot of the kamailio node

[SR-Users] Re: Kamailio not listening to SIP despite setup OK

nce with Cisco and Fortinet stuff I know that you need > to tweak the Fortnite (SIP ALG, SIP POLICY and so one). > > Go in that direction cause it works with Cisco before and the Fortigate > seems to break it. > > Kind regards > > Sergio Charrua via sr-users schrieb am Mi.,

[SR-Users] Kamailio not listening to SIP despite setup OK

Hi all! got a client with 2 Kamailio servers using Pacemaker/Corosync with a VIP. Kamailio is setup with dispatcher, and has been working since 2018. This previous week, client migrated from a Cisco ASA firewall to a Fortinet F5 firewall. One of the Kamailio server is, apparently, working fine, r

[SR-Users] Permissions module - automatic reload

Hi all, Just need some clarifications about the Permissions module: I currently use it with db_mode = 1 and was expecting that the parameter "trusted_cleanup_interval" would refresh the cache tables every X seconds, but apparently this is not the case. I was looking for a way for Kamailio to autom

[SR-Users] Re: Kamailio integration with REST services

be) more flexible in the long term Atenciosamente / Kind Regards / Cordialement / Un saludo, *Sérgio Charrua* On Tue, Aug 27, 2024 at 1:08 PM Fred Posner via sr-users < sr-users@lists.kamailio.org> wrote: > > > On Aug 27, 2024, at 6:58 AM, Sergio Charrua via sr-

[SR-Users] Re: Kamailio integration with REST services

automatic 24h expiry, just two options. The DB > could be used from multiple Kamailio servers, htable would be individual > per server. > > > > Cheers, > > > > Henning > > > > -- > > Henning Westerholt – https://skalatan.de/blog/ > > Kamailio servi

[SR-Users] Kamailio integration with REST services

Hi all! I need to integrate a REST Service with Kamailio. Integration is easy, not a problem, but my concern is that the REST Services has a JWT token authentication required for each request, common to many (it not all) REST Services. So, it is required to POST for authentication (username and pa

[SR-Users] Re: kamailio randomly crashes

Hi! just a long shot, but I see a lot of these errors in the dump you shared: error: Cannot access memory at address wondering if the fact you're using Kamailio and Kubernetes is related to the issue Did you try Kamailio running as a standard linux service, without kubernetes? Sérgio On Wed

[SR-Users] Re: RTJSON: No Json for routing

Hi Felipe, could you please share your code (or the most important part) and the RTJSON object/response ? Atenciosamente / Kind Regards / Cordialement, *Sérgio Charrua* On Wed, Jul 24, 2024 at 5:34 PM felipe.nunes--- via sr-users < sr-users@lists.kamailio.org> wrote: > Good afternoon > > I ha

[SR-Users] Re: (no subject)

t; 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@li

[SR-Users] (no subject)

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

[SR-Users] Re: routing call depending on which IP address SIP message reached Kamailio

to use different carrier > per trunk group. > > > > Henning > > > > *From:* Henning Westerholt > *Sent:* Friday, July 12, 2024 1:49 PM > *To:* Kamailio (SER) - Users Mailing List > *Cc:* Sergio Charrua > *Subject:* RE: [SR-Users] routing call depending on wh

[SR-Users] routing call depending on which IP address SIP message reached Kamailio

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

[SR-Users] HTTP Client configuration path

Hi all! I am using HTTP_CLIENT module, and have created a http configuration file. The module's config_file parameter is set as: modparam("http_client", "config_file", "/etc/kamailio/http_client.cfg") and it works OK on my DEV environment, because there is a symlink /etc/kamailio pointing to ph

[SR-Users] StriShaken module compilation

Hello all! I am struggling on compiling StriShaken module on RHEL 9.2. So far, the module was compiled as well as libstirshaken. Below are the steps used to compile (as sudo) both library and module (I hope this may help someone) and, of course, correct me if I am wrong in any step or if there is

[SR-Users] Re: Is there any funtion to route based on source phone number

Hi all! I guess you could use dispatcher group ID and depending on the invite phone number, use dispatcher group ID X *Sérgio Charrua* *www.voip.pt * Tel.: +351 91 631 11 44 Email : *sergio.char...@voip.pt * This message and any files or documents attached are str

[SR-Users] Re: A question about Db_flatstore Module in Kamailio

Hi Farzaneh, I have a similar script doing log rotation too, and after executing kamcmd, you may remove the old files (*.temp in your case). In my case, instead of deleting file, I move them to another folder (to be processed by the CDR processor) right after executing kamcmd, who will create new

[SR-Users] making HTTP Async request in stateless configuration

Hi all! Just wondering what would be you opinion on the following. We are integrating Kamailio as a stateless redirect server, using SIP 300 Multiple Choices response. The script requires that prior to the redirection, a HTTP request must be made. We thought about using HTTP Async Client to make

[SR-Users] Re: base64 decoding issue

Hi all! Solved the issue by using s.decode.base64t instead of s.decode.base64 ... Thanks anyway! *Sérgio Charrua* On Fri, May 10, 2024 at 9:40 AM Sergio Charrua wrote: > Hi all! > > I have been dealing with STIR/SHAKEN for a few weeks now, and while doing > some tests I have fo

[SR-Users] base64 decoding issue

Hi all! I have been dealing with STIR/SHAKEN for a few weeks now, and while doing some tests I have found an issue which I can't find the reason for. The script has the following route logic: route[HANDLE_STIRSHAKEN] { xlog("L_INFO", "HANDLE_STIRSHAKEN - STIR/SHAKEN Logic"); if ($hdrc(

[SR-Users] Re: Retransmission behaviour after ACK

ywords.html#next-url) > > > > > > Kaufman > > > > > > *From:* Sergio Charrua via sr-users > *Sent:* Monday, May 6, 2024 6:57 AM > *To:* Kamailio (SER) - Users Mailing List > *Cc:* Sergio Charrua > *Subject:* [SR-Users] Retransmission behaviour after A

[SR-Users] Retransmission behaviour after ACK

Hi all! I'm testing some redirect scenarios with Kamailio 5.7.4 and using Astterisk 13 and SIPp for unit testing. The kamailio's script is simple and uses TM, HTTP_ASYNC_MODULE, JSONRPC, JANSSON, RTJSON and XLog modules. Request route is simple: request_route { route(HANDLE_OPTIONS); route(

[SR-Users] Re: Kamailio works but voice is not present during the calls!

ease inform the sender immediately, delete it from your system and do not copy or disclose it or its contents or use it for any purpose. Please also note that transmission cannot be guaranteed to be secure or error-free. On Wed, May 1, 2024 at 8:49 AM christian.marinelli--- via sr-

[SR-Users] Re: Kamailio works but voice is not present during the calls!

e usage: > > > > rtpengine.conf > > > > ### for userspace forwarding only: > > table = -1 > > > > > > Mit freundlichen Grüssen > > > > -Benoît Panizzon- > > Hi @Benoit Panizzon and @Sergio Charrua, > i set up rtpengine on m

[SR-Users] Re: Kamailio works but voice is not present during the calls!

04-22-04/> As for question #2, certainly most of those 700Mb are related to outdated packages that need some updates. *Sérgio Charrua* On Fri, Apr 26, 2024 at 5:42 PM christian.marinelli--- via sr-users < sr-users@lists.kamailio.org> wrote: > Sergio Charrua wrote: > > The be

[SR-Users] Re: Kamailio works but voice is not present during the calls!

icult for a neophyte user with > SIP concepts. > > So i prefer the second one! > > Can you suggest me how i can change my Kamailio configuration to do this > behavior? > > Thanks you so much in advance > > Christian > > Hi @sergio charrua, > the last days i t

[SR-Users] Re: Kamailio works but voice is not present during the calls!

Benoit is right! Your issue is a common NAT issue: your SIP signalling is sending public IP address on headers, while the SDP content is publishing internal/private IP addresses: [image: image.png] Possible solutions: 1 - Use RTPEngine and integrate it with Kamailio 2 - let your Kamailio instanc

[SR-Users] Re: Kamailio works but voice is not present during the calls!

Christian, instead of sharing screenshots, please share the SIP PCAP (use tshark/wireshark or sngrep) file with SIP signaling, it will ease the analysis and get you better help. I can't understand why messages are repeated exactly 3 times. Retransmissions?! *Sérgio Charrua* On Mon, Apr 22, 202

[SR-Users] Re: making HTTP requests in stateless redirects

ashov via sr-users < sr-users@lists.kamailio.org> wrote: > > > > On Apr 8, 2024, at 11:12 AM, Sergio Charrua via sr-users < > sr-users@lists.kamailio.org> wrote: > > > > Also, as this is a stateless script, is there another way of using async > http or makin

[SR-Users] Re: making HTTP requests in stateless redirects

quest *Sérgio Charrua* On Mon, Apr 8, 2024 at 6:20 PM Nick Digalakis wrote: > I didn't test this, but why are you calling t_newtran(); if you only want > to send the 300 response? > > On Apr 8, 2024 18:12, Sergio Charrua via sr-users < > sr-users@lists.kamailio.org&g

[SR-Users] making HTTP requests in stateless redirects

Hi all! For testing purposes, while I am waiting for the ST/SH REST API to be available from other teams, I developed a small python REST API that returns a mockup of a JSON object with some of the required values. The kamailio script, completely stateless as I do not need to keep track of the se

[SR-Users] Re: low performance with no apparent reason

s this > is per process. This much is never needed. > > On the other hand, you should probably increase the shared memory if you > are having a lot of transactions going on, TLS etc.. This is per server, so > you can configure more. > > > > Cheers, > > > > Hennin

[SR-Users] Re: low performance with no apparent reason

Thank you all for helping! I wasn't expecting such a large number of replies! I ended up partially solving the issue with a different approach. Modifying the size of the UDP Buffer did not reveal any improvement. However, modifying the memory management did improve a lot: from 330 CPS to 1800 CPS

[SR-Users] Re: low performance with no apparent reason

amailio.org> wrote: > > > > > > I assume that you are using udp. > > > Please increase the length of the udp queue: > > > > https://medium.com/@CameronSparr/increase-os-udp-buffers-to-improve-performance-51d167bb1360 > > > > > > Regards. > >

[SR-Users] Re: low performance with no apparent reason

lists.kamailio.org> wrote: > >> I assume that you are using udp. >> Please increase the length of the udp queue: >> >> https://medium.com/@CameronSparr/increase-os-udp-buffers-to-improve-performance-51d167bb1360 >> >> Regards. >> Ovidiu Sas >> &

[SR-Users] low performance with no apparent reason

Hi all! I have been doing some performance tests with Kamailio 5.7.4 and SIPp. The infrastructure is as follows:3 VMs running on VMWare ESXi running: UAC on 10.20.0.1 with SIPP-> Kamailio on 10.20.0.5 -> UAS on 10.20.0.3 The Kamailio VM has 6 dedicated vCPU of type Intel(R) Xeon(R) Silver 4216 CP

[SR-Users] Re: direct media between UACs

th \r\n symbol because it will be added at the end of the SIP Header, which requires a blank line to separate from the SIP Body. Hope this helps someone out there! Cheers! *Sérgio Charrua* On Thu, Mar 7, 2024 at 3:44 PM Sergio Charrua wrote: > Hi all! > > some additional details

[SR-Users] Re: direct media between UACs

s *not* within endpoints, but somewhere in Kamailio (module or configuration). Any suggestions? *Sérgio Charrua* On Tue, Mar 5, 2024 at 1:05 PM Sergio Charrua wrote: > Hi all! > > got a weird behavior that I cannot understand the reason for... > In our LAB environment, we have 2

[SR-Users] direct media between UACs

Hi all! got a weird behavior that I cannot understand the reason for... In our LAB environment, we have 2 Asterisk instances (version 13.38.3 and chan_sip) and 1 Kamailio 5.7 in between. All servers are in the same network, so, there is no NAT involved. No RTPEngine either. Network is 10.20.0.0/24

[SR-Users] Re: DMQ module failure

<http://www.voip.pt/>* On Fri, Mar 1, 2024 at 3:52 PM Fred Posner wrote: > > On Mar 1, 2024, at 10:30 AM, Sergio Charrua via sr-users < > sr-users@lists.kamailio.org> wrote: > > > > Hi all! > > > > I'm testing the DMQ module, but having difficultie

[SR-Users] DMQ module failure

Hi all! I'm testing the DMQ module, but having difficulties setting it up. [...] My kamailio.cfg is as follows : [...] listen=udp:0.0.0.0:5060 listen=tcp:0.0.0.0:5060 [...] modparam("dmq", "server_address", "sip:10.20.0.2:5062") #DMQ_SERVER_ADDRESS modparam("dmq", "notification_address", "sip:10

[SR-Users] Re: best approach to setup SIP Session sharing between servers

lashov via sr-users < > sr-users@lists.kamailio.org> wrote: > > > > That would require transaction replication, which Kamailio doesn't have. > > > > Most messages can be processed statelessly, so this isn't a huge > obstacle. However, CANCELs won

[SR-Users] best approach to setup SIP Session sharing between servers

Hi all! I am having some difficulties/doubts on what would be the best approach to let multiple Kamailio servers share their SIP sessions between each other. The goal is to have HA on Kamailio cluster such that if a call is received and initiated on Kamailio #1, and during any moment of the call (

[SR-Users] Re: Setting kamailio to use another DB

Hi! I prefer to let Kamailio do what it does best (SIP) and let other tools manage whatever you need to do additionally. In your case, I would use a Keepalived service, shared between the 2 database nodes (or any other number of nodes) and create a Virtual IP (VIP) shared between DB nodes. By defa

[SR-Users] Re: Kamailio as SIP proxy for an Asterisk with Private IP Address

As far as i understand, Kamailio is probably in a Datacenter somewhere and the Asterisk box is in an Office without any public IP address forwarding data directly to the Asterisk. The best solution I can think of (in terms of security and ease of use) is using a VPN between Kamailio and your clien

[SR-Users] Re: Newbie question

Joao Paulo, add a second Asterisk server, and add a Virtual IP (VIP) using Keepalived. Then UAC connects to Asterisk using the VIP. Keepalived will switch to the 2nd node once the 1st is down (or even if the 1st one stops Asterisk service). In this case, you may not need a SIP Proxy as frontend.

[SR-Users] Re: Kamailio Lagging or Unreachable in Asterisk Peer

Hi! assuming you're using Mysql, you can execute in Mysql CLI: > SHOW FULL PROCESSLIST\G This will list all queries running at that moment, and who is executing the query. Depending on the query, if you are using table locks, this is normal behaviour. But I do not think this is a Kamailio issue.

[SR-Users] NAT issues with Kamailio + Asterisk with 2 NICs

Hi all! Got a random issue with NAT which I really have no clue how to solve. Setup is as follows: - Client A sends SIP INVITE through Public IP 1.2.3.4, where a Cisco Firewall forwards NATed data to a Kamailio instance (acting as a SIPProxy with dispatcher) listening with internal address 192.16

Re: [SR-Users] Kamailio Stop processing SIP traffic

o you see any errors in the logs? >> >> -ovidiu >> >> >> On Wed, Dec 22, 2021 at 11:44 AM pwerspire wrote: >> > >> > The CPU is around 30% during the test, is always really stable and >> memory doesn't increase much. >> > >>

Re: [SR-Users] Kamailio Stop processing SIP traffic

get the prompt of the shell back and > after that, all kamcmd commands stop returning any output or giving back > the prompt. > > If the Kamailio is then restarted, everything starts working correctly > again. Also, another note is that this seems only to happen when DMQ is in &g

Re: [SR-Users] Kamailio Stop processing SIP traffic

What is the call length of each call? How many NICs you have on that server? what networking switches are you using? all these variables have huge impacts on the call flow! Let's say you make 100CPS and the call length is 10 minutes. At 10 seconds you will have 1.000 ongoing calls using g711 (64kb

Re: [SR-Users] List of Header Names

Google is your friend (it has been mine since 1997) https://telnyx.com/resources/the-beginners-guide-to-sip-headers https://www.iana.org/assignments/sip-parameters/sip-parameters.xhtml Cheers *Sérgio Charrua* *www.voip.pt * Tel.: +351 21 130 71 77 Email : *sergio.char

Re: [SR-Users] Kamailio 5.3.8 and RTPengine 9.4.1.1+0~mr9.4.1.1

Hi Mihai! RTPEngine should be set on the same server where Kamailio is running. These are the options I use in RTPEngine on a NATed server, /etc/sysconfig/rtpengine # Add extra options here OPTIONS="--interface eth1/10.19.139.66 --listen-ng 127.0.0.1:6 -m 1 -M 2 -L 7 --log-facility=lo

Re: [SR-Users] Critical : Kamilio is not starting when loading Dispatcher module .

Hi! I don't think it is related to dispatcher module. the following error says that something is listening port 5060 : ERROR: [core/tcp_main.c:2863]: tcp_init(): bind(d, 0x7f81495f1efc, 16) on 127.0.0.1:5060 : Address already in use Try to execute the following command: # netstat -anp | grep 506

Re: [SR-Users] Kamailio RTP proxy require?

ure or error-free. On Thu, Dec 17, 2020 at 2:32 PM Richard Fuchs wrote: > On 17/12/2020 09.00, Sergio Charrua wrote: > > Personally I use NATHelper and RTPEngine modules > > > > RTPEngine, afaik, is the latest and upgraded version of RTPProxy > > (please, anyone c

Re: [SR-Users] Kamailio RTP proxy require?

nects to the Asterisk LAN. > > @Sergey I tried that scenario with double NAT, and, it work it. > > If i choose to go rtpproxy which modules should i load (nat_helper, > nat_traversal, rtpproxy) ? > > > > On Thu, Dec 17, 2020 at 2:38 PM Sergio Charrua > wrote: > >>

Re: [SR-Users] Kamailio RTP proxy require?

Another example of when to use RTPProxy / RTPEngine is: SIP user -> router -> INTERNET -> router / firewall -> Kamailio (private IP) -> Asterisk (private IP) Apart from the routers, all other equipments are private IPs, hence being NATed and requiring RTP Proxy / RTP Engine to forward RTP streams

Re: [SR-Users] Kamilio+asterisk = deregister after 2-3 minute

I will assume that you use Kamailio as SIP Proxy between a SIP Provider (TelCo) and your Asterisk boxes (using SIP.conf), and that Kamailio is receiving OPTIONS messages from TelCo, and forwarding those OPTIONS to Asterisk who will fail to answer. This is a common implementation with TelCos.The sol

Re: [SR-Users] Dispatcher algorithm 11 possible priority drifting

I noticed the exact same behaviour with my implementation, even though the priority and rweight are different. until now, i'm ignoring the "issue". But if there is a fix, I would like to know too. *Sérgio Charrua* *www.voip.pt * Tel.: +351 21 130 71 77 Email : *sergio.char

Re: [SR-Users] Kamilio dispatcher not registering with asterisk

just set allow_guest = yes. I don't know what to set on the asterisk, >> if you can guide a little bit ... >> >> On Mon, Dec 7, 2020 at 5:17 PM Sergio Charrua >> wrote: >> >>> Hi ! >>> >>> what is the "type" of your SIP Peer, on

Re: [SR-Users] Kamilio dispatcher not registering with asterisk

Hi ! what is the "type" of your SIP Peer, on Asterisk? Friend, Peer or User? and what about the "insecure" setting? *Sérgio Charrua* *www.voip.pt * Tel.: +351 21 130 71 77 Email : *sergio.char...@voip.pt * This message and any files or documents attached are strictly con

Re: [SR-Users] Kamailio in VMware with Fault Tolerance on

step > towards a public cloud deployment. > > > On Tue, 3 Nov 2020, 21:45 Sergio Charrua, wrote: > >> Patrick, >> >> I would rely on Corosync and Pacemaker for failover, using a shared IP >> address between (active + passive) servers. Once the active goes do

Re: [SR-Users] Kamailio in VMware with Fault Tolerance on

Patrick, I would rely on Corosync and Pacemaker for failover, using a shared IP address between (active + passive) servers. Once the active goes down, Pacemaker switches the shared IP to the passive server that will then become the active server. Hope this helps, *Sérgio Charrua* *www.voip.pt <

Re: [SR-Users] Dispatcher stops working

d, Sep 16, 2020 at 5:55 PM David Villasmil < > david.villasmil.w...@gmail.com> wrote: > >> Hello >> >> I would start by looking at the logs when that happens. The gateway would >> only get marked "ip" when there is no reply by the gateway or when it >

[SR-Users] Dispatcher stops working

Hello, Using Kamailio 5.1.6 for load balancing & failover, I noticed that from time to time, and very rarely, the dispatcher module stops dispatching to the Asterisk servers. I have to execute a "kamcmd dispatcher reload" for it to (re)start working. This is my dispatcher.list : 1 sip:10.19.XXX.Y

Re: [SR-Users] Dispatcher does not go to 2nd server

r-slave setup and have users created in both. Am i correct? > > Στις Τετ, 22 Ιουλ 2020, 11:36 ο χρήστης Sergio Charrua < > sergio.char...@voip.pt> έγραψε: > >> Try to install SNGREP command line application. It greatly helped me to >> understand what was going on in kamai

Re: [SR-Users] Dispatcher does not go to 2nd server

Try to install SNGREP command line application. It greatly helped me to understand what was going on in kamailio. Install it in your Kamailio server, and run it. You will then be able to see and understand all SIP requests and responses. Also, as Daniel said, the SIP requests between client->kamai

Re: [SR-Users] RTP relay

Try to remove the public IP from your Asterisk Box, this will even provide additional security. Then install RTPEngine, configure the RTP Engine accordingly, setup NAT to your Kamailio configuration and try. the following URLs are very good info on how to integrate everything: http://www.evaristesy

Re: [SR-Users] Kamalio + RTP Engine + SIP Client behind NAT

Olá Nuno, I've similar setup, and everything works fine. I can share the setup, if you wish. Contact me directly. Cheers, *Sérgio Charrua* *www.voip.pt * Mobile: +351 91 631 11 44 Email : *sergio.char...@voip.pt * Esta mensagem e quaisquer ficheiros anexos a ela são conf

[SR-Users] Getting a new RTP Port on reINVITE during answered call

Hi all, I've the following infrastructure (fake IPs) 62.128.128.68 (Telco SBC Inbound) | | 10.19.139.66 (Kamailio + RTPEngine) | | 10.19.139.69 (Asterisk 13) | | 62.128.128.68 (Telco SBC Outbound) Test scenario is: - a mobile calls a destination; the call com

[SR-Users] new RTP Port after INVITE during call

Hi all, I've the following infrastructure 62.128.128.68 (Telco SBC Inbound) | | 10.19.139.66 (Kamailio + RTPEngine) | | 10.19.139.69 (Asterisk 13) | | 62.128.128.68 (Telco SBC Outbound) Test scenario is: - a mobile calls a destination; the call comes in from

[SR-Users] Kamailio not recognizing SIP requests

Hi all! got a Kamailio doing load balancing to multiple Asterisk boxes. Kamailio is connected to a carrier's SIP trunk, and distributes calls to Asterisk. Receiving calls from carrier's SIP trunk works perfectly: kamailio will use dispatcher module to distribute calls between Asterisk boxes. No p