- sr-users - kamailio.org

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by Jyoti bansal 23 Dec '21

23 Dec '21

2 1

Active/active keepalived - Kamailio includes non-local virtual IP in "myself" which breaks DMQ call routing
by Rhys Hanrahan 22 Dec '21

22 Dec '21

Hi Everyone, I have Kamailio acting as a registrar, using DMQ to replicate usrloc between nodes. I’m relaying INVITEs to the home registrar, but skipping this if (uri == myself) is true. In my lab I just setup my SBCs in an active/active mode, similar to the link below. So I have 2x VIPs and both SBCs listen on both, but the first VIP is only master on SBC01 and the second VIP is only master on SBC02. https://docs.nginx.com/nginx/admin-guide/high-availability/ha-keepalived-no… My issue is that now “myself” will always return true even if the virtual IP is actually not active on this node at the moment, so the invite is never relayed properly if it’s on another node. I guess “myself” is just blindly looking at the list of Ips that are configured with either listen= or alias= and doesn’t care if the IP actually exists on the server at the time, or not? I am thinking I may have to ditch relying on “myself” entirely and come up with some other way of checking for local requests, but I am wondering if there’s a good way of approaching this? I was thinking of perhaps triggering an add/remove of listen directives (or something) when a failover of a VIP happens, though this seems kind of heavy handed. Hoping there is a better way. Has anyone else tried doing active/active Kamailio instances like this? Thanks! Rhys Hanrahan | Chief Information Officer e: rhys(a)nexusone.com.au<mailto:rhys@nexusone.com.au> [www.nexusone.com.au]<http://www.nexusone.com.au/> [signature_132622389] <http://www.fusiontech.com.au/> NEXUS ONE | FUSION TECHNOLOGY SOLUTIONS p: 1800 NEXUS1 (1800 639 871) or 1800 565 845 | a: Suite 12.03 Level 12, 227 Elizabeth Street, Sydney NSW 2000 www.nexusone.com.au<http://www.nexusone.com.au/> | www.fusiontech.com.au<http://www.fusiontech.com.au/> The information in this email and any accompanying attachments may contain; a. Confidential information of Fusion Technology Solutions Pty Ltd, Nexus One Pty Ltd or third parties; b. Legally privileged information of Fusion Technology Solutions Pty Ltd, Nexus One Pty Ltd or third parties; and or c. Copyright material Fusion Technology Solutions Pty Ltd, Nexus One Pty Ltd or third parties. If you have received this email in error, please notify the sender immediately and delete this message. Fusion Technology Solutions Pty Ltd, Nexus One Pty Ltd does not accept any responsibility for loss or damage arising from the use or distribution of this email. Please consider the environment before printing this email.

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Issue Bug in SEMS (transparent SBC profile)
by Mojtaba 21 Dec '21

21 Dec '21

Hello, Why in SEMS, some changes (in transparent SBC profile) in sip headers do not apply. For example i want to change CSeq or Contact header like this: # transparent SBC profile # defaults: transparent RURI=$r From=$f To=$t Call-ID=$ci_leg2 CSeq=$H(CSeq) Contact=$H(Contact) When I run SEMS, CSeq and Contact do not change, while other headers change. [#7fc688d7b1c0/9379] [onLoad, SBC.cpp:189] INFO: loading SBC call profiles from '/usr/local/etc/sems/etc/' [#7fc688d7b1c0/9379] [readFromConfiguration, SBCCallProfile.cpp:418] INFO: SBC: loaded SBC profile 'mo' - MD5: d6a2c29084043edde1357231f6866737 [#7fc688d7b1c0/9379] [readFromConfiguration, SBCCallProfile.cpp:423] INFO: SBC: RURI = '$r' [#7fc688d7b1c0/9379] [readFromConfiguration, SBCCallProfile.cpp:424] INFO: SBC: RURI-host = '' [#7fc688d7b1c0/9379] [readFromConfiguration, SBCCallProfile.cpp:425] INFO: SBC: From = '$f' [#7fc688d7b1c0/9379] [readFromConfiguration, SBCCallProfile.cpp:426] INFO: SBC: To = '$t' [#7fc688d7b1c0/9379] [readFromConfiguration, SBCCallProfile.cpp:431] INFO: SBC: Call-ID = '$ci_leg2222' [#7fc688d7b1c0/9379] [readFromConfiguration, SBCCallProfile.cpp:434] INFO: SBC: force outbound proxy: yes This issue caused the Asterisk could not pass the Authentication process in "match_req_to_dialog" function because the CSeq is the same as the dialogs. So the Asterisk thought this message was a LOOP message. Has anybody encountered this issue in SEMS? How did you solve this? I know there are some ways for doing this with Kamailio, But i focused on issues in SEMS With Best Regards *-- Mojtaba Esfandiari.S* *-- PhD student and Research Affiliate, * *-- Technical Manager at IP-PBX Laboratory, Ferdowsi University of Mashhad, Iran.* -- Address: IP-PBX Lab., Engineering Faculty, Ferdowsi University Main Campus, Mashhad, Iran. -- Tel: +98-51-38763635 -- Mobile: +98-915-117-6713

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uac module - use htable instead of database
by Markus Bönke 20 Dec '21

20 Dec '21

Hello all, we want to use the "remote registrations feature" of the uac module to let kamailio register with other sip providers. As far as I see the module works only with a database backend. As our kamailio script should be independent from a database, a solution could be to load the configuration on startup (and refresh it periodically) from a database view into a htable and then trigger the remote registrations based on the expiration of the htable columns. Could something like this be implemented by using the uac module or by an other workaround ? Or does it require some enhancements in the uac module ? Thanks and regards Markus

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Using kamcmd tm.t_uac_start to send SIP REGISTER messages in Kamailio 5.5
by Steve Sether 20 Dec '21

20 Dec '21

I'm trying to use the kamcmd tml.tc_uac_start command to send raw SIP messages in Kamailio 5.5, but I always get a error: 400 - Invalid headers. I've also tried sending OPTIONS messages with the same result. I've looked for examples of sending SIP messages via kamcmd, but can't find any. kamcmd help tm.t_uac_start specifies the following: kamcmd> help tm.t_uac_start starts a tm uac using a list of string parameters: method, ruri, dst_uri, send_sock, headers (CRLF separated) and body (optional) Essentially my problem is I don't really know how to properly encode these headers. How do I send special characters? Do I need to use double quotes, single quotes, etc around the whole header? Which characters are special? How do I encode the CRLF? I tried \r\n, but I still get this error. It says only From: and To: are required. It's a bit hard trying to make progress when all I ever get is one error message. The kamailio logs are a little more helpful, and largely point me down the road that it doesn't understand the CRLF encoding, and says my From: header includes the To: line, so it's not separating them. For example, I'm trying things like this: kamcmd -v tm.t_uac_start OPTIONS sip:200@exampleserver.com:5060 . . "From: <sip:example@192.168.0.1>;tag=d27ca05e-1789-4774-a698-dcfe91dec9b4\r\nTo: <sip:example@192.168.0.2>"\r\nExpires: 1200\r\nContent-Length: 0" Nothing seems to work, so whatever the syntax I'm using must be wrong. Just providing an example of sending SIP via kamcmd tm.t_uac_start would likely help a huge amount. Thanks! --

2 1

tcp_set_connection_lifetime(1) does not close connection
by Sebastian Damm 19 Dec '21

19 Dec '21

Hi, for having a maintenance mode, I need to implement some sort of draining functionality. All our clients register via TLS. So I thought, I just answer the next REGISTER of a client with a 503 and kill the tcp connection afterwards, so they need to open up a new connection to the loadbalancer in front and end up on a different server. This is my code: 510 if msg.is_method("REGISTER"): 1 if KSR.pv.get("$shv(MAINTENANCE_MODE)") == 1: 2 KSR.info("In maintenance mode, rejecting registration") 3 KSR.hdr.append_to_reply("Retry-After: 2\r\n") 4 KSR.sl.sl_send_reply(503, "Draining") 5 if KSR.tcpops.tcp_set_connection_lifetime(1) > 0: 6 KSR.info("Set Connection lifetime to 1") 7 return False (Side note: msg is some internal class, this is obviously KEMI with python) So now when I set the MAINTENANCE_MODE var to 1, Kamailio rejects the REGISTER requests, logs both lines and sends out a 503. Debug log tells me this: Dec 10 15:32:10 ifens5 /usr/sbin/kamailio[3851]: DEBUG: tcpops [tcpops.c:229]: tcpops_set_connection_lifetime(): new connection lifetime for conid=1: 908866403 Dec 10 15:32:21 ifens5 /usr/sbin/kamailio[3855]: DEBUG: <core> [core/tcp_main.c:4554]: tcpconn_main_timeout(): entering timer for 0x7f759cb26218 (ticks=908866499, timeout=908866499 (0 s), wr_timeout=0 (211631299 s)), write queue: 0 bytes In tcpdump I can see that the connection does not get closed immediately. Is there some timer in the tcp stack which cleans up expired connections from time to time? Can I manipulate how often this timer runs? I couldn't find anything, and clearly even though I tell Kamailio to close the connection after 1 second, it doesn't do it. Thanks for any input. Regards, Sebastian

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Kamailio SIP Application server
by Weiqi Yao 17 Dec '21

17 Dec '21

Hello community, We were testing using Kmailio work as an IMS, and then turn out the phone wants to do MMTel for mobile real-time multimedia communication. We are not familiar with this topic, but anyone know Kamailio have such module or any other MMTel service can work with Kamailio? BR, Weiqi

2 2

RPC command dlg.profile_get_size with per_value response
by David VILLAUME 17 Dec '21

17 Dec '21

Hello, I would like to suggest you an implementation of the rpc command dlg.profile_get_size (on a valued profile) that would return a list of values and the number of current dialogs for any of them . Din’t know where to suggest it so I sent it there, please accept my apologies if it’s not the good place. Regards, David

2 2

SRTP/DTLS + RTP Header encryption
by Rahul mm 15 Dec '21

15 Dec '21

Hi guys, Is it possible to achieve RTP header encryption along with payload in SRTP/DTLS ? So far what I've been able to see is RTP payload encryption. I'm using kamailo 5.5.0 and rtpengine 9.2.1.6 in webRTC scenario. Many thanks. BR, Rahul

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Next online devel meeting - Wed, Dec 15, 2021, 15:00UTC
by Daniel-Constantin Mierla 15 Dec '21

15 Dec '21

Hello, 2021 is almost over, time ran fast somehow this year, with a major release in the middle, which delayed the development discussion for the next one, so it's time for it! As usual, the aim is to sketch the roadmap to next major release so the developers and community members can syncronize and plan what should be done till Kamailio v5.6 will be out as well as discuss about current state of the project, look for new ideas to improve collaboration within the community, a.s.o. The meeting will be done on a matrix chat room. More details are in the wiki page of the event, available at: * https://www.kamailio.org/wiki/devel/irc-meetings/2021a Feel free to add there topics that you want to be discussed. The proposed date is next Wednesday, December 15, 2021, at 15:00UTC (15:00 London/Dublin, 16:00 most of West Europe, 10:00 New York), but other dates can be proposed if they suit for more participants. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda _______________________________________________ Kamailio (SER) - Development Mailing List sr-dev(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-dev

1 2

ACK is sent to wrong address when using serial forking
by Minh Phan 15 Dec '21

15 Dec '21

Dear List, As shown in the image below, Proxy 1 tries to do serial forking to Proxy 3 and then Proxy 4. Upon reception of the 480 answer ( blue frame) from Proxy 2, the Proxy 1 generated an ACK (red frame) and sent it to Proxy 1. The problem I have is that the ACK generated by Proxy 1 has a wrong RURI. I expect the ACK to be sent to the URI of the CONTACT header of the 480 message (blue frame). However, its RURI is the same as the one in the first INVITE of the dialog. Note that there is NO ROUTE header in the ACK message. As there is serial forking here, the ACK is generated by the Proxy 1 itself. Any hint would be appreciated, Best Regards, Minh Phan [image: image.png]

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RTP Proxy with Local IP and Public IP
by Vijo Jose 13 Dec '21

13 Dec '21

Hai Everyone. I have configured the RTP Proxy with Kamailio SIP Proxy Server. Where I have my Kamailio is configured with private IP 10.0.0.1 and Public UP 103.XX.XX.XX and the SIP user is connecting with RTP proxy with local IP 10.0.0.1 to register. I also have a SIP Asterisk server on the public Internet with IP 90.XX.XX.XX that makes the call to sipuser@103.XX.XX.XX:5060 which is coming on the sip proxy server and the sip proxy server forwards those packets to the SIP user who is registered with local IP 10.0.0.5. Once the call is answered the RTP is started to send from the SIP server to the SIP Proxy server over the public IP from 90.XX.XX.XX to 103.XX.XX.XX which is working as expected. But from the sip user 10.0.0.5 the RTP is flowing to Proxy server public IP 103.XX.XX.XX instead of RTP Proxy server Localip 10.0.0.1. How can we configure the same? Presently I have started my RTP proxy server with command /usr/local/bin/rtpproxy -F -A 103.XX.XXXX -l 10.0.0.1 -m 10000 -M 20000 -s udp:127.0.0.1:7722 -d DBUG:LOG_LOCAL5 Thanks in advance.

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Re: [SR-Users] Manipulation of request-line
by Mihovil Kolaric 12 Dec '21

12 Dec '21

Hi, unfortunately, Alex is right: manipulations ot the request-URI do not seem to re-write the whole line, the malformed line-ending (SIP/2.0<blank><CR><LF>) remains in the forwarded message.

2 2

remove rport in INVITE Via HDR and make kamailio behave asymmetric
by Nuno Miguel Reis 11 Dec '21

11 Dec '21

Hi guys. I've a scenario here where I'm receiving inbound traffic from a Mitel PBX on a LAN environment with a bunch of router hops between kamailio and the Mitel. Sometimes, not always, I get traffic from the MItel PBX where the source port is changed in transit (some kind of NAT, still don't know where exactly) from 5060 to a dynamic port different from 5060. Every INVITE comes with rport in via header as: .. Via: SIP/2.0/UDP 10.21.252.35:5060 ;branch=z9hG4bK-524287-1---d6bac01809527627*;rport* .. when this happens kamailio sends all the replies to the $si:$sp symmetrically (force_rport behavior) and I would like to force the opposite instead, making it asymmetric and sending the traffic to $si:5060 everytime. I'm having a hard time forcing the *unforce*_rport() behavior. Does anyone have a good suggestion for this? Looking forward to hear from you. Cheers, -- Nuno Miguel Reis Departamento de Engenharia Informática Faculdade de Ciências e Tecnologia Universidade de Coimbra

1 1

Unquoted values from #!defenv
by Ben Kaufman 09 Dec '21

09 Dec '21

Hello all, Looking for some clarification regarding quoting and !#defenv . I understand that the example below fails because the environmental variable HOME is not quoted, thus when used in the xlog() call it fails. How is it possible to evaluate and/or use the value of the environmental variable if defined as a preprocessor variable? Looking at the documentation the reason for this directive is "It is a simplified alternative of using #!substdef with $env(NAME) in the replacement part." But #!substdef would allow setting the value inside of a quoted string, so does it really meet that requirement? I'm aware that I could use $env(HOME) rather than a preprocessor directive of HOME. I'm just trying to understand the usage of the #!defenv feature. #!KAMAILIO ## Tested in 5.5.2 loadmodule "xlog" loadmodule "pv" loadmodule "evrexec" modparam("evrexec", "exec", "name=evrexec:timer;wait=1000;workers=1;") #!defenv HOME request_route{ forward(); } event_route[evrexec:timer] { xlog("L_N", "HOME: " + HOME + "\n"); } Ben Kaufman

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wireshark decode kamailio / rtpengine ng messages
by Karsten Horsmann 08 Dec '21

08 Dec '21

Hi List, maybe someone wrote a wireshark helper (didnt know the right term) to view kamailio to rtpengine communication in human form? Some hints would be great. -- Mit freundlichen Grüßen *Karsten Horsmann*

2 1

Re: [SR-Users] wireshark decode kamailio / rtpengine ng messages
by Alex Balashov 08 Dec '21

08 Dec '21

That sounds far too niche for a helper, though, then again, Wireshark has some phenomenally exotic helpers. — Sent from mobile, with due apologies for brevity and errors. > On Dec 8, 2021, at 8:14 AM, Karsten Horsmann <khorsmann(a)gmail.com> wrote: >

1 0

Kamailio v5.3.9 Released
by Daniel-Constantin Mierla 08 Dec '21

08 Dec '21

Hello, Kamailio SIP Server v5.3.9 stable release is out. This is a maintenance release of the old stable branch, 5.3, that includes fixes since the release of v5.3.8. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.3.x. Deployments running previous v5.3.x versions are strongly recommended to be upgraded to v5.3.9 or even better to v5.4.x or 5.5.x series. For more details about version 5.3.9 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2021/12/kamailio-v5-3-9-released/ RPM, Debian/Ubuntu packages will be available soon as well. Note: the branch 5.3 is an old stable branch, v5.3.9 being the last planned release out of it, to mark the end of official maintenance. The maintained stable branches are now 5.4 and 5.5, at this time with v5.5.3 being the latest stable release. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

Re: [SR-Users] Manipulation of request-line
by Alex Balashov 08 Dec '21

08 Dec '21

That will manipulate the request URI, but not the request line. — Sent from mobile, with due apologies for brevity and errors. > On Dec 7, 2021, at 4:16 PM, Karsten Horsmann <khorsmann(a)gmail.com> wrote: >

2 1

Releasing Kamailio 5.3.9 - last in 5.3.x series
by Daniel-Constantin Mierla 08 Dec '21

08 Dec '21

Hello, it is time to mark the end of official packaging for 5.3.x releases, therefore during the next days I am going to cut the v5.3.9 release from branch 5.3, likely on Wednesday, Dec 8, 2021. In other words, if no major regression is detected shortly after this release, the v5.3.9 is going to be the last one from the 5.3 branch. It is strongly recommended to move to a series that is under active maintenance, respectively 5.5.x or 5.4.x. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Online Feb 21-24, 2022 (America Timezone) * https://www.asipto.com/sw/kamailio-advanced-training-online/

1 1

HTTP Requests for API routing - Question on which module to use with KSR
by Emilio Panighetti 08 Dec '21

08 Dec '21

Looking into using Kamailio to route calls via an HTTP API. Basic scenario is this: INVITE comes in (no registration). And Kamailio sends an HTTP request to a server with data extracted from the INVITE. Based on the response, call KSR.setdsturi() and other actions like normalizing phone numbers, etc. (Not sure if changing Contact: headers would be a better option). For now I want to be able to make a simple call routing as a proof of concept. Planning on using Lua as a KEMI interpreter. Looking at different modules: http_client http_client_async xhttp Which module would be more appropriate? http_client_async provides a simple async method and callback, so the SIP threads can continue processing requests while the API server is preparing the response. But the call has to be parked (How is this done?) and this is also an overhead. http_client seems straightforward, but the SIP threads are tied up waiting for HTTP responses. Also looking at the KEMI documentation, the KSR function calls for HTTP_CLIENT doesn’t seem to match the config function documentation. For example, In KEMI manual: KSR.http_client.curl_connect_post() doesn’t have a corresponding in kamailio.cfg. So which would be the preferred approach? Also examples would be appreciated. Thanks in advance!

1 0

Manipulation of request-line
by Mihovil Kolaric 07 Dec '21

07 Dec '21

Hi, I have an issue with a SIP-client, that occasionally puts a blank between the version SIP/2.0 and the CRLF at the and of the request-line. (i.e. sends "ACK <uri> SIP/2.0 \r\n" instead of "ACK <uri> SIP/2.0\r\n"). Kamailio (as well as most SIP UAs) does not care about that error and successfully processes/forwards the request (also when using sanity_check()). But I also deal with a client that fails to parse that request. (Furthermore, wireshark's dissector fails to recognize such malformed messages as SIP, which makes debugging even harder). I tried to fix this issue using textops, but all functions I tried operate only on headers or message body, but do not seem to affect the request line. Is there any possibility to modify the request-line, or to replace it with a new one? Best Regards, Mihovil

2 1

Announcing public beta release of TeamsPhone - powered by Kamailio
by Sergiu Pojoga 06 Dec '21

06 Dec '21

Hello Kamailio-nians, Just to bug everyone quickly: today I'm happy to announce the public beta release of *TeamsPhone <https://teamsphone.net>*, a MS Teams <=> PBX integration solution for ITSPs, IP-PBX service providers and integrators, *powered by Kamailio*. Unlike other similar services offered as SaaS pay-per-user model, our product is oriented for ITSPs to be self-hosted and fully integrated into one's telecom network, providing an add-on service to supplement any Hosted PBX offering with seamless calling using Microsoft Teams *as-as-softphone*, or as we like to call it - TeamsPhone. Some insights about the platform: - 100% automation of MS Teams Direct Routing provisioning - multi-tenant application - API-driven - Microservices oriented architecture - RabbitMQ-powered Message Broker distribution - SIP server (you guessed it - Kamailio, of course!) Seeing how it works is worth a thousand words. A cloud *demo* instance is available at https://my.teamsphone.net for anyone to try it out, use it or abuse it. If you have questions about the product - feel free to contact me in private. Respectfully, Sergiu Pojoga Founder, teamsphone.net sergiu(a)teamsphone.net

1 0

detecting compact headers?
by Brooks Bridges 06 Dec '21

06 Dec '21

We have an internal component of our callflow that does not support compact headers (vendor says it's on the roadmap but no eta, etc) so I'm trying to find a clean way to detect if a message has compact headers and reject it. I haven't had much luck in finding a way to identify specifically if any of the headers of the message are compact however. Can anyone point me in a direction please? Thanks! Confidentiality Notice: This e-mail, and any attachment to it, contains privileged and confidential information intended only for the use of the individual(s) or entity named on the e-mail. If the reader of this e-mail is not the intended recipient, or the employee or agent responsible for delivering it to the intended recipient, you are hereby notified that reading this e-mail is strictly prohibited. If you have received this e-mail in error, please immediately return it to the sender and delete it from your system.

3 3

ENV Values in cfg
by H Yavari 06 Dec '21

06 Dec '21

Hi to all, Is it possible to set modules parameters with env values? I know that in routes we can use $env(NAME), but what about for other parameters? for example setting DB address in K8s cluster ENV values. Regards,Hossein

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sr-users