[SR-Users] Re: Proxy edge

In WebRTC (websocket) the contact header always has invalid URI, so, you MUST either use path module OR add contact alias for both REGISTER and SUBSCRIBE requests otherwise any subsequent incoming SIP requests such as INVITE and NOTIFY won't be routed to endpoint (SIP client). See websocket

Re: [SR-Users] About the WITH_VOICEMAIL Subscriber message 407 issue

Check if you have, # IP authorization and user authentication > route[AUTH] { > ... > # user authenticated - remove auth header > if(!is_method("REGISTER|PUBLISH")) > consume_credentials(); > ... > } This basically allows only REGISTER and PUBLISH to be authenticated

Re: [SR-Users] Need Suggestion/Idea to set call limit on 3 kamailios.

Setup a counter in redis server shared across all kamailios, add to counter when call is answered and subtract when call hangs up.You can use any existing db that you already have instead of redis as long as it is accessible across all kamailios.--Muhammad Shahzad ShafiTel: +49 176 99 83 10 85Sent

Re: [SR-Users] Multiple rtpengine servers

Have a look at RTPEngine usage documentation here,https://kamailio.org/docs/modules/5.6.x/modules/rtpengine.html#idm80It answers all your questions. In short, you can have multiple sets of RTPEngines, each set can have multiple instances and each instance has weightage to define its priority

Re: [SR-Users] Kamailio behind NAT

That's likely because kamailio does not know where the destination endpoint is. You must either share user location information with kamailio or somehow keep track of registered users on kamailio (dual registration or cascade registration or path header if supported by pbx etc.).--Muhammad Shahzad

Re: [SR-Users] Kamailio 5.5 STIR SHAKEN private key buffer size error

This error is seems to come from libstirshaken ( https://github.com/signalwire/libstirshaken/blob/master/include/stir_shaken.h line 46) and has nothing to do with Kamailio. Please open a bug with signalwire who owns and maintains this library. Per my understanding this library is bit old and

Re: [SR-Users] Phones are not getting registered. Is my setup correct?

Seems like you haven't set server IP. You need to SSH to server and then type "setup" then go to Network Configuration -> Edit device -> select eth0 (or whatever is your ethernet device name is) then uncheck DHCP and set static IP, also make sure set DNS correctly as well. Then save and exit

Re: [SR-Users] remove headers with less than 5 characters with textops

Removing headers with up 5 characters in name would make SIP message invalid since many critical headers such as TO, FROM, and VIA etc. will also be removed. So, even if the regex works, Kamailio will NOT relay this SIP message. Thank you. On Tue, Jun 7, 2022 at 5:00 PM Patrick Karton wrote:

Re: [SR-Users] xhttp_pi show dr_rules buffer 2 small Error.

You need to increase xhttp_pi_buf_size, https://kamailio.org/docs/modules/5.5.x/modules/xhttp_pi.html#idm193 -- Muhammad Shahzad Shafi Tel: +49 176 99 83 10 85 On Thu, Oct 28, 2021 at 7:34 PM Abdirahman Osman wrote: > Hello Everyone, > > I am using the xhttp_pi modules to see the list of

Re: [SR-Users] Is this idea even feasible?

Not really, these are PABX features and Kamailio is not such. You would need something like Asterisk or FreeSWITCH which have several options to do such stuff.Question is how would this media server know when to hold, transfer etc. If the endpoint is a dumb sip softphone as you described?Sent from

Re: [SR-Users] Kamailio python_exec getting randomly TypeError

The dlg_var is string, make sure you cast them to int for duration calculations. Also manually cast var(v_duration) to int as well, because python_exec method treats them as string (this shouldn't be a problem since python vars are typeless but just to be sure).Now about the Null values in

Re: [SR-Users] Guidance building a WebRTC to Asterisk Intermediate Proxy

The problem seems to that the kamailio is trying to route it over UDP port using WSS transport, which will never work. You are creating a transport bridge (UDP <-> WSS i.e. stateless transport <-> stateful transport) as well media bridge (RTP <-> SRTP), so it is going to be tricky. I strongly

Re: [SR-Users] Dialog - timeout for dlg with CallID

Please put dlg_manage immediately after loose_route condition, e.g. if (loose_route) { dlg_manage(); ... } There is no harm in calling it for every SIP method (not just BYE or ACK) within dialog. It will help for re-invites (call hold / unhold events) etc. from same buggy UAs as well. Thank

Re: [SR-Users] Dialog - timeout for dlg with CallID

Ok. Then possibly you have a buggy UA that is tempering with RR. Try following, 1. Use dialog match mode 1. 2. Add dlg_manage after loose_route. If problem persists then share sip trace of such call. Hope this helps. On Fri, 24 Jul 2020 at 16:33, Ilie Soltanici wrote: > Yes, I do use the

Re: [SR-Users] Dialog - timeout for dlg with CallID

What does the SIP trace at kamailio says? Do you see ACK pass through kamailio? Are you doing record-route on call because otherwise ACK and/or BYE may bypass kamailio. Hope this helps. On Fri, 24 Jul 2020 at 12:49, Ilie Soltanici wrote: > Hello, > > I'm trying to get CDR working in Kamailio

Re: [SR-Users] bug ? remap_503_500 breaks dialogs

The SIP code 503 is tricky in the sense that i can indicate either server maintenance or server overload. In both cases it can send Retry-After header and any subsequent requests from same source are ignored for the duration of Retry-After interval. [1]. Additionally RFC3261 and RFC3263 define

Re: [SR-Users] usrloc db_mode 2 and dropped contacts

ace condition along those general lines. > > That still leaves some questions surrounding the purpose and necessity of > various settings, however. > > — Alex > > — > Sent from mobile, with due apologies for brevity and errors. > > On Mar 7, 2020, at 1:28 AM, M S

Re: [SR-Users] usrloc db_mode 2 and dropped contacts

Hi, There are multiple scenarios and possibilities for this behavior, please check, 1. What kind of sip endpoints you have? Any TCP based transport e.g. TLS or WSS endpoint? Are they mobile apps (especially iOS devices) with push notification and "backgrounding" support? 2. How many distinct

Re: [SR-Users] R: Kamailio connection issue

You have many problems, 1. You do not seem to have TCP keepalives setup. So idle TCP connections may be disrupted, closed or reset by any router on the internet, forcing SIP client to re-register. As a result many more tcp connections are created by a single SIP end point then expected. You said

Re: [SR-Users] (no subject)

disable "verify_certificate" option in tls module. Hope this helps. On Tue, 3 Mar 2020, 08:53 ahmed moghazy, wrote: > I know that but i want voip program to be used with kamailio using tls i > you could suggest one , thank you > > Sent from my iPhone > > On Mar 3,

Re: [SR-Users] push notification via lua script on tsilo module

You don't need to use LUA for the task, you can use any language you are comfortable with, e.g. perl, python or php. You can even use remote nodejs and call it from kamailio as http request. There are tons of example scripts, tutorials and howtos online regarding how to send push notifications

Re: [SR-Users] (no subject)

kamailio. On Mon, 2 Mar 2020, 20:27 ahmed moghazy, wrote: > I put my own certficate in /res/raw/rootca.pem in linphone > > Sent from my iPhone > > On Mar 2, 2020, at 7:40 PM, M S wrote: > >  > Please read the tls module documentation, > > > https://kamail

Re: [SR-Users] I need to configure apns push notification in kamailio

p must what what kind of > parameters and how many parameters we needed ??? > > > Sent with ProtonMail <https://protonmail.com> Secure Email. > > ‐‐‐ Original Message ‐‐‐ > On Thursday, February 27, 2020 6:38 PM, M S wrote: > > Push notification setup is long

Re: [SR-Users] (no subject)

Please read the tls module documentation, https://kamailio.org/docs/modules/devel/modules/tls.html#tls.p.require_certificate when enabled it requires client to present certificate, but client does not have one, which is the likely cause of the problem. Hope this helps. On Mon, 2 Mar 2020,

Re: [SR-Users] I need to configure apns push notification in kamailio

Push notification setup is long and complex but generally has following process. Don't forget to read documentation of relevant kamailio modules for proper use. 1. The iOS app must send push notification parameters in SIP request e.g. SIP Register. In kamailio, when this request is received, you

Re: [SR-Users] Rtpproxy not work in 2 kamailio + rtpproxy setup

You are nesting one problem description into another and it all does not makes any sense to me. Anyways, what i understand is that you are trying to engage two RTP proxies in series from WAN to LAN to WAN again. So you need to create media bridge at each kamailio instance that routes RTP traffic

Re: [SR-Users] RTP Engine in docker

You would need to install kernel module in host machine, only then it will be available in docker container. You will also need to mark container as privileged container and enable 1-to-1 NAT for RTP port range between host and vm (i recommend using iptables for this instead of using docker port

[SR-Users] No way to send SIP reply from app_perl or jsonrpc modules

Hi, I have a use case where I need to send a statefull (or stateless) reply to non-INVITE SIP transaction from PERL script using app_perl module's "moduleFunction" method, https://kamailio.org/docs/modules/5.0.x/modules/app_perl.html#ID-7b0efd72c8ec89bc43dad940590b40c6 It fails with error

Re: [SR-Users] Very strange reinvite behaviour

ITSP). If however the callee can NOT fix this problem then you will have store the original Contact header in initial INVITE from caller side somewhere in kamailio script e.g. htable, and change RURI to this value BEFORE calling loose route. Which should fix the problem. Hope this helps. Thank you. --

Re: [SR-Users] Very strange reinvite behaviour

you. -- M. S On Tue, Aug 14, 2018 at 7:19 PM, Alex Balashov wrote: > One curiosity is that the reinvite does not appear to have the Via from > the UAC that sent the initial invite. But certainly that would not cause > this behaviour? > > On Tue, Aug 14, 2018 at 12:52:50PM -0400

Re: [SR-Users] Selectively enable / disable cdrs using acc module

I have tested it. It does not work. It only stops sip transaction accounting but not the cdrs. I think i have to add support in native code for dialog level flag or variable to allow the customization of cdrs...humm, lets see. Thank you. On Sat, 21 Jul 2018 at 17:40, M S wrote: > Ok, let

Re: [SR-Users] Selectively enable / disable cdrs using acc module

t > you give it a try. > > On Sat, Jul 21, 2018 at 2:16 AM M S wrote: > >> Humm, isn’t acc log_flag is for transaction logging, rather then cdrs? >> >> For cdrs, the acc module has just one relevant module parameter, that is >> cdr_enable which enables / disables cdr

Re: [SR-Users] Selectively enable / disable cdrs using acc module

set to FLT_ACC, it should be as simple as > not setting or resetting the FLT_ACC flag, depending on the script logic. > > You will probably need to do that for both INVITEs and BYEs > > > On Fri, Jul 20, 2018 at 2:48 PM M S wrote: > >> Hi, >> >> I have a kamailio v

[SR-Users] Selectively enable / disable cdrs using acc module

Hi, I have a kamailio v5.0 setup where cdrs are enabled using acc module. Is there a way to enable / disable cdrs on per call basis in kamailio script? For example i want to allow cdrs only for offnet calls and disable them for onnet calls? May be a dialog flag configurable in module parameters

Re: [SR-Users] MSILO: SIP stored offline MESSAGE repetitive delivery

Hi, When kamailio dumps the messages to destination party upon register, do you see its reply received by kamailio correctly? Most probable cause of the problem is that kamailio does not receives 200 OK for dumped message from destination party and assumes message delivery failure, therefore it

Re: [SR-Users] Return codes within a route

Hi, My understanding is that return value of last statement becomes the return value of named route if none explicitly specified. Therefore i always explicitly return 1 or exit. — Muhammad. On Tue 20. Feb 2018 at 22:48, Jeff Bilyk wrote: > Hello, > > Running into odd

Re: [SR-Users] PUBLISH and ETag

First, RFCs related to SIP presence are quite confusing sometimes and often not fully implemented by presence servers and endpoints. Secondly, dialog presence event for first call has already completed its life-cycle i.e. It has been terminated by second publish from Asterisk. You can not change

Re: [SR-Users] Kamailio fail to start with db_mongodb and ndb_mongodb modules

Hi, Enable core dump and provide backtrace of sigfault. See this for more info, https://sourceware.org/gdb/onlinedocs/gdb/Backtrace.html Thank you. On Fri, Dec 29, 2017 at 12:25 PM, Abdul Basit wrote: > Adding further from https://stackoverflow.com/questions/2549214/

Re: [SR-Users] Trying to fail a call over after 200 OK

Hi, I don't think you can FAIL a call that is already answered i.e. marked as SUCCESSFUL. The only way to achieve the functionality you desire is to use SIP REFER. On Kamailio you can achieve this using, https://kamailio.org/docs/modules/4.1.x/modules/dialog.html#idp17465536 On Asterisk you may

Re: [SR-Users] Loadbalancing authenticated agetways

Hi, Here is a simpler easiler solution for your case. Prerequisits: 1. Media gateways should register for somewhat longer period of time, e.g. 3600 seconds (shorter intervals would work too but longer is better). 2. All media gateways should register to kamailio through same kamailio listen

Re: [SR-Users] [sr-dev] Planning Kamailio v5.0.4

Hi, There is a Kamailio sigfault issue #1277 that is also present in current master branch. I would really appreciate if you can take a look at it before next release, so i can upgrade my kamailio setup to v5.0 upon next release. Thank you. On Mon, Oct 23, 2017 at 10:58 AM, Daniel-Constantin

Re: [SR-Users] Docker image

That's great. I was thinking of experimenting a few things with Kamailio in docker but couldn't find any suitable image to start with. Thank you for setting it up. I will definitely try it. Keep up the good work. -- M. S. On Mon, Sep 25, 2017 at 7:35 PM, Sergey Safarov <s.safa...@gmail.

Re: [SR-Users] TCP read/write buffer size on per connection or per module basis

-Constantin Mierla <mico...@gmail.com> wrote: > Hello, > > > On 02.05.17 13:02, M S wrote: > > Hi, > > > > I have a kamailio v4.4.5 deployment that serves TCP clients for SIP > > and XCAP requests (without SIP presence). The kamailio's core TCP > > para

[SR-Users] TCP read/write buffer size on per connection or per module basis

Hi, I have a kamailio v4.4.5 deployment that serves TCP clients for SIP and XCAP requests (without SIP presence). The kamailio's core TCP parameters allow us to set TCP read and write buffer sizes which are set globally for all TCP connections. However, i noticed that in general SIP over TCP