Re: [SR-Users] Disable DNS

On Tue, Apr 11, 2017 at 02:58:23PM +0200, Daniel-Constantin Mierla wrote: > > Just took a look at the dns options, strangely enough there is no way to > > disable it apparently. > > If DNS would be disabled, then how the request should be routed? By using dispatcher with ip adresses, which sets

[SR-Users] sipcapture/HEP msg sizes>1500

While looking into HOMER I saw that large INVITEs got truncated in the webinterface, looking at the HEP traffic these (UDP) messages got fragmented in transport (asterisk/res_hep) and I started to think these were not getting reassembled correctly. Only to find out the database schemas have set

Re: [SR-Users] Dispatcher call id hash for server side invite

On Thu, Aug 10, 2017 at 04:48:48PM -0400, Ryan Wagoner wrote: > In my REGFWD route I have this chunk of code to lookup the dispatcher > IP/port from the same hash table before calling uac_req_send(). The > regserver column contains the dispatcher set id. Is there a way I can call > ds_select and

Re: [SR-Users] Dispatcher call id hash for server side invite

On Wed, Aug 09, 2017 at 01:21:36PM -0400, Ryan Wagoner wrote: > How do I mark these server side invites so the call hash is known by > kamailio? Or am I thinking about this the wrong way? It is the latter, in dialog replies should be routed on headers only. In the "default" config these

Re: [SR-Users] Route calls from Kamailio to Asterisk behind NAT

On Fri, Aug 11, 2017 at 01:29:18PM +0300, v...@cell.tel wrote: > How its possible to route all 7XXX destined calls from Kamailio users to > Asterisk, while Asterisk is behind NAT??? >   > Config examples, i found, were writen only for situation with fixed asterisk > IP > and PORT, without nat (

Re: [SR-Users] Route calls from Kamailio to Asterisk behind NAT

On Sat, Aug 12, 2017 at 08:49:45AM +0300, v...@cell.tel wrote: >Using alias_db - 7XXX calls were succesfully aliased and routed to >Asterisk through user (asterisk server did not accept connections to >sip ports from Internet due to security reasons). >On asterisk side, all

Re: [SR-Users] Route calls from Kamailio to Asterisk behind NAT

On Fri, Aug 11, 2017 at 09:35:45AM -0700, Benjamin Fitzgerald wrote: > Another amazing tool for this is sipgrep ( > https://github.com/sipcapture/sipgrep). It's like ngrep but for SIP. And here I thought sngrep was the ngrep for SIP :) ___ Kamailio

Re: [SR-Users] Dispatcher call id hash for server side invite

On Thu, Aug 10, 2017 at 08:51:24AM -0400, Ryan Wagoner wrote: > Thanks for the explanation! I finally got the dispatcher working in an > active / passive Kamailio cluster in front of three FreePBX servers. I was > using the Asipto Kamailio and Asterisk real time guide as a starting point > so it

Re: [SR-Users] Kamilio new user - help get going

On Thu, Jul 13, 2017 at 09:46:03PM +, Kenneth Knudsen wrote: > Thanks, but then I’m missing the next step, how do we configure the > connection between Kamilio and the FreePBX servers ? > > How do we configure the routes, public ip, internal FreePBX servers. > > This make no sense to me at

Re: [SR-Users] Issue in Dialplan Module

On Wed, Aug 09, 2017 at 01:34:13PM +0530, Logeshwaran G wrote: > I have given the below query, > > INSERT INTO dialplan(id, dpid, pr, match_op, match_exp, match_len, > subst_exp, repl_exp, attrs) VALUES (1, "1", "1", 1, "^000[0-9]+$ ", 0, > "^000([0-9]+)$", "\1", " "); > > But the Rule is tking

Re: [SR-Users] Registrar limit

On Wed, Aug 09, 2017 at 03:11:21PM +0300, Yu Boot wrote: > How to limit one user login with one AOR (IP) and always use IP from last > registration attempt? See https://www.kamailio.org/docs/modules/stable/modules/registrar.html#registrar.f.save flags for save(): 0x04 - store and maintain one

Re: [SR-Users] Usleep on Reply

On Tue, Aug 01, 2017 at 03:40:56PM +, Nicolas Breuer wrote: > I think it will not match because I want to send a "180" in case I receive a > 183 without SDP and no 180 from the carrier. ... > I tried to delay the processing of the "183" and : > > If a 180 is received before the delay

Re: [SR-Users] Kamailio: how to route RTP media directly to endpoint

On Wed, Aug 09, 2017 at 04:48:02PM +0300, wsotest.512 wrote: > > UserA ---sip--> Kamailio --> Asterisk --> UserB > >\-rtp--> --> --> UserB > > Is it possible at all? Maybe someone already did it . It should work, but Asterisk is broken in this respect and may break codecs/dtmf:

Re: [SR-Users] Module for Call placement between the two Different Switches

On Sat, Aug 05, 2017 at 11:34:18AM +0530, Logeshwaran G wrote: > I am Having my own kamailio Pbx setup, I Have my local subscriber list > database, I need to allow the calls from the outside switch(For example > Nexge or Itel etc...) What module I have to refer for this?? There is no need for

Re: [SR-Users] Set up caller ID

On Tue, Aug 22, 2017 at 04:28:35PM +0200, Nelson Migliaro wrote: > I am using the uac_replace_from function but I would like to pull the > caller id from a database to set up the number based on the sip vendor the > call is forwarded > > What would be the correct way to do that? > > I was

[SR-Users] siptrace: how to duplicate the complete dialog?

I'm trying to trace a complete dialog on a loadbalancer/proxy: UAC <-> Proxy<-> backend 109.235.34.a <-> 109.235.32.b <-> 185.61.68.c Proxy runs 4.3.x (but with 4.4.x I appear to have the same results). siptrace config: =

Re: [SR-Users] rtpengine cpu usage

On Wed, May 03, 2017 at 12:10:57PM -0300, Diego Nadares wrote: > I'm using rtpengine (compiled from source) and I see a lot of difference in > cpu usage comparing with rtpproxy. Is this normal? Yes, rtpengine uses less cpu (when configured correctly) > With 600 calls: > > cpu > *41.7%*

[SR-Users] Why is the To URI the default in save()?

The save function from the registrar module uses the To header to disect and store the username for the location table according to observations and documentation http://www.kamailio.org/docs/modules/stable/modules/registrar.html#registrar.f.save After troubleshooting a ticket from an enduser

Re: [SR-Users] permissions module db

On Wed, May 10, 2017 at 06:54:49PM +, David Villasmil wrote: > I don't think I can... there's no cast to smallint :( Indeed, but take a look at this workaround with a custom function returning a smallint: http://ask.brothersoft.com/how-to-convert-from-int-to-smallint-86641.html (which

Re: [SR-Users] Live streaming from Kamailio World Conference 2017

On Wed, May 10, 2017 at 11:26:23AM +0200, Daniel-Constantin Mierla wrote: > if you haven't noticed from other social networking channels, there is a > live stream from Kamailio World 2017, thanks to Pascom team: > > - http://live.pascom.net/ > > Enjoy! I enjoyed it indeed. Thanks for the

Re: [SR-Users] permissions module db

On Wed, May 10, 2017 at 05:31:28PM +0200, David Villasmil wrote: > The view i created on mysql returns an INT for port and mask, it looks like > kamailio doesn't like this... It looks to be expecting a smallint?? > > the view returns an INT and that fails > ᐧ Sound awfully familiar:

Re: [SR-Users] Why is the To URI the default in save()?

On Mon, May 15, 2017 at 03:06:38PM +0200, Daniel-Constantin Mierla wrote: > > This opens the door to hijacking incoming calls to other users on the > > same kamailio registrar if one knows/guesses other usernames and use > > those in the To header. > SIP allows third party registrations. From

Re: [SR-Users] Better place to change the FROM header for every INVITE

On Thu, Jun 22, 2017 at 04:33:51PM +0200, Emanuele Gambaro wrote: > I need to change the FROM header for every invite and replace the > caller username with another data taken from mysql database. Which is > the best place in kamailio routes to do a uac_replace_from ? > > It is correct to change

Re: [SR-Users] get online user from api

On Tue, Jun 27, 2017 at 03:50:17PM +0800, 赵国杰 wrote: > I want to know how many users are currently registered to my > kamailio server. I understand can be used to achieve this. I'm > wondering if there's RESTful or RPC interface that can be > requested by external service. Yes there are RPC

Re: [SR-Users] Detect network range in UDP traffic

On Tue, Jun 27, 2017 at 11:03:00AM +0200, Emanuele Gambaro wrote: > Hi to all > I have the following scenario (Kamailio 5 and rtpproxy) > *) Bob on a wifi network calls Alice > *) During the call, Bob losts his wifi network (and so also changes his IP > address) > *) The RTP flow is interrupted >

Re: [SR-Users] two way routing

On Wed, Jun 14, 2017 at 05:28:35PM -0400, Jean Cérien wrote: > I have the registration working, but when my kamailio receives an INVITE, I > am a bit lost regarding what to do. in my route(AUTH), I am trying to check > uac_reg_status, but no success. > and the invite arriving > INVITE

Re: [SR-Users] best practice to re-install rtpengine after jessie to stretch upgrade?

On Thu, Jun 15, 2017 at 09:33:29AM +0300, Juha Heinanen wrote: > I made a test upgrade from Debian Jessie to Stretch while rtpengine > 5.1.1 was running on Jessie. What is the best practice to get > xt_RTPENGINE module installed in the new Stretch linux kernel, since it > does not happen

Re: [SR-Users] two way routing

On Fri, Jun 16, 2017 at 11:55:06AM -0400, Jean Cérien wrote: > Is there a way to lookup in the uacreg table with the IP ? I could only > find uac_reg_lookup, but it required the luuid. I've worked around that > adding an entry into address, but I am not really satisfied with this > method, You

Re: [SR-Users] Issue with INVITE to NATed client

On Mon, May 08, 2017 at 08:04:27AM +0100, Iskren Hadzhinedev wrote: > I'm already adding a Path header and my PBX sends correctly to Kamailio. > However, Kamailio tries to send to the RFC1918 IP address. > I tried replacing `add_path()` with `add_path_received()` but that had no > effect. In that

Re: [SR-Users] handle_uri_alias persists through failure_route

On Mon, May 01, 2017 at 02:26:55PM +0200, Daniel-Constantin Mierla wrote: > failure route is preserving the attributing of the branch that received > the failure reply. > > You have to use resetdsturi() in this case -- see the readme of kex module. Is resetdsturi in any different from $du=$null

Re: [SR-Users] Issue with INVITE to NATed client

On Fri, May 05, 2017 at 03:56:10PM +0100, Iskren Hadzhinedev wrote: > However, when a call comes from PSTN to the client, kamailio sends the > INVITE to the client's RFC1918 IP and I can't figure out how to send it to > the correct destination. I tried loading the registrar and usrloc modules >

Re: [SR-Users] kamailio 4.4 registering to a SIP trunk provider using UAC then outbound routing to it.

On Wed, Sep 13, 2017 at 03:33:19PM +, Jonathan Hunter wrote: > I have a SIP provider who only accept registrations from users, so I > have registered my kamailio Server with them using the uac module. > > It is registered fine, and I can get inbound calls on the trunk. > > My query is the

Re: [SR-Users] how to play ring tune when callee declines

On Tue, Sep 19, 2017 at 11:26:46AM +0800, 赵国杰 wrote: > Thanks Daniel, I've done some digging, and from Andrew Prokop's blog, > it says this envolves early midia. Usually this is done by reply a 183 > to the caller with media ip and port in the SDP. This makes sense but > i still have no idea how

Re: [SR-Users] Manipulate From URI username

On Wed, Oct 04, 2017 at 10:34:34AM +0200, Gertjan Wolzak wrote: > Im trying to manipulate the username from the from uri, but not having any > luck, I must be doing something wrong. > > I receive a from uri username that is prefixed with 888, now I am trying to > remove the triple 8. > > > I

Re: [SR-Users] Handle SDP ICE candidates

On Mon, Oct 09, 2017 at 05:45:01PM -0300, Thiago Guimarães wrote: > I would like to know if there is any specific configuration in > kamailio.conf to enable/disable handle the array of ICE candidates. > > In my tests, using PJSIP for the development of my User Agent, I > notice that when ICE is

Re: [SR-Users] Inserting local path into shared USRLOC table.

On Tue, Oct 10, 2017 at 11:36:56PM +0100, Marrold wrote: > I've included add_path_received() in my REGISTRAR routing block, however I > see 'Path: [not set]' in the USRLOC table. > > Is it possible to force the insertion of the local proxy into the path > field? Like Daniel already mentioned,

Re: [SR-Users] handle_tcpconn_ev(): connect failed

On Thu, Sep 07, 2017 at 11:03:49AM +0300, Donat Zenichev wrote: [snip] > ERROR: [tcp_main.c:4070]: handle_tcpconn_ev(): connect > XXX.XXX.XXX.XXX:52185 failed > > The port that kamailio2 tries to use to relay the ACK, is port that > endpoint used to establish the dialog with kamailio1 and

Re: [SR-Users] handle_tcpconn_ev(): connect failed

On Thu, Sep 07, 2017 at 01:05:02PM +0300, Donat Zenichev wrote: > > Having a similar setup with failover for the loadbalancers, I take for > > granted that TCP/TLS will fail in case of a failover (but UDP will keep > > working after failover due to the stateless nature of it). > > Well, your

Re: [SR-Users] sql_do_query_async(): the db driver module doesn't support async query

On Tue, Aug 29, 2017 at 07:44:34PM +0300, Juha Heinanen wrote: > > Are you really using mysql directly? For example the cluster driver > > doesn't support async, regardless of the backend driver support. > > I re-checked, and, yes, I was using cluster driver. Had a feeling that would be case :)

Re: [SR-Users] UACREG - answering to 401 on invite

On Tue, Sep 26, 2017 at 09:08:26AM -0400, Jean Cérien wrote: > With the UACREG module, I am registering to a remote provider. Register > goes out, 401 back, Register goes out with nonce & co, OK > > Later, when I send an invite, the provider issues an 401 Unauthorized. I > guess it expects me to

Re: [SR-Users] UACREG - answering to 401 on invite

On Tue, Sep 26, 2017 at 09:36:19AM -0400, Jean Cérien wrote: > I've inserted the following block on the failure route: > if (t_check_status("401|407")) { > xlog("L_INFO","failure_route(ROUTEFAIL) @@ call to > uac_auth()\n"); > uac_auth(); >

Re: [SR-Users] siptrace: how to duplicate the complete dialog?

On Mon, Aug 21, 2017 at 10:54:15AM +0200, Daniel Tryba wrote: > > http://kamailio.org/docs/modules/4.4.x/modules/siptrace.html#siptrace.p.trace_mode > > > > Seems to be introduced in 4.4.0 > > Exactly what I was looking for. Well, not exactly since I don't care ab

Re: [SR-Users] sql_do_query_async(): the db driver module doesn't support async query

On Sat, Aug 26, 2017 at 05:52:40PM +0300, Juha Heinanen wrote: > sqlops README tells: > > 4.5. sql_query_async(connection, query) > > Make an async SQL query using 'connection', if implemented by db > driver module (e.g., db_mysql) > > my sqlops sqlcon parameter specifies mysql driver,

Re: [SR-Users] Kamailio / Asterisk and multidomain

On Tue, Aug 22, 2017 at 02:50:13PM +0200, Cyrille Demaret wrote: > If I use this dial string in my Asterisk dialplan "PJSIP/ kamailio > /sip:2...@testdomain.com", Asterisk contact directly testcomain.com without > going through the local IP of my Kamailio. > > I can't send a domain to Kamailio in

Re: [SR-Users] sql_do_query_async(): the db driver module doesn't support async query

On Thu, Aug 31, 2017 at 09:28:46AM +0300, Juha Heinanen wrote: > > > Our solution was to switch to the mysql driver in kamailio and use the > > > galera loadbalancer > > > > In this solution, also loadbalancer must be made highly available, > > using, for example, corosync/pacemaker. > > Or run

Re: [SR-Users] Homer confusion ....

On Fri, Sep 01, 2017 at 12:50:50PM +0100, Mark Boyce wrote: > Trying to get a basic version of homer working on Kamailio V5. So far I have > sipcapture throwing everything in to a single mysql table. > > So first question; Should I be using sipcapture or siptrace? I’m not sure I > can see the

Re: [SR-Users] Kamailio and FreeSwitch integration

On Thu, Oct 19, 2017 at 12:13:47PM +0100, Timothy oladapo olawuyi wrote: > Kamailio will act as our SIP control while Freeswitch will act media server > for incoming calls. > > Freeswitch will send all outgoing calls to Kamailio for onward transfer to > our SIP provider network. > > No

Re: [SR-Users] Loadbalancing authenticated agetways

On Thu, Nov 23, 2017 at 12:14:36AM -0500, SamyGo wrote: > I've a scenario where multiple mediaservers will be registered to Kamailio > and for any incoming call from Upstream provider I've to perform > load-balancing to the actively registered media-servers. > > dispatcher module does

Re: [SR-Users] DBURL password in clear

On Wed, Nov 15, 2017 at 08:46:58AM +0100, Daniel-Constantin Mierla wrote: > > I???m working for a UK high street bank and our Kamailio implementation has > > been challenged because we???ve got database passwords held in clear in the > > configuration file. ... > > My requirement is simple, I

Re: [SR-Users] In need of a simple, stateless 302 redirect for INVITE

On Thu, Nov 30, 2017 at 09:39:24PM +, Robert wrote: > I???ve re-read the intros & tutorials, and other reading material such > as this > http://kamailio.org/docs/ser-getting-started/SER-GettingStarted.pdf > > but, this, even

Re: [SR-Users] usrloc module does not work with db_cluster

On Wed, Oct 25, 2017 at 01:31:30PM +0300, Uri Shacked wrote: > I started using db_cluster module. > DB are master-slave replication both ways. > > #!define DBURL "cluster://kamailio" > > when i stop the primary mysql server, all subscribers fails to register > again. > seems like the usrloc

Re: [SR-Users] How to add new user to Kamailio using PHP shell_exec?

On Wed, Dec 20, 2017 at 04:45:07PM +0530, Arun NV wrote: > There is a requirement to add a new user to *Kamailio* programmatically > using *PHP*. I tried to add username and password to *subscriber* table > manually using *SQL* queries, but it won't works because *ha1* and > *ha1b* should > be

Re: [SR-Users] TOPOS (5.1.[23]) breaks PRACK

On Mon, May 07, 2018 at 12:36:29PM +0200, Daniel-Constantin Mierla wrote: > can you give the logs with debug=3 for the entire call (from initial > INVITE)? Sure. Attached. Problem appears to be that the topos query can't find callid-totag (from the response). I'll try the same scenario with the

Re: [SR-Users] "Sleeping Giant - No Snore" Kam?

On Wed, May 09, 2018 at 04:52:27PM -0400, Alex Balashov wrote: > How would you define "foes" in a programmatic sense? :) That will dictate the > answer. A serious answer would be honeypots with the only function to populate blacklists (either for the permissions module or firewall rules). Just

[SR-Users] Anybody using 2 instances with topos communicating with each other?

Scenario: endpoint (bob) <-> kamailio proxy (5.1.3) <-> kamailio backend (5.1.2) Both kamailio instances are running the topos module (mysql backend). Call setup is fine, but forwarding BYEs fail. With a call from endpoint to kamailio, topos on the proxy scrambles Contact to kamailio backend.

[SR-Users] TOPOS (5.1.[23]) breaks PRACK

Enabling topos breaks PRACK in my simple (near default) proxy setup (attached kamailio.conf). asterisk 13 (10.0.3.147) <-> kamailio (10.0.3.87) <-> asterisk 13 (10.0.3.154) With the event route topos:msg-outgoing dropping all. The flow of messages is as expected. INVITE -> <- Ringing PRACK ->

Re: [SR-Users] NatHelper's Keep-alive timeout behind proxy

On Thu, May 17, 2018 at 11:43:36AM +0100, Asgaroth wrote: [OPTIONS and seperate proxy/registrar] > Does anyone have any suggestions or pointers on how we can achieve this with > the registrars being seperate from the edge proxies. > > Any pointrs/tips etc would be greatly appreciated. I noticed

Re: [SR-Users] Transport question

On Wed, May 16, 2018 at 10:59:40AM +, Wilkins, Steve wrote: > Yes, I do. It seems so strange that nothing I try will enable it to > go out over TCP. There is very little configuration to force it out > TCP. I have tried a few thing that Alex has suggested but nothing > sends it over TCP.

Re: [SR-Users] SIP Server Hardware or VM ??

On Wed, May 23, 2018 at 02:57:57PM +, Pinter, Gerd. wrote: > Thank you very much, this underpins my consideration. > How to tell my Boss :o Since I wasn't explicit in my reply: we are running rtpengine in vmware. The test I did bare metal vs. vmware was to confirm there is no problem with

Re: [SR-Users] Switching form UDP to TCP causes authentication errors

On Thu, May 24, 2018 at 11:29:54PM +, Wilkins, Steve wrote: > When I switched from UDP to TCP I started getting Authentication > Errors, Asterisk responds to an INVITE via Kamailio with a '401 > Unauthorized', but Kamailio does nothing with it. Did you get 401 responses using UDP? Do you have

Re: [SR-Users] TOPOS (5.1.[23]) breaks PRACK

On Wed, May 23, 2018 at 10:51:37PM +0200, Daniel-Constantin Mierla wrote: > I got a bit of time to look at this issue, can you try with the patch > from the next commit? > > ?? - > https://github.com/kamailio/kamailio/commit/2a3ca5942291d29de05b14338a36cecbec11f129 > This off by 1 fix solved

[SR-Users] Randomizing t_load_contacts() with same q value?

I'm looking for a way to randomize the contacts for serial forking if they have the same q value (so instead of doing parallel forking to those targets). Without taking the q value into account I could do something with the defined contacts_avp before calling t_next_contacts(). But I have no idea

Re: [SR-Users] Registration of UACs i n a partially encrypted scenario

On Tue, Jun 12, 2018 at 03:05:47PM +0200, mip FKF wrote: > We have a number of UACs in a small network which are required to > communicate without encryption because the are not able to consume > certificates. We want to use kamailio (as a proxy?) to establish an > encrypted connection to a

Re: [SR-Users] Kamailio + FreeSwitch + WebRTC

On Fri, Jun 15, 2018 at 08:21:56AM +, Pan Christensen wrote: > > I based my > > test on https://github.com/havfo/WEBRTC-to- > > SIP/blob/master/etc/kamailio/kamailio.cfg > > but I blamed my failure on an old rtpengine :) > > You need to add 'SDES-off' to the rtpengine_manage strings for calls

Re: [SR-Users] Kamailio as outbound proxy for PBX

On Thu, Jun 14, 2018 at 10:56:51AM +, Wilkins, Steve wrote: > If a PBX(Asterisk) uses an outbound_proxy (such as Kamailio), can Kamailio > actually make the SIP call? Ehhh, yes. Why wouldn't that be possible? > At some point I would like outbound calls to be controlled by Kamailio > so

Re: [SR-Users] Kamailio + FreeSwitch + WebRTC

On Thu, Jun 14, 2018 at 04:48:40PM -0300, Emanuel Gianico wrote: > From the logs I see the jssip throw this error: > > "Failed to set remote offer sdp: Called with SDP without DTLS fingerprint." > > I would like to avoid RTPEngine, because from what I understand, FreeSwitch > can handle the

Re: [SR-Users] Switching form UDP to TCP causes authentication errors

On Tue, May 29, 2018 at 11:08:06AM +, Wilkins, Steve wrote: > Thank you for your response. > > I do not have this issue with UDP, it is when I switch to TCP that an INVITE > is sending the Unauthorized. > > If I attempt something like , the code below, the INVITE is not sent out > again

Re: [SR-Users] Anybody using 2 instances with topos communicating with each other?

On Wed, May 23, 2018 at 10:58:16PM +0200, Daniel-Constantin Mierla wrote: > From the excerpts you present, there is an unexpected Record-Route with > the proxy address in the 200ok towards endpoint. Is it so or just a > mistake in what you pasted here? Because there should be no record-route >

Re: [SR-Users] Serial forking

On Mon, May 28, 2018 at 04:55:30PM +0200, igor.potjevle...@gmail.com wrote: > I know that the default behaviour of two aliases in kamailio.aliases result > in parallel forking. > > I'm wondering if its possible to do serial forking instead? For example, in > case that the first INVITE has no

Re: [SR-Users] Kamailio installation

On Wed, Jun 20, 2018 at 06:43:49PM +, Danny Roberts wrote: > I keep getting "Bad archive mirror" when trying to download the Debian > packages to install Debian 9.4. any help with this will be > appreciated. What did you do so far? https://www.kamailio.org/wiki/packages/debs still applies

Re: [SR-Users] Kamailio installation

> https://www.kamailio.org/wiki/packages/debs still applies to 9.4. http://deb.kamailio.org/ gives the same info and lists the current releases. ___ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org

Re: [SR-Users] Routing inbound call from carrier to user or voice mail

On Wed, May 02, 2018 at 05:26:07AM +, KamDev Essa wrote: > How do I : >- Associate the 10 digit DID to the 201 or 202? Can be done with (but not limited to) https://www.kamailio.org/docs/modules/5.1.x/modules/alias_db.html or a simple hardcoded line: if($rU=="0123456789") {

Re: [SR-Users] sip invite proxy-authorization

On Fri, Jan 19, 2018 at 11:59:52AM -, Dave & Hazel wrote: > MYSQL is enabled and the subscriber table is populated but I can dial > in directly without being prompted for authentication. The clue may > well be in the kamailio.cfg I've inherited. This is a near default config, the AUTH route

Re: [SR-Users] Normal behaviour with loose_routing

On Tue, Jan 30, 2018 at 11:14:18AM +0100, Loic Chabert wrote: > Thanks Daniel, > > Your snippet code must be used in proxy/registrar or on LB ? It's for the loadbalancer. The registrar also is a near default supplied config, only added support for Path to be used for REGISTERs and dispatcher.

Re: [SR-Users] Session Timers

On Tue, Jan 30, 2018 at 12:33:28PM +, Duarte Rocha wrote: > 1) I have my Kamailio between my clients CPE's and my GW to the PSTN. I > want to configure it in a way that i always try to make the CPE do the > refreshing so I can save resources on my kamailio and in the GW. If the CPE > doesn't

Re: [SR-Users] Normal behaviour with loose_routing

On Tue, Jan 30, 2018 at 10:22:39AM +0100, Loic Chabert wrote: > I have a setup configurered has diagram below: > > UAC -> Proxy LB (proxy lb) -> Proxy Router/registrar (proxy B) -> Asterisk > SBC > > My UAC is SIPP and sending a classic call scenario: INVITE - ACK - BYE. > > But i have some

Re: [SR-Users] Kamailio mod ALIAS_DB: multiple aliases

On Tue, Feb 20, 2018 at 09:04:14AM +0100, Stefan R??etschli wrote: > So, if the user uses an alias which does not belong to him, i want to > overwrite it to the first alias for his username in the aliases-db. > At the moment, the function alias_db_find returns me the last alias found in > the db.

Re: [SR-Users] User state

On Fri, Feb 16, 2018 at 09:33:07PM +, Magnus L??fqvist wrote: > In asterisk I would use hints and listing in events in AMI. But what > way are the best in Kamailio? A way to do this is by using the dialog module. Create a profile with value and store either $au or $rU depending on direction.

Re: [SR-Users] Inserting a message into a transaction?

On Tue, Feb 20, 2018 at 05:41:49PM +0100, Emmanuel BUU wrote: > sl_send_reply("180", "Make this retarded endpoint ring"); Well, this is embarrassing :) This is so obvious that I didn't even try. I was under the impression that a send_reply in on_reply would send to the source of the reply.

[SR-Users] Inserting a message into a transaction?

For a retarded endpoint (an old ISDN PBX behind a Smartnode) I need to insert a 180 reply before sending a 183 (with SDP), otherwise the PBX fails to playback the inband audio to the caller behind the PBX. I can't find hints on how to do this. uac_req_send() can be used to start a new

Re: [SR-Users] Handling redirect replies and send to new host

On Wed, Dec 27, 2017 at 03:33:16PM +, voipspace voipspace wrote: > (A) On receipt of the 300 response I do not want this redirect to go back to > the user. > > onreply_route { > if (t_check_status("3[0-9][0-9]")) { > // How do I send ACK to (B) to finish this

Re: [SR-Users] Critical Error

On Thu, Dec 28, 2017 at 04:21:28PM +, Nicolas Breuer wrote: > > I see this : > > > SIP/2.0 183 Session Progress. > CSeq: 1258951486 INVITE. > Require: 100rel. > > SIP/2.0 180 Ringing. > CSeq: 1258951486 INVITE. > Require: 100rel. > > PRACK sip:3222430244@80.84.31.129:5060 SIP/2.0. > CSeq:

Re: [SR-Users] New blog article: Kamailio as an SBC - five years on

On Tue, Jun 19, 2018 at 01:18:49AM -0400, Alex Balashov wrote: > This is an update to my original 2013 article on this topic: > > http://www.evaristesys.com/blog/kamailio-as-an-sbc-five-years-on/ Thank you for sharing your insights. Over the past 5 years I stumbled over your (and other

Re: [SR-Users] Kamailio installation

On Thu, Jun 21, 2018 at 09:08:53PM +, Danny Roberts wrote: > I am very new to this. What???s the command to install the application. Those were in my mail you responded to and thanks to TOFU are still below: [snip] > So get the public key for the repo: > wget -O-

Re: [SR-Users] Ratio between cores ram and kamailio limits

On Fri, Aug 10, 2018 at 04:59:21PM +0200, Renato Gallo wrote: > We're here to ask you what's the ratio between cpu cores/ghz, ram and > how many contemporary calls kamailio can manage before it maxes out > its capabilities and crash. You will not get any definite answer to this question. It

Re: [SR-Users] What is the typical network setup for kamailio?

On Thu, Aug 16, 2018 at 11:57:03AM +0200, Kevin Olbrich wrote: > Most examples show that Asterisk is deployed on the same network as the > external interface of Kamailio (-> Asterisk exposed to the public network). > In our tests, this works much better but I have great security concerns > because

Re: [SR-Users] Migrate from RtpProxy to Rtpengine

On Tue, Aug 14, 2018 at 06:41:32AM +, Nicolas Breuer wrote: > No. Using RTPengine on Kamailio. No asterisk behind. I think subst_body is > not supported. > > Someone can confirm ? Sounds like you are making changes to the body before the rtpengine mangling is commit. See

Re: [SR-Users] Can Kamailio interface with an asterisk server based on the fqdn of a call?

On Mon, Aug 13, 2018 at 05:48:49PM +, Wilkins, Steve wrote: > Is it possible for Kamailio to interface with a particular Asterisk > Server based on the FQDN of a caller? Yes. > I would like to pass calls received by Kamailio through to different > Asterisk Servers based on the FQDN of the

Re: [SR-Users] Migrate from RtpProxy to Rtpengine

On Tue, Aug 14, 2018 at 08:23:28AM +, Nicolas Breuer wrote: > substr > msg_apply_changes() > rtpengine_manage() > > > This will not work because msg_apply_changes is working only in request route. > I do substr in reply routes also. From documentation: "This function can be used from

Re: [SR-Users] Registered User limit

On Thu, Aug 16, 2018 at 07:05:09PM +0530, Pravin . wrote: > i have one qiery, Please don't hijack threads with these sudden topic changes. People not interested in a registrion limit will not read your message. > does Kamailio SIP server complied with lawful intercept , What is layful

Re: [SR-Users] UDP workers block when one or more rtpengine instances go offline

On Mon, Aug 06, 2018 at 12:58:00PM +0200, Sebastian Damm wrote: > we run multiple rtpengine servers to share the load. Whenever we need > to take an rtpengine server offline, we used to just block the control > port via iptables, then no new calls ended up on this instance of > rtpengine. This

Re: [SR-Users] UDP workers block when one or more rtpengine instances go offline

On Tue, Aug 07, 2018 at 02:49:58PM +0200, Sebastian Damm wrote: > Oh, and we tested the disabling and enabling via kamctl before, but as > far as I remember, while disabling still worked, Kamailio crashed > reproducably when enabling an rtpengine again. We had the same problem, this was fixed in

Re: [SR-Users] UDP workers block when one or more rtpengine instances go offline

On Mon, Aug 06, 2018 at 02:37:15PM +0200, Sebastian Damm wrote: > > kamcmd rtpengine.enable udp:x.y.z.a:port 0/1 > > on the kamailio machines? > > Of course, that's probably the better way. But as far as I know, that > command wasn't available before 5.0. So I guess our blocking of the > control

Re: [SR-Users] Upgrade from 4.4 to 5.1

On Mon, Aug 13, 2018 at 02:16:36PM +, Nicolas Breuer wrote: > Just put version_table="version44" in Kamailio.conf ? I tried ammend my message, but (hopefully) more clear: version_table points to an alternate table in the database that contains the versions of the other tables. If both

Re: [SR-Users] Upgrade from 4.4 to 5.1

On Mon, Aug 13, 2018 at 03:23:29PM +0200, Daniel Tryba wrote: > version_table="version44" > > Set this directive in the 4.4.x machines. Offcourse after copying it from the standard versions table. signature.asc Description: PGP signature _

Re: [SR-Users] Upgrade from 4.4 to 5.1

On Mon, Aug 13, 2018 at 10:52:59AM +, Nicolas Breuer wrote: > I want to upgrade my both Kamailio from 4.4 to 5.1 > I have to upgrade the SQL structure before running the 5.1 release. > > But my SQL servers are synchronized - Kamailio 4.4 can works with the 5.1 SQL > structure ? If there are

Re: [SR-Users] UDP workers block when one or more rtpengine instances go offline

On Mon, Aug 06, 2018 at 02:37:15PM +0200, Sebastian Damm wrote: > Of course, that's probably the better way. But as far as I know, that > command wasn't available before 5.0. So I guess our blocking of the > control port was the way we did it before that. > > But anyhow, if an rtpengine crashes,

Re: [SR-Users] Kamailio 5.1 Nathelper sipping on different transport protocols.

On Mon, Aug 20, 2018 at 07:03:22PM +0300, ?? ?? wrote: > It works fine if client is connected over UDP. But as soon as I will switch > to TCP or register with client that uses WS, Kamailio stops generating > OPTIONS. > > Does anyone know if it is correct behavior or

Re: [SR-Users] Struggling with RTPProxy and RTPEngine

On Wed, Aug 22, 2018 at 12:49:54PM +, Wilkins, Steve wrote: > My offer and answer => > rtpengine_offer("trust-address replace-session-connection replace-origin"); If this is really your config you should change the offer to answer for the answer part of the config. Either use rtpengine_offer

Re: [SR-Users] Struggling with RTPProxy and RTPEngine

On Wed, Aug 22, 2018 at 05:05:02PM +, Wilkins, Steve wrote: > The SIP traffic is working this way for me but I still see RTP traffic going > directly from Asterisk to the UAC, which means they need to whitelist > asterisk IP. Am I missing something? In what sense do they need whitelisting?

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