- sr-users - kamailio.org

Re: [SR-Users] TOPOS (5.1.[23]) breaks PRACK
by Daniel Tryba 02 May '23

02 May '23

On Mon, May 07, 2018 at 04:44:14PM +0200, Daniel Tryba wrote: > Sure. Attached. Problem appears to be that the topos query can't find > callid-totag (from the response). > > I'll try the same scenario with the mysql backend to see if it behaves > different. Config works fine with mysql as topos backend. So the bug is restricted to topos-redis.

5 19

Loading the LCR Module Tables at startup/reload
by harneet singh 30 Apr '23

30 Apr '23

Hi All, Can someone point to a sample as to how we can *load the LCR Module tables* namely the following at startup from a csv file or any other mechanism. * lcr_gw_table * lcr_rule_table * lcr_rule_target_table The idea is to not need to manually feed in the entries into the table but instead add these in the beginning through either a *wrapper script *OR *through the kamailio cfg file* if possible. Any pointers are much appreciated. Thanks & Regards, Harneet Singh -- "Once you eliminate the impossible, whatever remains, no matter how improbable, must be the truth" - Sir Arthur Conan Doyle

2 3

Route Header based routing for initial INVITE
by harneet singh 28 Apr '23

28 Apr '23

Hi Experts, I believe the loose_route() in kamailio can be used for initial Out of the dialog INVITE processing as well. Is that the correct understanding? If so, the request will be forwarded ahead to the destination corresponding to the Route Header remaining (if any) after Kamailio has stripped off it's own Address(es) from the Route Header(s). Is that correct? Regards, Harneet Singh -- "Once you eliminate the impossible, whatever remains, no matter how improbable, must be the truth" - Sir Arthur Conan Doyle

3 3

Strange RTCP issue
by Kiss Zoltán 27 Apr '23

27 Apr '23

Hello, We have a transparent proxy in front of Asterisk boxes. The media is handled by rtpengine, and the "manage" work is done by Kamailio in the route[NATMANAGE] section like this: if(nat_uac_test("8")) { # Packet from backend server if(ds_is_from_list()){ if (rtpengine_manage("replace-origin replace-session-connection direction=internal direction=pub")) { xdbg("NATMANAGE --- Success rtpengine_manage() RTP from INTERNAL to EXTERNAL $ru"); } } else { if (rtpengine_manage("replace-origin replace-session-connection direction=pub direction=internal")) { xdbg("NATMANAGE --- Success rtpengine_manage() RTP from EXTERNAL to INTERNAL $ru"); } } } else { # Packet from backend server if(ds_is_from_list()){ if (rtpengine_manage("replace-origin replace-session-connection trust-address direction=internal direction=pub")) { xdbg("NATMANAGE --- Success rtpengine_manage() RTP from INTERNAL to EXTERNAL $ru"); } } else { if (rtpengine_manage("replace-origin replace-session-connection trust-address direction=pub direction=internal")) { xdbg("NATMANAGE --- Success rtpengine_manage() RTP from EXTERNAL to INTERNAL $ru"); } } } Everything is working fine, but with some clients (like Grandstream phone) the RTCP session wants to go tot he private address of the phone. Here is the log of one of these strange calls: Apr 27 16:54:38 rtp1 rtpengine[2273]: INFO: [312ed76c31f21b71452e91e5184ad25b@172.16.2.210:5060]: [core] --------- Port 178.238.213.14:11088 <> 81.183.216.3:5068 , SSRC 77c19488, 625 p, 107500 b, 0 e, 29 ts Apr 27 16:54:38 rtp1 rtpengine[2273]: INFO: [312ed76c31f21b71452e91e5184ad25b@172.16.2.210:5060]: [core] --------- Port 178.238.213.14:11089 <> 10.0.5.192:5069 (RTCP), SSRC 0, 0 p, 0 b, 0 e, 43 ts As you can see the RTP itself is okay, but the RTCP will somehow go to the private address. If we tracing the sdp messages in sngrep, then we can see that Kamailio transforming IP addresses for the backend servers to the public (and the private address of the rtpengine) addresses. We are missing something? Thank you! With kind regards, Zoltan

2 1

Sensible values for pike mudule?
by Benoit Panizzon 27 Apr '23

27 Apr '23

Hi Before I go too deep into try and error, I guess others have been there too. What are sensible value you use for the pike module to detect / mitigate abusive behavior, especially dictionary attacks? Are there better solutions that the pike module? Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

2 1

Fwd: Sensible values for pike mudule?
by Stefan 27 Apr '23

27 Apr '23

Hi Benoît We use pike in combination with fail2ban. Log a message -> create a fail2ban rule to block the ip for xxx time. example: if ($ua =~ "(friendly-scanner|sipvicious)") { sl_send_reply("488", "Go away."); xlog("L_INFO","$ci Blocking traffic from $si\n"); exit; } Cool thing is, you can write a log entry from every part in your kamailio-config, and then fail2ban does the rest. Regards Stefan Am 27.04.23 um 16:07 schrieb Benoit Panizzon: > Hi > > Before I go too deep into try and error, I guess others have been there > too. > > What are sensible value you use for the pike module to detect / > mitigate abusive behavior, especially dictionary attacks? > > Are there better solutions that the pike module? > > Mit freundlichen Grüssen > > -Benoît Panizzon-

1 0

hooking mechanism for routing blocks
by Valentin Christoph 27 Apr '23

27 Apr '23

Dear all, I am looking for some hooking mechanism for user defined routing blocks. Is something like this already available? 1) a common part of the routing configuration shall not (or only slightly) be changed, after some hooks have been implemented Something like a function "engage_hooks()" of some "hook_module"? : Routing config that should not be changed by application developers engage_hooks(HOOK_TYPE); Routing config that should not be changed by application developers : 2) one application developer may "hook" new parts of the routing configuration into this, by modparam("hook_module", "hook", "name=MY_RB;type=HOOK_TYPE;prio=50); : : Route [MY_RB] { Do something at hook HOOK_TYPE; } : : 3) another application developer might want to "hook" into the same HOOK_TYPE with a higher priority modparam("hook_module", "hook", "name=ANOTHER_RB;type=HOOK_TYPE;prio=20); : : Route [ANOTHER_RB] { Do another thing with higher priority at hook HOOK_TYPE } : : Kr, Christoph

2 2

Kamailio use case
by jdprsaga＠gmail.com 26 Apr '23

26 Apr '23

Hello All, new subscriber here. I've been reading kamailio documentation to check if kamailio has a use case on the architecture for the solution for my company's software, here an small architecture diagram of our solution. sipclient <----> [ Middleware ] <-----> SBC Our middle ware is both a sip proxy and media proxy. we have a large number of sip clients but a small calling number, our current bottleneck is that the middleware can't handle a high number of registration, i would like to deploy Kamailio as a load balancer for registrations, making sure it is statefull so registrations and calls are hashed into the same middleware server. sipclients < ----> [ kamailio ] <------> [ middleware1, middleware2,...,middleware N] <-----> SBC Any thoughts or pointers?

4 3

rtpengine module: Actions caused by rtpengine_manage() on reply-codes?
by Benoît Panizzon 26 Apr '23

26 Apr '23

Hi all I wonder, if there is a documentation on what action is performed by rtpengine_manage() on which kind reply. Am I right, that rtpengine_manage(), on any 4XX reply causes a delete? So if there is a re-invite (for example to switch to T.38) and the other party replies with 488 which in turn is passed through rtpengine_manage() the existing RTP stream is deleted from rtpengine instead of being left running with the previously agreed codec? Has anyone already figured out a list of what reply codes should be exempt from calling rtpengine_manage()? -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

8 12

Configuring dynamic Destination identification for Kamailio Routing
by harneet singh 25 Apr '23

25 Apr '23

Hi Experts, We have been using Kamailio as a sip proxy for a fair bit of time now(version 5.3.2 though) and do understand the native routing mechanism that can be instrumented through the kamailio cfg scripts. We initially had only two sides, where the traffic could come from into Kamailio and it just passed through to the other side(ie Traffic from A was routed to B and that from B was routed to A). Off late, a few more sip elements have been introduced in the deployment and now there is a need for a more formal/robust mechanism to select the intended destination to forward an initial SIP INVITE to. I believe that we could still keep all the logic in the native kamailio script and do the routing(for example, an incoming call where the *From URI* contains *sip.elementA.com <http://sip.elementA.com>* should be routed to destination *sip.elementB.com <http://sip.elementB.com>*, where as if the* To URI *contains *sip.elementC.com <http://sip.elementC.com>*, the request should be routed to *sip.elementC.com <http://sip.elementC.com>)**. *Since we have many sip elements like A,B.C,D,E etc where the requests might have to be routed to, based on certain different criteria, this logic would become cumbersome. *What would be the best way to accommodate this?* *Would it be possible to rewire this routing logic at runtime(*like in some csv file that can be reloaded*) without a Kamailio restart? * Regards, Harneet Singh -- "Once you eliminate the impossible, whatever remains, no matter how improbable, must be the truth" - Sir Arthur Conan Doyle

3 5

sips to sip with TLS proxy
by Kiss Zoltán 25 Apr '23

25 Apr '23

Hi all, We have a working Kamailio setup, lets call it a transparent proxy for Asterisk boxes. Its based on domain and dispatcher modules and everything is working as expected with the test clients (more or less microsip, softphone for ios, etc). We are tried to register with a Grandstream deskphone today, and we see that the phone sending sips:xxx in the Reg Contact field for example. Because the sips schema, the register is working, but we cannot initiate calls from this phone. If we are turning SIP scheme to sip from sips in the phone, then everything is working as expected. I think we can transform those requests from sips to sip with Kamailio, but currently we dont know where can we start. Has anybody a suggestion about this issue? I know that we can transform ruri, contact, etc with textops, nathelper and a lot of other modules, but what is the best for this sips->sip translation? Thanks for your help. With kind regards, Zoltan

4 10

TCP/TLS connection (id: 0) for WebSocket could not be found
by Joey Golan 22 Apr '23

22 Apr '23

Hello, I have 2 clients. 1 using websocket and 1 mobile using TLS. Both can call each other perfectly. After 24-48 hours calls from mobile to websocket failed with an error message: TCP/TLS connection (id: 0) for WebSocket could not be found. At the same time I can make calls from the websocket client to the mobile. I know that the search of the connection is done by target address so I printed the addresses + ports and they are matched to the ones that appear on the ul.dump and ws.dump. *kamcmd ul.dump output:* Info: { AoR: usrmt63ly_1040 HashID: 1830036546 Contacts: { Contact: { Address: sip:usrmt63ly_1040@example.com;transport=ws Expires: 283 Q: -1 Call-ID: tvne1pskgkm2a01pnqm677 CSeq: 1060 User-Agent: JsSIP 3.9.0 Received: sip:*10.218.51.86:24061 <http://10.218.51.86:24061>*;transport=ws Path: <sip:10.218.154.228:50443;transport=ws;received=sip:*10.218.51.86:24061 <http://10.218.51.86:24061>*;transport=ws;lr> State: CS_NEW Flags: 0 CFlags: 64 Socket: tls:10.218.154.228:50443 Methods: 7071 Ruid: uloc-61ebba84-64242101-89ba-1 Instance: <urn:uuid:7573726d-7436-116c-915f-313034304077> Reg-Id: 1 Server-Id: 1642838660 Tcpconn-Id: 100420 Keepalive: 1 Last-Keepalive: 1680417282 KA-Roundtrip: 0 Last-Modified: 1680417282 } } } *kamcmd ws.dump* { connections: { 100420: wss:*10.218.51.86:24061 <http://10.218.51.86:24061>* -> wss: 10.218.154.228:50443 (state: OPEN, last used 14s ago, sub-protocol: sip) } info: { wscounter: 1 truncated: no } } *Call from mobile (TLS) to websocket: Failed with error message: TCP/TLS connection (id: 0) for WebSocket could not be found* 30(35263) INFO: {1 2563 INVITE 4248c155-cd92-49b8-b68a-eef5f2cb3e88} <script>: request_route: method [INVITE] from [ sip:usrrcv4o4_1040@example.com] socket [sip:10.218.51.86:50909;transport=tls] to [sip:usrmt63ly_1040@example.com] 30(35263) INFO: {1 2563 INVITE 4248c155-cd92-49b8-b68a-eef5f2cb3e88} <script>: [RELAY]: method [INVITE] from [sip:usrrcv4o4_1040@example.com] socket [sip:10.218.51.86:50909;transport=tls] to [ sip:usrmt63ly_1040@example.com] socket [sip:*10.218.51.86:24061 <http://10.218.51.86:24061>*;transport=ws] 30(35263) WARNING: {1 2563 INVITE 4248c155-cd92-49b8-b68a-eef5f2cb3e88} <core> [core/msg_translator.c:2863]: *via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found* 30(35263) ERROR: {1 2563 INVITE 4248c155-cd92-49b8-b68a-eef5f2cb3e88} <core> [core/msg_translator.c:2040]: build_req_buf_from_sip_req(): could not create Via header 30(35263) ERROR: {1 2563 INVITE 4248c155-cd92-49b8-b68a-eef5f2cb3e88} tm [t_fwd.c:479]: prepare_new_uac(): could not build request 30(35263) ERROR: {1 2563 INVITE 4248c155-cd92-49b8-b68a-eef5f2cb3e88} tm [t_fwd.c:1759]: t_forward_nonack(): failure to add branches 30(35263) ERROR: {1 2563 INVITE 4248c155-cd92-49b8-b68a-eef5f2cb3e88} sl [sl_funcs.c:418]: sl_reply_error(): stateless error reply used: No error (2/SL) 30(35263) NOTICE: {1 2563 INVITE 4248c155-cd92-49b8-b68a-eef5f2cb3e88} acc [acc.c:287]: acc_log_request(): ACC: call missed: timestamp=1680417920;method=INVITE;from_tag=e168ffa6-e577-48ec-b9c9-a73f5be318e8;to_tag=0c3e625e9f223a7ee2b093e8e31b2526-b7165bcb;call_id=4248c155-cd92-49b8-b68a-eef5f2cb3e88;code=500;reason=Server Internal Error;src_user=usrrcv4o4_1040;src_domain=example.com ;src_ip=10.218.51.86;dst_ouser=usrmt63ly_1040;dst_user=usrmt63ly_1040;dst_domain= example.com *Call from websocket to mobile (TLS):* 38(35271) INFO: {1 7488 INVITE urcnete8r71dpkhmngd8} <script>: request_route: method [INVITE] from [sip:usrmt63ly_1040@example.com] socket [sip:*10.218.51.86:24061 <http://10.218.51.86:24061>*;transport=ws] to [ sip:usrrcv4o4_1040@example.com] 38(35271) INFO: {1 7488 INVITE urcnete8r71dpkhmngd8} <script>: [RELAY]: method [INVITE] from [sip:usrmt63ly_1040@example.com] socket [sip:*10.218.51.86:24061 <http://10.218.51.86:24061>*;transport=ws] to [ sip:usrrcv4o4_1040@example.com] socket [sip:10.218.51.86:20674 ;transport=tls] 39(35272) INFO: {2 7488 INVITE urcnete8r71dpkhmngd8} <script>: onreply_route[MANAGE_REPLY] method [INVITE] code [100] from [ sip:usrmt63ly_1040@example.com] to [sip:usrrcv4o4_1040@example.com] 39(35272) INFO: {2 7488 INVITE urcnete8r71dpkhmngd8} <script>: onreply_route[MANAGE_REPLY] method [INVITE] code [200] from [ sip:usrmt63ly_1040@example.com] to [sip:usrrcv4o4_1040@example.com] 39(35272) NOTICE: {2 7488 INVITE urcnete8r71dpkhmngd8} acc [acc.c:287]: acc_log_request(): ACC: transaction answered: timestamp=1680417926;method=INVITE;from_tag=18q0p37nfo;to_tag=a88272a0-25ca-43c2-8ce9-8615831b7e9a;call_id=urcnete8r71dpkhmngd8;code=200;reason=OK;src_user=usrmt63ly_1040;src_domain= example.com ;src_ip=10.218.51.86;dst_ouser=usrrcv4o4_1040;dst_user=usrrcv4o4_1040;dst_domain=10.218.51.86 38(35271) INFO: {1 7488 ACK urcnete8r71dpkhmngd8} <script>: request_route: method [ACK] from [sip:usrmt63ly_1040@example.com] socket [sip:*10.218.51.86:24061 <http://10.218.51.86:24061>*;transport=ws] to [ sip:usrrcv4o4_1040@example.com] 38(35271) INFO: {1 7488 ACK urcnete8r71dpkhmngd8} <script>: [RELAY]: method [ACK] from [sip:usrmt63ly_1040@example.com] socket [sip:*10.218.51.86:24061 <http://10.218.51.86:24061>*;transport=ws] to [ sip:usrrcv4o4_1040@example.com] socket [sip:10.218.51.86:20674 ;transport=tls] *Version: kamailio 5.6.2 (x86_64/linux)* What could be the reason for that? Your help would be much appreciated. Thanks, Joey.

3 4

How to log expiring dialogue?
by Benoît Panizzon 21 Apr '23

21 Apr '23

Hi alle I think I have an issue with the dialogue and session timer module with Kamailio 5.4 I sync the Session Expire Timer with the dialogue timeout via a common AVP. When I check a dialogue via kamcmd dlg.list start_ts: 1682066352 init_ts: 1682066346 end_ts: 0 timeout: 1682066952 lifetime: 600 Timeout is increased on the first session refreshing re-invite, as expected (in this case after 300 seconds). But it looks like the dialogue just disappears (from kamcmd dlg.list, not from the backing database) when the lifetime ist reached, leaving an orphan database entry. I would like to digg deeper into this. Is there a way to log more details from the dialogue module like when it triggers dialogue expiry? -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

2 2

Roadmap to next major release series v5.7.x
by Daniel-Constantin Mierla 20 Apr '23

20 Apr '23

Hello, it is time to make the roadmap to the next major release series v5.7.x. In my opinion, I would try to freeze the development during the first part of April 2023, tentatively proposing April 14 as freezing date, being like 4 weeks from now. Then testing phase and first release (v5.7.0) sometime during May 2023. Alternative timelines can be proposed, of course. If anyone wants to add new features/modules, they have to be published till freezing date, either pushed in the git repository or proposed as pull request. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com Kamailio Advanced Training - Online - March 27-30, 2023 - www.asipto.com

3 5

Rtpengine: no audio after kernelization.
by Michel Pelletier 17 Apr '23

17 Apr '23

Hello, I am proxying all RTP through RTPEngine. Everything works fine until about 5 seconds into the call, when rtpengine enters kernelization, after which all RTP forwarding ceases. I've checked the required iptables entries, and all looks good. Here is a description of my environment: # cat os-release PRETTY_NAME="Debian GNU/Linux 11 (bullseye)" NAME="Debian GNU/Linux" VERSION_ID="11" VERSION="11 (bullseye)" VERSION_CODENAME=bullseye ID=debian HOME_URL="https://www.debian.org/" SUPPORT_URL="https://www.debian.org/support" BUG_REPORT_URL="https://bugs.debian.org/" # kamailio -v version: kamailio 5.6.2 (x86_64/linux) 54a9c1 flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, 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_BLOCKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED 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: 54a9c1 compiled on 16:02:49 Dec 1 2022 with gcc 10.2.1 # rtpengine -v Version: 11.1.1.3-1~bpo11+1 Michel Pelletier

4 13

Docker repository removed
by Manel Villar 17 Apr '23

17 Apr '23

Hi, I was using the kamailio/kamailio docker repository in one of my projects, and now the image is not available as the repo has been removed. Is the "kamailio-ci" repo a replacement for it? Thanks! -- Manel Villar Steegerstr. 59 13359 Berlin Germany Tel: (+49) (0)162 6574604 skype: manel.villar manelvf(a)gmail.com *"Ars longa, vita brevis"*

4 5

Development for v5.7.x is frozen
by Daniel-Constantin Mierla 17 Apr '23

17 Apr '23

Hello, the formal notification that the development for the next major version 5.7.0 is now frozen. The focus has to be on testing the master branch. Also, the master branch should not get commits with new features till the branch 5.7 is created, expected to happen in 2-4 weeks, a matter of how testing goes on. Meanwhile, the commits with new features in the C code can be pushed to personal branches, new pull requests can still be done, but they will be merged after branching 5.7. Can still be done commits with documentation improvements, enhancements to related tools (e.g., kamctl, kamcmd), merging exiting pull requests at this moment, exporting missing KEMI functions and completing the functionality of the new modules added for 5.7. Once the branch 5.7 is created, new features can be pushed again to master branch as usual. From that moment, the v5.7.0 should be out very soon, time used for further testing but also preparing the release of packages. If someone is not sure if a commit brings a new feature, just make a pull request and it can be discussed there on github portal or via sr-dev mailing list. A summary of what is new in upcoming 5.7 is going to be built at: * https://www.kamailio.org/wikidocs/features/new-in-5.7.x/ Upgrade guidelines will be collected at: * https://www.kamailio.org/wikidocs/install/upgrade/5.6.x-to-5.7.0/ Everyone is more than welcome to contribute to the above wiki pages, especially to the upgrade guidelines, to help everyone else during the migration process from v5.6.x to 5.7.x. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

2 2

Assigning a new RTPEngine for a call for the sake of failover
by mehdithreem+kamailio＠gmail.com 15 Apr '23

15 Apr '23

Hello, I am trying to implement RTPEngine failover, where if a selected RTPEngine node fails in the middle of the call (and not responding NG requests), I want to select an alternative node. However, when I call rtpengine_manage(), it does not seem to select an alternative node in case of a failure. To address this, I have attempted to check if $avp(RTP_INSTANCE) is null after calling rtpengine_manage(), and then use rtpengine_delete() to forget the selected RTPEngine node for the specific call, with the expectation that Kamailio would assign a new RTPEngine node for subsequent SIP message retries. However, it appears that rtpengine_delete() does not have the desired effect in this regard. Here is an excerpt of my code for reference: ``` route[RTP] { #!ifdef WITH_RTPENGINE if (has_body("application/sdp")) { if ($avp(x_source) == "INTERNAL") { rtpengine_manage("replace-session-connection replace-origin direction=internal direction=external"); } else if ($avp(x_source) == "EXTERNAL") { rtpengine_manage("replace-session-connection replace-origin direction=external direction=internal"); } if ($avp(RTP_INSTANCE) == $null) { rtpengine_delete(); drop(); } } #!endif } ``` Can you please advise on how I can achieve the desired failover behavior? Is there another function similar to `rtpengine_reset_call` that can make Kamailio forget the RTPEngine node for a specific call? Thank you!

2 1

CLI & destination based LCR
by John Cahill 14 Apr '23

14 Apr '23

Hi, I'd like to build an LCR solution that routes on both CLI & destination e.g. CLI matches ^\+49*, destinations matches ^\+44* => route to gateway 5 for a large rule set (~ tens of thousands). Any recommendations for which modules to use? Can kamailio perform this particular task well or should I look at another solution? I'd be very grateful to hear about any experience of doing this. Many Thanks. Regards, John

5 5

Kamailio + PostgreSQL
by georgy.grigoryev＠snapcom.fr 14 Apr '23

14 Apr '23

Hello everyone, I'm trying to connect Kamailio 5.5.5 to my PostgreSQL 14. They are on two different servers. I have created the Kamailio DB on the Postgre with the help of the command "kamdbctl create" I'm able to connect to the database with no problem. The problem is in the kamailio.cfg file, i have added this so far: #!define WITH_PGSQL #!ifdef WITH_PGSQL #!define DBURL "postgres://USER:PASSWORD@IP:POPT/kamailio" #!endif #!ifdef WITH_PGSQL loadmodule "db_postgres.so" #!endif modparam("db_postgres", "retries", 3) modparam("db_postgres", "timeout", 10) modparam("db_postgres", "tcp_keepalive", 600) modparam("db_postgres", "lockset", 6) modparam("db_postgres", "bytea_output_escape", 0) But my kamailio doenst like this configuration, can someone help me find the problem? Thank you in advance! Best Regards!

3 2

manipulate Authentication header
by David Villasmil 13 Apr '23

13 Apr '23

Hello guys, I can't find a simple way of changing the username and recalculate the has in a REGISTER... do i have to use textops? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 1

How to correctly use the SST module?
by Benoît Panizzon 11 Apr '23

11 Apr '23

Hi I still have odd behaviour with the SST module: modparam("sst", "min_se", 300) if (sstCheckMin("1")) { xlog("L_ERR", "$cfg(route) Session Timer too small\n"); exit; } setflag(FLT_DLG); # set the dialog flag setflag(FLT_SST); # Set the sst flag if(is_present_hf("Session-Expires")) { $avp(dlgtimeout) = $(hdr(Session-Expires){s.int}); } On an INVITE containing those header: Session-Expires: 3600;refresher=uac Min-SE: 600 Supported: 100rel,timer Expires: 330 Now Kamailio replies: SIP/2.0 422 Session Timer Too Small Min-SE: 300 What am I doing wrong? -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

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dlg_onroute(): unable to find dialog for INVITE with route param
by Benoît Panizzon 10 Apr '23

10 Apr '23

Happy Eastern Kamailio 5.4 used. On dialogues calls, I noticed this happening on the second re-invite to refresh the session from the B side. dlg_onroute(): unable to find dialog for INVITE with route param 'df9.da75' [2557:22445] and call-id '14148a68-f1b2-485d-b5a2-f9dd7773d2c0' The did df9.da75 is identical to the one in the route-header on the first re-invite from the B side, before this one. No CDR was (yet) generated, and the dialogues is still present and on-going according to the database: CallID [14148a68-f1b2-485d-b5a2-f9dd7773d2c0] H-Entry: 2557 H-ID: 22445 But most probably not in memory as kamcmd dlg.stat_active shows zero (this is on a development node, so no other calls running). There are no sip messages that end the call in between those two re-invites. I see no session timeouts or similar in the logs. Any hint what could cause Kamailio not to find the corresponding dialogue on this re-invite? -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

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Fwd: Kamailio + PostgreSQL
by Stefan 07 Apr '23

07 Apr '23

Hi Give it a try without modparams to see if it works, I have the same config but without modparams and it runs without problems. Else have a look into the kamailio logfile for errors. Or check config with "kamailio -c"? Regards Stefan Am 06.04.23 um 15:18 schrieb georgy.grigoryev(a)snapcom.fr: > Hello everyone, > > I'm trying to connect Kamailio 5.5.5 to my PostgreSQL 14. > They are on two different servers. > I have created the Kamailio DB on the Postgre with the help of the > command "kamdbctl create" > I'm able to connect to the database with no problem. > The problem is in the kamailio.cfg file, i have added this so far: > > #!define WITH_PGSQL > > #!ifdef WITH_PGSQL > #!define DBURL "postgres://USER:PASSWORD@IP:POPT/kamailio" > #!endif > > #!ifdef WITH_PGSQL > loadmodule "db_postgres.so" > #!endif > > modparam("db_postgres", "retries", 3) > modparam("db_postgres", "timeout", 10) > modparam("db_postgres", "tcp_keepalive", 600) > modparam("db_postgres", "lockset", 6) > modparam("db_postgres", "bytea_output_escape", 0) > > But my kamailio doenst like this configuration, can someone help me > find the problem? > > Thank you in advance! > > Best Regards! > __________________________________________________________ > Kamailio - Users Mailing List - Non Commercial Discussions > To unsubscribe send an email to sr-users-leave(a)lists.kamailio.org > Important: keep the mailing list in the recipients, do not reply only > to the sender! > Edit mailing list options or unsubscribe:

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Malformed To Header
by Serdar GUCLUER 06 Apr '23

06 Apr '23

Hello all, In my situation, it is possible to call forwarding to registered users multiple times on same leg. For each forward i have to update To uri and i am using uac_replace_to to do that. But on second relaying, To uri was malformed. On below, Im attaching configuration to illustrate my problem. (version-5.4.9) First relay - /To: //sip:xyz@192.168.0.51/ Second relay - /To: sip:xyz@192.168.0.51sip:abc@192.168.0.51/ Thanks in advance. Serdar /loadmodule "uac.so"// //modparam("uac","restore_mode","auto")// // // //request_route {// // ...// // $avp(x) = "abc";// // $avp(x) = "xyz";// // $avp(myip) = "192.168.0.51";// // $avp(i) = 0;// // route(SERDAR);// // ...// //}// // //route[SERDAR] {// // if ($(avp(x)[$avp(i)])!=$null) {// // $avp(destination) = $(avp(x)[$avp(i)]);// // $avp(i) = $avp(i) + 1;// // $rU=$_s($avp(destination));// //uac_replace_to("$_s(sip:$avp(destination)@$avp(myip))");// // route(LOCATION);// // } else {// // send_reply("403", "Opps!");// // exit;// // }// //}// // //route[LOCATION] {// // $avp(oexten) = $rU;// // if (!lookup("location") ) {// // t_newtran();// // switch ($var(rc)) {// // case -1:// // case -3:// // send_reply("480", "Temporarily Unavailable");// // exit;// // case -2:// // send_reply("405", "Method Not Allowed");// // exit;// // }// // }// // t_on_failure("SERDAR_FAILURE");// // if(!t_relay()) sl_reply_error();// // exit;// //}// // //failure_route[SERDAR_FAILURE] {// // route(NATMANAGE);// // if (t_is_canceled()) //exit;// // route(SERDAR);// //}/

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