- sr-users - kamailio.org

Kamailio v5.1.4 Released
by Daniel-Constantin Mierla 05 Jun '18

05 Jun '18

Hello, Kamailio SIP Server v5.1.4 stable release is out. This is a maintenance release of the latest stable branch, 5.1, that includes fixes since the release of v5.1.3. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.1.x. Deployments running previous v5.1.x versions are strongly recommended to be upgraded to v5.1.4. For more details about version 5.1.4 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2018/06/kamailio-v5-1-4-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.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - www.kamailioworld.com

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WARNING: topos [tps_storage.c:400]: tps_storage_record(): no local address
by Minh Phan 05 Jun '18

05 Jun '18

Hi List, I have this warning message in kamailio log when Kamailio handle an OPTIONS message with topos module enabled (I am using Kamailio 5.1.2): WARNING: topos [tps_storage.c:400]: tps_storage_record(): no local address - do record routing for all initial requests In my config file, OPTIONS message is handled this way: if ( is_method("OPTIONS") ) { t_newtran(); record_route(); t_reply("200","OK"); exit; } I have found this issue on Github but it is closed and I am not sure what was the solution: #1483 <https://github.com/kamailio/kamailio/issues/1483> My server receives 4 OPTIONS messages every 10 seconds that give as many warning in my log file. What could I do to get rid of them? Thanks, Minh

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Releasing Kamailio v5.1.4
by Daniel-Constantin Mierla 05 Jun '18

05 Jun '18

Hello, I am considering to release a new minor version out of branch 5.1, respectively the v5.1.4, sometime next week, likely on Tuesday, June 5, 2018. As usual, if there is any issue not yet reported on github tracker, do it as soon as possible to give it a chance to be fixed. Soon I am going to do the backports of commits with fixes in master that apply in branch 5.1, so keep a close eye and if someone is missing, let me know to push it in the stable branch. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference -- www.kamailioworld.com

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Transaction flags and branch_routes
by Alex Balashov 05 Jun '18

05 Jun '18

Hi, Given: flags PROXY_MEDIA_SET : 10, SIPPING : 11, DUMMY2 : 12 And: request_route { ... route(XYZ); } route[XYZ] { setflag(PROXY_MEDIA_SET); t_on_branch("BRANCH_ROUTE"); if(!t_relay()) sl_reply_error(); } branch_route[BRANCH_ROUTE] { if(isflagset(PROXY_MEDIA_SET)) { ... } ... } Is PROXY_MEDIA_SET meant to be visible inside the branch_route? Because it doesn't seem to be. Thanks! -- 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/

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ACK to 200 OK without to tag
by Kjeld Flarup 04 Jun '18

04 Jun '18

We have a SIP doorstation Grandstream GDS3710. When I connect it via kamailio, it does not setup call correct. Kamailio sends a 200 OK with this to field: t: <sip:0001@192.168.1.150:5069>;tag=e33a13d3-c18c-4df1-b68f-446e42957de6 Grandstream answers with: To: <sip:0001@192.168.1.150:5069> As the code follows this example: https://kamailio.org/docs/modules/4.3.x/modules/dispatcher.html#dispatcher.…, it seems obvious that this will fail, as there is not route[WITHINDLG] { if (has_totag()) { There simply is not to tag. Is this a bug in the grandstream? If so how could that go unnoticed, or is it me who misses something? I have attached the full sip text for the 200OK and the ACK -- -------------------- Med Liberalistiske Hilsner ---------------------- Civilingeniør, Kjeld Flarup - Mit sind er mere åbent end min tegnebog Sofienlundvej 6B, 7560 Hjerm, Tlf: 40 29 41 49 Den ikke akademiske hjemmeside for liberalismen - www.liberalismen.dk

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Jumping on new TCP connection during a SIP session?
by Abdoul Osséni 04 Jun '18

04 Jun '18

Hello group, The call flow is: useragent (TLS) -> Kamailio -> Sems Server. During the establishment of a call (before the 200OK or during a retransmission of the 200OK issued by Sems Server via Kamailio), if a user agent (TCP) loses its connection (the user agent is on a mobile network for example or the network is not reliable.) and sends a "register" to Kamailio (same source IP address but a different source TCP port), is it possible for Kamailio to take into account this new TCP port with the function t_relay () ? Abdoul

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Module sanity - memory leaks
by Konstantin Polyakov 04 Jun '18

04 Jun '18

Hello, We had memory leak on a production server and to find it we removed optional modules one by one. As result we found that problem is in the sanity module. Unfortunately I don't have time to fix it since I am going to vacation. We used the following checks: ruri_sip_version (1), required_headers (4), cseq_method (32), cseq_value (64), cotent_length (128), expires_value (256), proxy_require (512), parse uri's (1024). So be aware that this module is not ready to use in production. Best regards, Konstantin

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configuring websockts in kamilio
by vinod mn 04 Jun '18

04 Jun '18

hi configured websockets on kamilio 5.1.3 I am trying to register the client based on webrtc client ,(sip.js) the registration is fine but i am not getting the correct address if i run "*kamctl ul show"* * In the address filed I want the sip:<id>@<ip>:<port>ex: sip:12345@127.0.0.1:12345 <http://sip:12345@127.0.0.1:12345> * output shows: { "jsonrpc": "2.0", "result": { "Domains": [{ "Domain": { "Domain": "location", "Size": 1024, "AoRs": [{ "Info": { "AoR": "1xxxxxxxxx", "HashID": 1xxxxx, "Contacts": [{ "Contact": { *"Address": "sip:eg84vjhr@p6ggglrk0gsh.invalid;transport=ws"*, "Expires": 397, "Q": 1, "Call-ID": "ma4feu2i43bnu6ophe4k78", "CSeq": 86, "User-Agent": "SIP.js/0.5.0-devel BB", "Received": "sip:X.X.X.X:56087;transport=ws", "Path": "[not set]", "State": "CS_NEW", "Flags": 0, "CFlags": 192, "Socket": "tcp:X.X.X.X:8080", "Methods": 783, "Ruid": "uloc-5b116457-1dc1-03", "Instance": "<urn:uuid:0e879ebf-0a31-45a8-9fd9-6b937d3328d4>", "Reg-Id": 1, "Server-Id": 0, "Tcpconn-Id": 259, "Keepalive": 1, "Last-Keepalive": 1527876355, "Last-Modified": 1527876355 } }] } } ], "Stats": { "Records": 1, "Max-Slots": 1 } } }] }, "id": 18958 } In the address filed I want the sip:<id>@<ip>:<port> ex: sip:12345@127.0.0.1:12345 Please give me any suggestions on this, which webrtc client can i use to register to kamailio and able to receive on the client and to make outbound calls from the client. -- Thanks and regards Vinod.M.N

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Kamailio Behind TCP Load Balancer
by Colin Morelli 04 Jun '18

04 Jun '18

Hey all, I'm looking to put Kamailio behind a TCP load balancer that is SIP-unaware. My application is deployed in AWS and I'm tying to place Kamailio behind an ELB. For the most part, everything is fine. For my specific implementation I'm disabling UDP as a signaling transport and using only TLS. This enables me to not have to worry about a SIP-aware LB at the edge because replies to an incoming request will be sent over the existing established TCP socket (avoiding any crazy routing requirements). However - this poses an issue with source addresses. Does Kamailio support anything like the proxy protocol ( http://www.haproxy.org/download/1.5/doc/proxy-protocol.txt) for getting TCP stream information from a load balancer? Do I need to go back to exposing it directly to the world so that I can get source addresses? Thanks, Colin

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Re: [SR-Users] configuring websockts in kamilio
by Konstantin Polyakov 02 Jun '18

02 Jun '18

Hello Vinod, We used Doubango sipML5 web phone https://www.doubango.org/sipml5/call.htm?svn=252 It was quite good for debugging and testing webrtc calls. Most of work in kamailio config I made by using that client. Best regards. Konstantin > >Message: 8 >Date: Fri, 1 Jun 2018 23:50:50 +0530 >From: vinod mn < vinodmn.dbit(a)gmail.com > >To: sr-users(a)lists.kamailio.org >Subject: [SR-Users] configuring websockts in kamilio >Message-ID: >< CAHeWXy7h4Tcp4Sg9_eM3teCHBdJ5bwHa9TYSJy+7P4e+bm1hqQ(a)mail.gmail.com > >Content-Type: text/plain; charset="utf-8" > >hi configured websockets on kamilio 5.1.3 >I am trying to register the client based on webrtc client ,(sip.js) > the registration is fine but i am not getting the correct address if i >run "*kamctl ul show"* > > >* In the address filed I want the sip:<id>@<ip>:<port>ex: >sip:12345@127.0.0.1:12345 < http://sip:12345@127.0.0.1:12345 > * >output shows: >{ > "jsonrpc": "2.0", > "result": { > "Domains": [{ > "Domain": { > "Domain": "location", > "Size": 1024, > "AoRs": [{ > "Info": { > "AoR": "1xxxxxxxxx", > "HashID": 1xxxxx, > "Contacts": [{ > "Contact": { > *"Address": >"sip:eg84vjhr@p6ggglrk0gsh.invalid;transport=ws"*, > "Expires": 397, > "Q": 1, > "Call-ID": "ma4feu2i43bnu6ophe4k78", > "CSeq": 86, > "User-Agent": "SIP.js/0.5.0-devel BB", > "Received": "sip:X.X.X.X:56087;transport=ws", > "Path": "[not set]", > "State": "CS_NEW", > "Flags": 0, > "CFlags": 192, > "Socket": "tcp:X.X.X.X:8080", > "Methods": 783, > "Ruid": "uloc-5b116457-1dc1-03", > "Instance": >"<urn:uuid:0e879ebf-0a31-45a8-9fd9-6b937d3328d4>", > "Reg-Id": 1, > "Server-Id": 0, > "Tcpconn-Id": 259, > "Keepalive": 1, > "Last-Keepalive": 1527876355, > "Last-Modified": 1527876355 > } > }] > } > } > ], > "Stats": { > "Records": 1, > "Max-Slots": 1 > } > } > }] > }, > "id": 18958 >} > >In the address filed I want the sip:<id>@<ip>:<port> >ex: sip:12345@127.0.0.1:12345 > >Please give me any suggestions on this, which webrtc client can i use to >register to kamailio and able to receive on the client and to make outbound >calls from the client. > >-- >Thanks and regards >Vinod.M.N >

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CDRs
by David Villasmil 02 Jun '18

02 Jun '18

Hello guys, I setup the CDRs according to the instructions on https://kb.asipto.com/siremis:install40x:accounting I get CDRs nicely, but the cdrs, even though they have a src_ip, there's no dest_ip, I mean I need the IP where the call was forwarded to... What do i need to do to get that in the cdr? Thanks! David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

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Upgrade issue from 4.4.7 to 5.1.3
by Kevin Greene 01 Jun '18

01 Jun '18

Hello List users: I am trying to upgrade to the current version of Kamailio from the older 4.4 version to the current version. The 4.4 is functional but there are some features of 5.1 that we wanted to test with. It seems that my version of Linux (RHEL 6.9 or Oracle 6.9) has an issue with a library file. Any help would be useful. Thank you. Kevin Notes: I used the "yum update kamailio" function. (edited for clarity - it showed a long list of modules to be updated) "--> Finished Dependency Resolution Error: Package: kamailio-http-utils-4.4.7-1.1.x86_64 (@home_kamailio_v4.4.x-rpms) Requires: kamailio = 4.4.7 Removing: kamailio-4.4.7-1.1.x86_64 (@home_kamailio_v4.4.x-rpms) kamailio = 4.4.7-1.1 Updated By: kamailio-5.1.3-1.el6.x86_64 (home_kamailio_v5.1.x-rpms) kamailio = 5.1.3-1.el6 Error: Package: kamailio-http-utils-4.4.7-1.1.x86_64 (@home_kamailio_v4.4.x-rpms) Requires: libkmi.so.1()(64bit) Removing: kamailio-4.4.7-1.1.x86_64 (@home_kamailio_v4.4.x-rpms) libkmi.so.1()(64bit) Updated By: kamailio-5.1.3-1.el6.x86_64 (home_kamailio_v5.1.x-rpms) Not found Error: Package: kamailio-xml-4.4.7-1.1.x86_64 (@home_kamailio_v4.4.x-rpms) Requires: kamailio = 4.4.7 Removing: kamailio-4.4.7-1.1.x86_64 (@home_kamailio_v4.4.x-rpms) kamailio = 4.4.7-1.1 Updated By: kamailio-5.1.3-1.el6.x86_64 (home_kamailio_v5.1.x-rpms) kamailio = 5.1.3-1.el6 You could try using --skip-broken to work around the problem You could try running: rpm -Va --nofiles --nodigest" However the "missing" library is there in /usr/lib64. [root@sipserver tmp]# find / -name libkmi.so.1 /usr/lib64/kamailio/libkmi.so.1 /usr/lib64/libkmi.so.1 /lib64/libkmi.so.1 To provide additional help, See: [root@sipserver tmp]# kamailio -version version: kamailio 4.4.7 (x86_64/linux) 35d8ab flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, 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: 35d8ab compiled on 15:38:14 Feb 26 2018 with gcc 4.4.7

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configuring the websocket in kamailio 5.1
by vinod mn 01 Jun '18

01 Jun '18

Hi, I configured the websocket in kamailio and it is running properly, which is the best webrtc client (with proper code as I dont't want to spend more time in writing the codes based on the example snippets ) can I use for registering, and making/ receiving the calls on the webrtc-client . requirement: webrtc-client(??)->register-> kamailio(already configured to use websocket) webrtc-client -> makecall to kamailio webrtc-client receives call from another client(sip not webrtc-client) -- Thanks and regards Vinod.M.N

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Thank You
by Wilkins, Steve 01 Jun '18

01 Jun '18

Hi All, I normally get on this board to ask questions but today I just wanted to take a minute and say - Thank you to all of you who have taken the time to help me. Some of you have spent a lot of time schooling me on Kamailio, its Interactions with a PBX and different clients. I have learned tons from this board and just want to say again, Thank You All! -Steve

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Multiple CDR rate groups using Siremis sql procedures
by Sergiu Pojoga 01 Jun '18

01 Jun '18

Hi, The SQL procedure takes a param, rgroup, which is default (by default). That works fine. In a real case scenario, having multiple rate groups, I wonder what's the best approach to rate CDRs based on the rate group a client belongs to? Do you store the rate_group in the ACC table and then adjust 'kamailio_cdrs' and 'kamailio_rating' procedures to take into account the rate_group of a CDR? If so, does anybody have a working example? Thanks.

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Anybody using 2 instances with topos communicating with each other?
by Daniel Tryba 01 Jun '18

01 Jun '18

Scenario: endpoint (bob) <-> kamailio proxy (5.1.3) <-> kamailio backend (5.1.2) Both kamailio instances are running the topos module (mysql backend). Call setup is fine, but forwarding BYEs fail. With a call from endpoint to kamailio, topos on the proxy scrambles Contact to kamailio backend. But in responses from backend contact remains untouched. In the ACK to the 200, topos on the proxy does nothing. If the caller hangsup, the BYE will loop internally till max hops is exceeded (483 which will be send to caller eventually). When the backend initiates the call to the endpoint, topos appears to work, but the BYE from the endpoint on hangup get a 404 Not here from the proxy (route[WITHINDLG]). Disabling topos on the proxy will result in a functioning proxy (config attached). Is there something wrong with my config? Or is this an unforseen scenario that just doesn't work with topos? * Endpoint to backend scenario endpoint:5060 -> proxy:5060 INVITE sip:callee@sip.pocos.nl;transport=tcp SIP/2.0 Via: SIP/2.0/UDP endpoint:5060;rport;branch=... Max-Forwards: 70 Route: <sip:proxy:5060;lr> Contact: <sip:bob@endpoint:5060;ob> proxy:5060 -> backend:5060 INVITE sip:callee@sip.itco.nl;transport=tcp SIP/2.0 Via: SIP/2.0/UDP proxy;branch=... CSeq: 29848 INVITE Route: <sip:proxy:5060;lr> Contact: <sip:btpsh-5af5972d-bf5e-2@proxy> backend:5060 -> proxy:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP proxy;branch=... Contact: <sip:atpsh-5af19bf0-9f94-7@backend> proxy:5060 -> endpoint:5060 SIP/2.0 200 OK Contact: <sip:atpsh-5af19bf0-9f94-7@backend> Via: SIP/2.0/UDP endpoint:5060;received=endpoint;rport=5060;branch=... Record-Route: <sip:proxy;lr> endpoint:5060 -> proxy:5060 ACK sip:atpsh-5af19bf0-9f94-7@backend SIP/2.0 Via: SIP/2.0/UDP endpoint:5060;rport;branch=... Route: <sip:proxy;lr> proxy:5060 -> backend:5060 ACK sip:atpsh-5af19bf0-9f94-7@backend SIP/2.0 Via: SIP/2.0/UDP proxy;branch=... Via: SIP/2.0/UDP endpoint:5060;received=endpoint;rport=5060;branch=... endpoint:5060 -> proxy:5060 BYE sip:atpsh-5af19bf0-9f94-7@backend SIP/2.0 Via: SIP/2.0/UDP endpoint:5060;rport;branch=... Max-Forwards: 70 Route: <sip:proxy;lr> proxy:5060 -> proxy:5060 BYE sip:atpsh-5af19bf0-9f94-7@backend SIP/2.0 Via: SIP/2.0/UDP proxy;branch=... Via: SIP/2.0/UDP endpoint:5060;received=endpoint;rport=5060;branch=... Max-Forwards: 69 Route: <sip:proxy;lr> proxy:5060 -> proxy:5060 BYE sip:atpsh-5af19bf0-9f94-7@backend SIP/2.0 Via: SIP/2.0/UDP proxy;branch=... Via: SIP/2.0/UDP proxy;rport=5060;branch=... Via: SIP/2.0/UDP endpoint:5060;received=endpoint;rport=5060;branch=... Max-Forwards: 68 Route: <sip:proxy;lr> [not going to mention the 50+ BYEs adding Vias and decrementing Max-Forwards. * Backend to endpoint scenario backend:5060 -> proxy:5060 INVITE sip:bob@endpoint:5060;ob SIP/2.0 Via: SIP/2.0/UDP backend;branch=... Route: <sip:loadbalancer@proxy;lr;received=sip:endpoint:5060> Contact: <sip:btpsh-5af07a27-72b-502c1@backend> proxy:5060 -> endpoint:5060 INVITE sip:bob@endpoint:5060;ob SIP/2.0 Via: SIP/2.0/UDP proxy;branch=... Contact: <sip:btpsh-5af5972d-bf5d-5@proxy> endpoint:5060 -> proxy:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP proxy;received=proxy;branch=... Contact: <sip:bob@endpoint:5060;ob> proxy:5060 -> backend:5060 SIP/2.0 200 OK Contact: <sip:bob@endpoint:5060;ob;alias=endpoint~5060~1> Via: SIP/2.0/UDP backend;rport=5060;branch=... Record-Route: <sip:proxy;lr> endpoint:5060 -> proxy:5060 BYE sip:btpsh-5af5972d-bf5d-5@proxy SIP/2.0 Via: SIP/2.0/UDP endpoint:5060;rport;branch=... proxy:5060 -> endpoint:5060 SIP/2.0 404 Not here Via: SIP/2.0/UDP endpoint:5060;rport=5060;branch=...

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Dispatcher module ERROR
by Aleksandar Sosic 01 Jun '18

01 Jun '18

Hi everyone, We have an error using the dispatcher module. In the logs: ```0(107) ERROR: dispatcher [dispatcher.c:815]: ds_warn_fixup(): failover functions used, but required AVP parameters are NULL -- feature disabled``` We are running 5.2.0~dev5+0~20180523010152.1118+xenial The configuration file parts regarding dispatcher are as follows: ``` loadmodule "dispatcher.so" ... # ----- dispatcher params ----- modparam("dispatcher", "list_file", "/tmp/dispatcher.list") # Dispatcher: Enable Failover-Support modparam("dispatcher", "flags", 2) # Dispatcher: Overwrite Destination address, if required. modparam("dispatcher", "force_dst", 1) # AVP's required for Fail-Over-Support: modparam("dispatcher", "dst_avp", "$avp(DISPATCHER_DST_AVP)") modparam("dispatcher", "grp_avp", "$avp(DISPATCHER_GRP_AVP)") modparam("dispatcher", "cnt_avp", "$avp(DISPATCHER_CNT_AVP)") # Try to recover disabled destinations every 15 seconds. modparam("dispatcher", "ds_ping_interval", 15) # Actively query the gateways: modparam("dispatcher", "ds_probing_mode",0)# ----- dispatcher params ----- modparam("dispatcher", "list_file", "/tmp/dispatcher.list") # Dispatcher: Enable Failover-Support modparam("dispatcher", "flags", 2) # Dispatcher: Overwrite Destination address, if required. modparam("dispatcher", "force_dst", 1) # AVP's required for Fail-Over-Support: modparam("dispatcher", "dst_avp", "$avp(DISPATCHER_DST_AVP)") modparam("dispatcher", "grp_avp", "$avp(DISPATCHER_GRP_AVP)") modparam("dispatcher", "cnt_avp", "$avp(DISPATCHER_CNT_AVP)") # Try to recover disabled destinations every 15 seconds. modparam("dispatcher", "ds_ping_interval", 15) # Actively query the gateways: modparam("dispatcher", "ds_probing_mode",0)``` What are we doing wrong? What are we missing? Thanks, -- Aleksandar Sosic linkedin.com/in/alexsosic/ mail: alex.sosic(a)evosip.cloud

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Authenticate against outbound carrier
by Krzysztof Grobelak 01 Jun '18

01 Jun '18

Hello List, I have a working setup with a kamailio + freeswitch where i can register extension and make succesfull calls between them. I added now a "external" path for outbound call to which i can successfully route using the dialplan + dispatcher combo. My problem is that my outbound path requires authentication and i'm struggling to find a way to successfully authenticate with the remote end. I configured the UAC module to register against the remote end with a set of credentials but in the SIP trace i can see that the kamailio simply uses the local user credentials instead of the external ones. I believe that this is because the 401 Unauthorized from the carrier gets proxied to the extension which responds with its own credentials. Can anybody suggest how can i stop forwarding them in this case to extension and instead use the UAC to reply with different set of credentials? Regards, Krzysztof The config: request_route { ... if($rU=~"[0-9]{5,}"){ xlog("L_INFO","Looking up route for $rU\n"); dp_match("1", "$rU"); $var(dsid) = $(var(attrs){s.int}); if(!ds_select_dst("$var(dsid)", "4")) { send_reply("404", "No destination"); exit; } if(!ds_select_domain("$var(dsid)", "4")) { send_reply("404", "No destination"); exit; } xlog("L_INFO", "Looking up $fn in database\n"); $avp(s:auth_realm_avp) = "realm.com"; $avp(s:auth_username_avp) = "username"; $avp(s:auth_password_avp) = "password"; if(uac_reg_request_to("$fn", 1)) { xlog("L_INFO", "Found remote user [$rU] on [$rd] via [$du]"); t_on_failure("REMOTE_AUTH"); xlog("L_INFO", "going to <$ru> via <$du>\n"); route(RELAY); } exit; } failure_route[REMOTE_AUTH] { if ($T_reply_code == 401 or $T_reply_code == 407) { xlog("L_INFO", "Remote asked for authentication"); uac_auth(); } } -- PGP: 9E5D E635 5A03 485C 2E00 4530 EB83 920A 9E21 FBA8

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Acc module and PostgreSQL
by Javohir Maxmarajabov 01 Jun '18

01 Jun '18

Is it possible to enable accounting module with PostgreSQL? All I found is documentation and examples that showing accounting module is used with MySQL module. -- Javokhir M.M. ________________________________________________ Disclaimer: The information in this e-mail is confidential. If you are not addressed recipient then please return and delete this e-mail from your system. Unauthorised use of or disclose the contents of this e-mail may be unlawful.

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Strange behavior with $shtinc
by Brooks Bridges 01 Jun '18

01 Jun '18

Let me preface this with the statement that as best I can tell from docs and code, this function is supposed to be 100% atomic and this shouldn't be able to happen. I also have not found anything in the module's github commit logs that would indicate this behavior has changed from previous versions. That being said, I appear to have run into an oddity where intermittently I'm getting 2 slots being put in the same entry, which results in trying to parse the output for graphing and reporting not working quite right. The mechanism I'm using is a very simple counter mechanism to record statistics about calls. As an example, I am using the following mechanism: $var(z) = $shtinc(callstats=>$avp(realm),cps_exceeded); or $var(z) = $shtinc(callstats=>$avp(realm),sessions_exceeded); This typically results in something like this: { entry: 4335 size: 1 slot: { { name: realm1,total_requests value: 2365034 type: int } } } { entry: 4532 size: 1 slot: { { name: realm2,cps_exceeded value: 30 type: int } } } And so on... However, one or more of them sometimes end up like this: { entry: 4646 size: 2 slot: { { name: realm1,total_requests value: 15958026 type: int } { name: realm2,cps_exceeded value: 6432103 type: int } } } Thanks in advance for your help and/or suggestions! Brooks Bridges | Engineering Manager O1 Communications 4359 Town Center Boulevard, Suite 217 El Dorado Hills, California 95762 office: 916.235.2097 | main: 888.444.1111, Option 2 email: bbridges(a)o1.com<mailto:bbridges@o1.com> | web: www.o1.com<http://www.o1.com/>

3 5

Authentication against outbound carrier trunk
by Krzysztof Grobelak 31 May '18

31 May '18

Hello List, I have a working setup with a kamailio + freeswitch where i can register extension and make succesfull calls between them. I added now a "external" path for outbound call to which i can successfully route using the dialplan + dispatcher combo. My problem is that my outbound path requires authentication and i'm struggling to find a way to successfully authenticate with the remote end. I configured the UAC module to register against the remote end with a set of credentials but in the SIP trace i can see that the kamailio simply uses the local user credentials instead of the external ones. I believe that this is because the 401 Unauthorized from the carrier gets proxied to the extension which responds with its own credentials. Can anybody suggest how can i stop forwarding them in this case to extension and instead use the UAC to reply with different set of credentials? Regards, Krzysztof The config: request_route { ... if($rU=~"[0-9]{5,}"){ xlog("L_INFO","Looking up route for $rU\n"); dp_match("1", "$rU"); $var(dsid) = $(var(attrs){s.int}); if(!ds_select_dst("$var(dsid)", "4")) { send_reply("404", "No destination"); exit; } if(!ds_select_domain("$var(dsid)", "4")) { send_reply("404", "No destination"); exit; } xlog("L_INFO", "Looking up $fn in database\n"); $avp(s:auth_realm_avp) = "realm.com"; $avp(s:auth_username_avp) = "username"; $avp(s:auth_password_avp) = "password"; if(uac_reg_request_to("$fn", 1)) { xlog("L_INFO", "Found remote user [$rU] on [$rd] via [$du]"); t_on_failure("REMOTE_AUTH"); xlog("L_INFO", "going to <$ru> via <$du>\n"); route(RELAY); } exit; } failure_route[REMOTE_AUTH] { if ($T_reply_code == 401 or $T_reply_code == 407) { xlog("L_INFO", "Remote asked for authentication"); uac_auth(); } } -- PGP: 9E5D E635 5A03 485C 2E00 4530 EB83 920A 9E21 FBA8

1 0

error in tls configuration in kamailio 5.1.3
by Narasimhaiah, Vinod Maralenahalli 31 May '18

31 May '18

Hi, I upgraded kamailio 4.4.2 to kamailio5.1.3 and i am getting this error *"Unable to load certificate file '/etc/kamailio/kamailio-selfsigned.pem",* kamilio 4.4.2 does not need the ssl file, but kamailio 5.1.3 need ssl file, what changes required in kamailio 5.1.3 to run with tls. Please help me in resolving this issue. Thank you vinod.M.N -- To download the app, click: iOS <http://goo.gl/ZJx3y1> | Android <http://goo.gl/G1ehjZ> -- ***This transmission contains information that is confidential and proprietary of Kirusa Inc.,its affiliates and/or its partners and customers. This information is intended to be shared only with the recipient of this e-mail who is bound by confidentiality. If you received this e-mail in error, please delete the e-mail and notify the sender by return e-mail.***

4 4

Upgrade issue with old Kamailio 3.0.3 in table rls_presentity
by Kristijan Vrban 31 May '18

31 May '18

Hi, i try a upgrade from old 3.0.3 Kamailio mysql db. Got stuck with update the structure : ALTER TABLE rls_presentity MODIFY callid VARCHAR(255) NOT NULL;ALTER TABLE rls_presentity MODIFY contact VARCHAR(128) NOT NULL; found on: https://www.kamailio.org/wiki/install/upgrade/3.1.x-to-3.2.0 But this two table does not exist? See also table definition at: https://kamailio.org/docs/db-tables/kamailio-db-3.1.x.html#GEN-DB-RLS-PRESE… I reckon it is safe to skip this to tables? Kristijan

2 4

Register on kamailio.org wiki page not working
by Kristijan Vrban 31 May '18

31 May '18

Hi, https://www.kamailio.org/wiki/start?do=register Is not working. You get a empty page here. Kristijan

2 2

Sharing static variables in external library across child proceses
by User 721 31 May '18

31 May '18

Hi, I am adding a kamailio module that references an external C library that maintains a hashtable in global scope. I am wondering what practice I should follow so that: - This hashtable (which I don't control directly) is shared by all child processes - The hashtable is changed by one child process at a time Thanks in advance

2 1

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