- sr-users - kamailio.org

NATHELPER issue
by Ali Taher 20 Jul '18

20 Jul '18

Hello, I'm using Kamailio 4.2 as proxy with nathelper enabled. Yet , the ACK packet sent from the proxy to the origination's private IP. The ACK is sent as reply on the following 200 OK sent from the origination : 8m2EJN41BN/6WSIP/2.0 200 OK From: <sip:+4444331234567@X.X.X.X;user=phone>;tag=XQBQNjvjgp4Ze To: <sip:+905362695933@172.16.45.65;user=phone>;tag=12033368836000 Via: SIP/2.0/UDP X.X.X.X:5065;branch=z9hG4bK2959.233ecbc5eff949f946d8763ce25e5e6d.0;received= X.X.X.X,SIP/2.0/UDP X.X.X.X;received=X.X.X.X;rport=5060;branch=z9hG4bKN93cXvv26vDDN Record-Route: <sip:X.X.X.X:5065;lr;ftag=XQBQNjvjgp4Ze> Call-ID: CbeX8453909200habfGhEfElPce(a)BC00.XXXXXXXXXXXXXX CSeq: 125698370 INVITE Accept: application/sdp Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,UPDATE P-Charging-Vector: icid-value=B0912C3D70-0720-09394507;icid-generated-at=BC00.XXXXXXXXXXXXXX.XX ;orig-ioi=MXXXXXXXXXXXXXX Content-Type: application/sdp Contact: <sip:172.16.45.65:5060;transport=UDP> Content-Length: 268 v=0 o=- 5838243 5838244 IN IP4 BC00.XXXXXXXXXXXXXX s=- c=IN IP4 172.16.45.144 t=0 0 a=sendrecv m=audio 47588 RTP/AVP 18 96 c=IN IP4 172.16.45.144 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=maxptime:20 Following is the header of the sent ACK packet: Request-Line: ACK sip:172.16.45.65:5060;transport=UDP SIP/2.0 Record-Route: <sip:X.X.X.X:5065;lr;ftag=XQBQNjvjgp4Ze> Via: SIP/2.0/UDP X.X.X.X:5065;branch=z9hG4bK2959.871535fd341bbe3099d0bf60d6460e18.0 Via: SIP/2.0/UDP X.X.X.X;received=X.X.X.X;rport=5060;branch=z9hG4bKQUpy0jy90etjc Max-Forwards: 69 From: <sip:+4444331234567@X.X.X.X;user=phone>;tag=XQBQNjvjgp4Ze To: <sip:+905362695933@172.16.45.65;user=phone>;tag=12033368836000 Call-ID: CbeX8453909200habfGhEfElPce(a)BC00.XXXXXXXXXXXXXX CSeq: 125698370 ACK Content-Length: 0 Where X.X.X.X is Kamailio server public IP. Following is part of my config file : route { route(NATDETECT); record_route(); if(!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); exit; } # Maybe some sanity_check() here. if(has_totag()) { if(loose_route()) { route(DLGURI); if(!t_relay()) sl_reply_error(); exit; } else { if(is_method("ACK")) { route(DLGURI); if(t_check_trans()) { t_relay(); } } else sl_send_reply("403", "Forbidden"); } exit; } ..... } route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { add_contact_alias(); } setflag(FLT_NATS); } #!endif return; } route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } Can you please check why the ACK is still sent on private IP ? Thanks Ali Taher

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The problem of get_aor_hash function
by tyd 20 Jul '18

20 Jul '18

Dear all, I'm trying Kamailio 5.1.4 and IMS module. When registering to Kamailio IMS using the same ip and port for 30 users (different contact user part), the P-CSCF get the same hash number and aor value. The function get_aor_hash(_d, &_ci->via_host, _ci->via_port, _ci->via_prot) in ims_usrloc_pcscf pcontact.c seems the problem. Because hashing with host, port, prot will get the same hash value. Anyone can help this ?

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dialog module : questions about dialog states
by Abdoul Osséni 19 Jul '18

19 Jul '18

Hello list, When the dialog state for a call is < 4, is it possible to use dlg_bye() function to end the call? Best regards Abdoul OSSENI Ingénieur DevOps chez Néo-Soft Co-Fondateur de ON SERVICES Tél : +33 601 135 167

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Locking htable entries
by Alex Balashov 19 Jul '18

19 Jul '18

Hi, Sorry if this question has been answered before, but I'm rather puzzled as to the intent behind sht_lock() / sht_unlock(). Presumably, "thread"-safe / shm mutex locking is already implemented in htable query and mutation operations, otherwise htable would be unusable in a route script context. So what is the purpose of this higher-level lock? Is it to create a way to block config script execution in multiple processes as to serialise operations? Why would one want to do this under any ordinary circumstances? Perhaps it is a failure of my imagination... 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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executing python routine on demand
by Walter Martín Villalba 18 Jul '18

18 Jul '18

Hello, TL;DR How can I execute a routine in my Kamailio python module (kamailio_python.py) on demand? For example, using kamcmd from the command line. I know you can execute *app_python.reload *but I don't need to reload the whole python module, and furthermore, docs for this call say it's unstable and can even crash your kamailio instance. So my goal is to execute one single python routine on demand. https://www.kamailio.org/docs/modules/5.1.x/modules/app_python.html#app_pyt… Full explanation: I'm currently implementing OIDC token authentication for Kamailio. I have a python routine that extracts the token from a custom SIP header, and validates the token as per the OIDC spec (I'm using pyjwt plus other custom checks for this). In order to do offline validation of the token, you need the Identity Provider (IdP) Server's certificate, which I get in advance and cache in a local file on my Kamailio server. I also have an external python script which I can run to get the latest certificate from the IdP server and update it locally. The part I'm missing is how to tell Kamailio to reload the value of the certificate from the local cache, WITHOUT stopping kamailio or interrupting any sessions. This is needed in case the certificate is updated (e.g. due to expiration or compromise) on the IdP server; if you don't update it so that Kamailio can use the latest one, all token verification will fail (tokens signed by IdP server using latest private key corresponding to latest certificate, but Kamailio trying to validate the signature using the public key extracted from previous certificate -- this results in a signature error, as Kamailio needs to be using the public key corresponding to the private one that was used to sign the token). By being able to execute one single python routine on demand (e.g. using kamcmd), I can tell Kamailio to reload the certificate value from the local cache, which I just updated by running my other (external and unrelated to Kamailio software) python script. Hope this makes sense, and hopefully I can get some tips on how to achieve this. Thanks, Martín.

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in-dialog requests
by Ali Taher 18 Jul '18

18 Jul '18

Hello, I'm using Kamailio as a proxy behind freeswitch to alter some packets (183 packets precisely). Here is my scenario: (---Customer---) --> ((port 5065)---Kamailio---) --> ((port5060)---Freeswitch---) --> (---Supplier---) Here is my request route : route{ if (is_method("INVITE")) { xlog("L_INFO","INSIDE request route $si \n"); insert_hf("X-AUTH-IP: $si\r\n"); } route(FWD); } reply_route { #xlog("L_INFO","INSDE reply \n"); if (status == "183") { change_reply_status(180, "Ringing"); exit; } } route[FWD] { #xlog("L_INFO","INSIDE FWD \n"); forward(X.X.X.X,5060); } Where X.X.X.X is the Freeswitch IP This script is working normally as long as the customer is sending the requests(INVITE,BYE,ACK,CANCEL..) . But when the supplier or freeswitch initiated the request , above script causes loop where the packet is bounced between kamailio and freeswitch. As I know , loose_route is used to handle such requests , but I'm not sure how to forward these packets to the customer after receiving them from freeswitch. I'm not sure also if what's I'm doing is statefull or stateless. Any help is appreciated. Thanks, Ali Taher

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Kamailio dialog presence (BLF)
by Michal Škuta 18 Jul '18

18 Jul '18

Hello, I am trying make presence module work with Kamailio. Right now I am testing these modules: presence, presence_xml, presence_dialoginfo, pua and pua_dialoginfo. I had some success, the BLF is working only when the monitored extension is calling. When the monitored extension is calling, then the BLF changes its color (red). After end of the call the BLF changes color to green. My problem is when a User comes on-line then it sends SUBSCRIBE with *Event: dialog* and don't get notified of its subscribers state and BLF is not changed. I found similar problem in this mail list : https://lists.kamailio.org/pipermail/sr-users/2017-March/096314.html But is not working for me. Second question is about BLF and offline state. I can not found anything in RFC, there is no offline state, no way to turn off the BLF lamp from green to OFF. I found only this pdf from grandstream: http://www.grandstream.com/sites/default/files/Resources/GXP21x0_Eventlist_… mentioning offline state on page 8. Does anyone have experience with it?

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Using the SIP related functions in a lib
by Bobby Mah 18 Jul '18

18 Jul '18

Hello all, Is it worth my time to try and create a SIP lib out of the SIP functionality available in Kamailio? Thanks for all your input, Bobby

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SIP re-invite Packets
by Ali Taher 18 Jul '18

18 Jul '18

Hello, I'm using below routing logic to change the 183 reply to 180 and send it back to the origination. Noting that Kamailio is listening on 5065. route{ if (is_method("INVITE")) { xlog("L_INFO","INSIDE request route $si \n"); insert_hf("X-AUTH-IP: $si\r\n"); } route(FWD); } reply_route { #xlog("L_INFO","INSDE reply \n"); if (status == "183") { change_reply_status(180, "Ringing"); exit; } } route[FWD] { #xlog("L_INFO","INSIDE FWD \n"); forward(x.x.x.x,5060); } Below is the scenario , where the customer is sending packets to Kamailio , and Kamailio is forwarding them the supplier (with IP x.x.x.x) , then the supplier reply back to kamailio and kamailio relay the reply to the customer ----Customer---- --> ----Kamailio ---> ---Supplier--- I have a case where the supplier sends in some cases an in-dialog invite (Re-Invite) , which is considered I think as SIP-Request and thus as per above routing logic , kamailio is sending it back to the supplier , yet I need to send it to the customer. How can I solve this issue? Thanks Ali Taher

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Error on Kamailio 4.3.7 and dialog
by Laura 18 Jul '18

18 Jul '18

Hi guys, on a test server with kamailio 4.3.7 statefull configuration with dialog I see a lot of messages like this.. Jul 16 15:20:26 vm-nextaitz-01 /usr/sbin/kamailio[31288]: ERROR: dialog [dlg_req_within.c:340]: send_bye(): terminating non-confirmed dialogs not supported Jul 16 15:20:26 vm-nextaitz-01 /usr/sbin/kamailio[31288]: ERROR: dialog [dlg_req_within.c:340]: send_bye(): terminating non-confirmed dialogs not supported Jul 16 15:20:26 vm-nextaitz-01 media-dispatcher[31555]: error: failed to end dialog: 500 Operation failed I'm not find the issue and the way for fix.. Any idea ? Regards Laura

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ACK Cseq incrases after 404 on uac_auth()
by Yuriy Gorlichenko 17 Jul '18

17 Jul '18

Hi. I found some additional scenario when kamailio works incorrect with ACK Cseq after succesfull auth with uac_auth but unsuccesfull call: 1.Caller via kamailio as proxy (Cseq 1) 2.Kamailio resends call to provider with Auth method (Cseq 1) 3. Provider answers 401/407 (Cseq 1) 4. Kamailio inrases Cseq and sends invite with credentians (dialog track_cseq_updates = 1, uac_auth()) (Cseq 2) 5. Provider says - 404. not found (Cseq 2) 6. Kamailio resends INVITE to another server (lets say VM). It contans not tracked yet Cseq (Cseq 1) 7. VM server says 200 (Cseq 1) 8. Kamailio resends 200 to the caller (Сseq 1) 9. Caller says ACK (Cseq 1) 10. Kamailio incrases CSeq of ACK and resends t VM server (Сseq 2) So in this case kamialio tries to track Sceq because thining that this ACK belongs to the dialog that was made to the auth server So as I see here is a misunderstanding on the dialog module Any suggestions how to avoid it or exclude resent INVITE from the dialog module tracking?

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RTPEngine serial forking with early media
by Alex Balashov 16 Jul '18

16 Jul '18

Hi, I've got a scenario where call branch #1 goes to an in-call announcement server; the message is played out via unidirectional early media (183+SDP), then that branch is disposed of with a 503. A subsequent call then goes out to the real SIP provider that will ultimately answer the call. There's some trouble handling this with RTPEngine: Two subsequent calls to rtpengine_offer() in a normal request route context will result in two SDP message lumps concatenated in the second INVITE (to the provider). This can be fixed by putting the rtpengine_offer() invocations into branch routes. However, the next problem is that then the identical offer goes to the second callee with no regard for RTPEngine's internal state, so RTPEngine ends up transmitting from a different port to the one that is declared in the SDP offer, requiring NAT latching from the provider to achieve proper symmetry in the return stream. This can be fixed with: rtpengine_offer("replace-origin replace-session-connection ICE=remove via-branch=1 internal external"); And in the reply handling: rtpengine_answer("replace-origin replace-session-connection ICE=remove via-branch=2"); An alternative to using via-branch[1] is to simply delete the call before initiating branch #2: rtpengine_delete("delete-delay=0"); rtpengine_offer("replace-origin replace-session-connection ICE=remove internal external"); However, both of these lead to the problem that caller receives two distinct SDP answers on the respective branches. I am not sure offhand whether this is a protocol semantics violation. On the one hand, RFC 3261 § 13.2.1 ("Creating the Initial INVITE") says: If the initial offer is in an INVITE, the answer MUST be in a reliable non-failure message from UAS back to UAC which is correlated to that INVITE. For this specification, that is only the final 2xx response to that INVITE. That same exact answer MAY also be placed in any provisional responses sent prior to the answer. The UAC MUST treat the first session description it receives as the answer, and MUST ignore any session descriptions in subsequent responses to the initial INVITE. So, I always understood that the first answer is the final answer, absent a session-updating request cycle. On the other hand, RFC 3960 ("Early Media and Ringing Tone Generation in the Session Initiation Protocol (SIP)") Section 4 says: The application server model consists of having the UAS behave as an application server to establish early media sessions with the UAC. The UAC indicates support for the early-session disposition type (defined in [2]) using the early-session option tag. This way, UASs know that they can keep offer/answer exchanges for early media (early-session disposition type) separate from regular media (session disposition type). I take this, along with RFC 3959 Section 3, to imply an amendment to 3261 § 13.2.1, but I'm not sure. Regardless, I'm not convinced all UAs will handle this situation. So, what I would most prefer is a means of recycling the same answer port on the "back" side of RTPEngine while pivoting it on the front across branches/multiple offers. Is that possible? If so, how can it be achieved? Many thanks, -- Alex [1] As best as I understood how to use it; the documentation is a bit unclear! -- 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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Convert 183+SDP and 180+SDP into a single reply w/SDP
by yu＠yu-boot.ru 16 Jul '18

16 Jul '18

Hello. My uplink usually gives ringback as both 183 Progress with SDP and 180 Ringing also with (different) SDP. One of my clients can't handle that behavior and it just don't plays ringback at all. Is it possible to relay only last provisional response with SDP and drop anything else 183? Any ideas plz? Can't figure out even a logic itself. Thanks. -- sent from myMail for Android

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Encapsulate and de-encapsulate SIP traffic
by William Phillips 16 Jul '18

16 Jul '18

I have a need to encapsulate SIP traffic depending on destination and obviously de-encapsulate traffic incoming from those destinations. The approach to dealing with outbound SIP traffic seems pretty straight forward, I create an appropriate routing block that determines if I need to encap the SIP message, if I do, I forward it off to my module. The incoming traffic isn't particularly complicated, I'm just not 100% sure on the way to give that de-encapsulated SIP message back to kamailio to be processed. It looks like I can achieve this using the TM module with the uac_req_t struct and tm_bind.request but I really only see it used for SIP text messages in modules like XMPP and SMS. Is this the appropriate approach to achieve what I need? Will

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5.1.3 issue with rtpengine.reload
by Marcin Kowalczyk 16 Jul '18

16 Jul '18

Hi, I have Kam 5.1.3 running on top of Ubuntu Xenial docker image. Today I've added new RTPEngine set and tried to reload rtpengine from db but no luck. Command executes without any errors (but way too fast) root@dfa7aae0b6ef:/# time kamcmd rtpengine.reload real 0m0.002s user 0m0.000s sys 0m0.000s root@dfa7aae0b6ef:/# Last option would be restart but I would prefer not as has multiple WSS connections active. Do you have idea what I could try to reload rtpengine set?

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File transfer with Kamailio IMS??
by Ashok kumar Marra 16 Jul '18

16 Jul '18

Hi, I m using cent-os 6.2, kamailio-5.1.2 and openims-hss. I am using rtpengine to relay voice, but I am unable to send files or chat messages using it. I looked at rtpengine logs, it was saying unknown codec for chat and file-transfer requests. Then, I thought of using Kamailio msrp , tried but no success. Does current Kamailio IMS support file transfer? If so, please give me suggestions on doing it. -- With Best Regards, Ashok Kumar.

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rtpengine/kamailio teardown method
by Abdoul Osséni 16 Jul '18

16 Jul '18

Hello all, There are some rtpengine parameters that permit to send an RPC command to kamailio in order to terminate a dialog. - timeout = xx - b2b-url = http://xxxx:xx/RPC - xmlrpc-format = 1 Each rpc command sent to Kamailio exited with status 256. By checking the code of rtpengine, I see rtpengine seems to use "teardown" method. I think Kamailio does not know this method. Can you help me please ? Abdoul OSSENI Ingénieur DevOps chez Néo-Soft Co-Fondateur de ON SERVICES Tél : +33 601 135 167

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Communication between Kamailio Proxy and Registrar Server
by amit Kumar 16 Jul '18

16 Jul '18

Hi All, As per our requirement, we want to configure one Kamailio as a " Kamailio PROXY" and 2nd Kamailio as a " Kamailio REGISTRAR". Whenever any INVITE request come at " Kamailio PROXY" we want to get the UA information from " Kamailio REGISTRAR " and send INVITE to particular UA. Could you please let us know how achieve this. Thanks, Amit

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SIP/2.0 500 I'm terribly sorry, server error occurred (1/TM)
by Aqs Younas 13 Jul '18

13 Jul '18

Greeting list, I am using two async functions(http_async_query) in configuration one for authentication and other one for push notifications. I am also using tsilio module to suspend the transaction and resumed it. I am resetting this flag 'T_ASYNC_SUSPENDED' before suspending the transaction. Everything is working fine except. When caller send CANCEL on the call, kamailio successfully completes sip flow at callee(B party) side but send back below packet towards caller. 2018/07/10 19:47:41.125893 172.31.21.87:5060 -> 39.38.156.111:59969 SIP/2.0 500 I'm terribly sorry, server error occurred (1/TM ) Via: SIP/2.0/UDP 39.38.156.111:59969 ;branch=z9hG4bK-524287-1---91c6114330fe6293;rport=59969 To: <sip:923335550609@sip.talkhomeappcall.com ;transport=UDP>;tag=9fe2d4c84d97a4af81aa65ca8fc51b3d-62cb From: <sip:923135082344@sip.talkhomeappcall.com;transport=UDP>;tag=d72eaf4c Call-ID: VkAwxpSgkKSGn5dXaXgn5A.. CSeq: 2 INVITE Content-Length: 0 root@sip:~# kamailio -V version: kamailio 5.1.3 (x86_64/linux) 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: unknown compiled on 08:06:26 May 2 2018 with gcc 4.9.2 Could someone please help me out how to fix this. Cancels are handles automatically so, i cannot hardcore anything to resolve it. Best Regards, Aqs Younas

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Re: [SR-Users] sr-users Digest, Vol 158, Issue 12
by Ne Mo 13 Jul '18

13 Jul '18

Thanks Daniel! I'm folow: https://www.kamailio.org/docs/modules/stable/modules/ dispatcher.html#dispatcher.ex.install but my kamailio get errors: parameter <dstid_avp> of type <1> not found in module <dispatcher> parameter <attrs_avp> of type <1> not found in module <dispatcher> parameter <sock_avp> of type <1> not found in module <dispatcher> parameter <cnt_avp> of type <1> not found in module <dispatcher> I have no infor how to fix it Can you show me more detail about how to do: - *Fix error* * - Set the socket attribute to your private ip socket to do bridge from external to internal networks. * (just add to "attrs(str,opt)", and in my situation, it is my private IP of Kamailio server1 ? forexample: 1 sip:<PublicIP>:<Port> 0 0 duid=kamilioserver1;socket=<protocol>:<KamailioServer1_PrivateIP>:<Port>;my=xyz) (I don't understand what my=xyz mean..) * - Do i have to do something in Kamailio server2 and KamailioServer3 (or just create kamailio database each server and synchronyze kamailio database between them?)* *Thanks!* And thanks Mojtaba! I had read your recomment tutorial with OpenSIP, but do not know how to apply to my system. Have you got an other more clearly tutorial for this? On Thu, Jul 12, 2018 at 5:00 PM, <sr-users-request(a)lists.kamailio.org> wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.kamailio.org > > To subscribe or unsubscribe via the World Wide Web, visit > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.kamailio.org > > You can reach the person managing the list at > sr-users-owner(a)lists.kamailio.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. UAC register (Ben Hood) > 2. Re: UAC register (Daniel-Constantin Mierla) > 3. Re: UAC register (Daniel-Constantin Mierla) > 4. dispatcher module complete tutorial (SamSung TV) > 5. Re: UAC register (Ben Hood) > 6. Re: UAC register (Ben Hood) > 7. Re: dispatcher module complete tutorial (Mojtaba) > 8. Re: SIP/2.0 500 I'm terribly sorry, server error occurred > (1/TM) (Aqs Younas) > 9. Re: dispatcher module complete tutorial (Daniel-Constantin Mierla) > 10. Re: UAC register (Daniel-Constantin Mierla) > 11. Re: UAC register (Daniel-Constantin Mierla) > 12. Re: SIP/2.0 500 I'm terribly sorry, server error occurred > (1/TM) (Daniel-Constantin Mierla) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 11 Jul 2018 12:36:36 +0100 > From: Ben Hood <ben(a)relops.com> > To: sr-users(a)lists.sip-router.org > Subject: [SR-Users] UAC register > Message-ID: <7B1A15CF-7BCF-4BCC-8DF0-20E2906BA1DA(a)relops.com> > Content-Type: text/plain; charset="utf-8" > > Hi, > > I’m wondering if there is a flag somewhere in the UAC module config that > prevents remote registration from happening. > > It looks like I’ve loaded the uacreg table but I can’t see any REGISTER > methods being sent from Kamailio to the remote peer. > > The remote registrants are loaded using this config: > > listen=udp:172.16.176.150:5060 > > loadmodule "db_postgres.so" > loadmodule "kex.so" > loadmodule "corex.so" > loadmodule "tm.so" > loadmodule "tmx.so" > loadmodule "sl.so" > loadmodule "rr.so" > loadmodule "pv.so" > loadmodule "maxfwd.so" > loadmodule "textops.so" > loadmodule "siputils.so" > loadmodule "xlog.so" > loadmodule "sanity.so" > loadmodule "ctl.so" > loadmodule "cfg_rpc.so" > loadmodule "counters.so" > loadmodule "uac.so" > > modparam("uac", "reg_db_url", DBURL) > modparam("uac", "reg_contact_addr", “172.16.176.150:5060") > modparam("uac", "reg_db_table", "uacreg") > modparam("uac", "reg_timer_interval", 10) > modparam("uac", "reg_retry_interval", 10) > > request_route { > route(RELAY); > } > > route[RELAY] { > > t_on_reply("MANAGE_REPLY"); > > if (!t_relay()) { > sl_reply_error(); > } > exit; > } > > And it looks like this has been consumed by Kamailio: > > $ kamcmd uac.reg_dump > { > l_uuid: 2cc6776d-61ed-4ea1-a3ae-055f89bd4d3f > l_username: 126231943099723776 > l_domain: 172.16.176.150 > r_username: 1000 > r_domain: 172.16.176.129 > realm: 172.16.176.129 > auth_username: 1000 > auth_password: password > auth_proxy: sip:172.16.176.129:5080 > expires: 0 > flags: 0 > diff_expires: -1531235076 > timer_expires: 0 > reg_init: 1531234841 > reg_delay: 0 > } > > Doing a refresh successfully reloads the uacreg data > > $ kamcmd uac.reg_refresh 2cc6776d-61ed-4ea1-a3ae-055f89bd4d3f > > I can get Kamailio to forward an OPTIONS message to the remote peer () to > verify network connectivity, e.g. : > > $ru = "sip:1000@172.16.176.129:5080"; > $fs = "udp:172.16.176.150:5060"; > > But Kamailio does not appear to generate REGISTER messages either after > first boot or as a result of the manual refresh. > > This is using an instance of Kamailio 5.0.1. Should I try with a newer > version? > > Is there a way to get extra debugging of the uac module? > > If I increase the debug verbosity, I get quite granular detail for the DB > traffic, but the uac module doesn’t appear to emit much diagnostics. > > Many thanks, > > Ben > > > > > >

2 1

UAC register
by Ben Hood 12 Jul '18

12 Jul '18

Hi, I’m wondering if there is a flag somewhere in the UAC module config that prevents remote registration from happening. It looks like I’ve loaded the uacreg table but I can’t see any REGISTER methods being sent from Kamailio to the remote peer. The remote registrants are loaded using this config: listen=udp:172.16.176.150:5060 loadmodule "db_postgres.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "counters.so" loadmodule "uac.so" modparam("uac", "reg_db_url", DBURL) modparam("uac", "reg_contact_addr", “172.16.176.150:5060") modparam("uac", "reg_db_table", "uacreg") modparam("uac", "reg_timer_interval", 10) modparam("uac", "reg_retry_interval", 10) request_route { route(RELAY); } route[RELAY] { t_on_reply("MANAGE_REPLY"); if (!t_relay()) { sl_reply_error(); } exit; } And it looks like this has been consumed by Kamailio: $ kamcmd uac.reg_dump { l_uuid: 2cc6776d-61ed-4ea1-a3ae-055f89bd4d3f l_username: 126231943099723776 l_domain: 172.16.176.150 r_username: 1000 r_domain: 172.16.176.129 realm: 172.16.176.129 auth_username: 1000 auth_password: password auth_proxy: sip:172.16.176.129:5080 expires: 0 flags: 0 diff_expires: -1531235076 timer_expires: 0 reg_init: 1531234841 reg_delay: 0 } Doing a refresh successfully reloads the uacreg data $ kamcmd uac.reg_refresh 2cc6776d-61ed-4ea1-a3ae-055f89bd4d3f I can get Kamailio to forward an OPTIONS message to the remote peer () to verify network connectivity, e.g. : $ru = "sip:1000@172.16.176.129:5080"; $fs = "udp:172.16.176.150:5060"; But Kamailio does not appear to generate REGISTER messages either after first boot or as a result of the manual refresh. This is using an instance of Kamailio 5.0.1. Should I try with a newer version? Is there a way to get extra debugging of the uac module? If I increase the debug verbosity, I get quite granular detail for the DB traffic, but the uac module doesn’t appear to emit much diagnostics. Many thanks, Ben

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dispatcher module complete tutorial
by SamSung TV 12 Jul '18

12 Jul '18

I want to use kamailio as load balancing server my topo: (Private IP) Kamailio server2 / / (public iP)Kamailio server1(load balance)(Private IP) \ \ (Private IP) Kamailio server1 Any body can recomment me some tutorial to do this? Thanks any advance!

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Kamailio, forking and multiple answer
by Marino Mileti 10 Jul '18

10 Jul '18

Hi guys, i've a Kamailio with forking scenario using addbranch technique.Now I've a problem; i know it's very very very rare but it could be possible so it's better for me to find a solution. A -> B,C,D in forking parallel mode, so B C and D rings simultaneulsy. What is happen if B anser the call, and at the same time C decline the call?It seems that, using my cfg scripts, C sends DECLINE but this message never arrive to A because it receives a 200OK of B. How I can prevent this? I would like that if DECLINE is the first message, all other answers will be ignored (first answer win). Any help or comment? ThanksMarino

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TLS CRL configuration
by Amarnath Kanchivanam 10 Jul '18

10 Jul '18

Hi All, I'm trying to configured kamailio as TLS server with below configuration (tls.cfg) and TLS server is started successfully. [server:default] method = TLSv1+ verify_certificate = yes require_certificate = yes private_key = ./sip/server.key certificate = ./sip/server.crt ca_list = ./bundle.crt crl = ./sip_crl.pem verify_depth = 9 [client:default] verify_certificate = no require_certificate = no TLS connection works fine. Later i have updated the sip_crl.pem with server certificate revoked details and performed tls.reload command to load the latest update. After this I expect any TLS client trying to establish TLS connection should fail, as the client and server certificates are signed by same authority and server certificate is revoked. But the clients are able to establish TLS connection without any errors. I'm not getting any traces to confirm CRL validation has been performed before accepting the TLS connection. Any advice would be help to proceed with evaluating CRL functionality. -Amar

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CPS
by Sergiu Pojoga 10 Jul '18

10 Jul '18

Hi folks, Trying to implement CPS throttling based on this <https://lists.kamailio.org/pipermail/sr-users/2013-September/079713.html>article, having some trouble. modparam("htable", "htable", "rhs=>size=32;initval=0;autoexpire=300;" modparam("htable", "htable", "rhm=>size=32;initval=0;autoexpire=1800;") route[CPS] { $var(rateHashSec) = $si+":sec:"+$timef(%Y/%m/%d_%H_%M_%S); $var(rateHashMin) = $si+":min:"+$timef(%Y/%m/%d_%H_%M_00); xlog("L_INFO", "var(rateHashSec)=$var(rateHashSec); var(rateHashMin)=$var(rateHashMin)\n"); $var(ratePerSec) = $shtinc(rhs=>$var(rateHashSec)); $var(ratePerMin) = $shtinc(rhm=>$var(rateHashMin)); xlog("L_INFO", "var(ratePerSec)=$var(ratePerSec); var(ratePerMin)=$var(ratePerMin)\n"); $var(limitPerSec) = 1; $var(limitPerMin) = 1; if ($var(ratePerSec) > $var(limitPerSec) || $var(ratePerMin) > $var(limitPerMin)) { xlog("L_INFO", "CPS Limit on IP $si\n"); send_reply("503", "CPS Limit on IP $si"); exit; } } Logs show hashes are created but *$var(ratePerSec/Min)* won't increment no matter how many calls I place per second or minute: Jul 9 14:35:52 proxy /usr/local/sbin/kamailio[1484]: INFO: <script>: var(rateHashSec)=65.39.XX.XX:sec:2018/07/09_14_35_52; var(rateHashMin)=65.39.XX.XX:min:2018/07/09_14_35_00 Jul 9 14:35:52 proxy /usr/local/sbin/kamailio[1484]: INFO: <script>: var(ratePerSec)=0; var(ratePerMin)=0 Much obliged. --Sergiu

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