- sr-users - kamailio.org

Re: [SR-Users] NATHELPER issue
by Konstantin Polyakov 23 Jul '18

23 Jul '18

Hello Ali, ACK is sent by UAC to the Contact which is received in OK from UAS. So from my point of view you need to fix that Contact in OK on your proxy. My NATDETECT looks like: if (nat_uac_test("19")) { fix_nated_contact(); } I call it for requests and responses. Best regards. Konstantin >Message: 23 >Date: Fri, 20 Jul 2018 11:13:42 +0300 >From: "Ali Taher" < ataher(a)vanrise.com > >To: < sr-users(a)lists.sip-router.org > >Subject: [SR-Users] NATHELPER issue >Message-ID: < 070901d42001$939be2b0$bad3a810$(a)vanrise.com > >Content-Type: text/plain; charset="utf-8" > >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 > > "Ali Taher" < ataher(a)vanrise.com >

2 1

Selectively enable / disable cdrs using acc module
by M S 23 Jul '18

23 Jul '18

Hi, I have a kamailio v5.0 setup where cdrs are enabled using acc module. Is there a way to enable / disable cdrs on per call basis in kamailio script? For example i want to allow cdrs only for offnet calls and disable them for onnet calls? May be a dialog flag configurable in module parameters to signal acc module whether or not to generate cdr for current dialog at the end of call?

2 5

Options for SIP traffic duplicating to third party tcp port
by Максим Баранов 23 Jul '18

23 Jul '18

Hello, what are the options for duplicating sip traffic to arbitrary tcp port in Kamailio?

3 5

forbidden domain not served
by eyas barhouk 22 Jul '18

22 Jul '18

Dears i'm using kamilio 5.2.0 as IMS with Zoiper client and RTP Engine when i'm trying to terminate a video call ,BYE message does not arrived to destination with error from S-cscf side "forbidden domain not served " has any one had the same issue before ? and is there any ideas about how to solve it ? ? thanks in advance

2 3

dialog module with dmq
by E. Schmidbauer 22 Jul '18

22 Jul '18

Is anyone using dialog module with DMQ to share dialogs across multiple kamailio instances? i've been testing it and it seems unreliable. sometimes the dialog data is sent from nodeA to nodeB, sometimes it is not. im wondering if im missing some settings or it's a bug. I tested latest master branch and version 5.1.4 Thanks! Emmanuel

3 2

$var(name) - Private memory variables
by Sergey Safarov 21 Jul '18

21 Jul '18

Hello I have faced when case when private var defined as $var(json_request) is share between different SIP messages. In my case first time this var is used from registration processing route. Then var with same name is used PUBLISH messge processing. And i be able to this value defined from registration processing. I expecting that vars is not shared between threads and message handling routes for var defined as $var(name). Is that not true? How i can prevent race condition between different message processing routes when two thread simultaneously change same variable? Sergey

3 5

uri==myself
by Karthik Srinivasan 20 Jul '18

20 Jul '18

Hello, Can someone explain to me why statement A returns True and statement B returns False. Statement A: is_myself("$ru") Statement B: uri == myself Isn't uri and $ru referencing the same data? With my current config i am expecting the is_myself to return True; i was expecting the same for Statement B but unfortunately it is returning False. Thanks, Karthik

2 11

how to configure ims in kamailio
by vinod mn 20 Jul '18

20 Jul '18

Hi what are the requirements to configure the ims to work with kamailio. -- Thanks and regards Vinod.M.N

3 2

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

1 0

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 ?

3 3

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

2 5

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/

2 4

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.

1 0

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

2 4

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?

2 2

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

2 1

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

3 6

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

3 8

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?

2 4

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/

2 1

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

5 7

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

2 1

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?

2 1

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.

2 1

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

2 1

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