- sr-users - kamailio.org

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

28 Oct '20

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

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

27 Oct '20

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

1 0

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

27 Oct '20

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

1 1

KEMI and Native mixing
by Marcin Kowalczyk 26 Oct '20

26 Oct '20

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

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

26 Oct '20

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

2 1

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

25 Oct '20

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

1 0

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

22 Oct '20

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

1 0

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

22 Oct '20

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

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

21 Oct '20

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

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

21 Oct '20

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

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Kamailio + rtpengine
by Amit Pal 21 Oct '20

21 Oct '20

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

2 1

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

21 Oct '20

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

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

21 Oct '20

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

1 0

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

20 Oct '20

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

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

20 Oct '20

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

2 1

No retry on INVITE
by Allan Kristensen 20 Oct '20

20 Oct '20

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

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

20 Oct '20

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

1 0

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

20 Oct '20

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

1 0

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

20 Oct '20

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

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dialog_active_dialogs erroneous value
by Sergiu Pojoga 20 Oct '20

20 Oct '20

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

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TOPOS module and fix_nated_contact
by Minh Phan 19 Oct '20

19 Oct '20

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

2 1

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

19 Oct '20

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

2 1

Kamailio IMS RTP Engine
by Pavithra Mohanraja 19 Oct '20

19 Oct '20

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

2 2

Libphonenumber - Updating metadata
by Zach Nielsen 18 Oct '20

18 Oct '20

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

3 3

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

17 Oct '20

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

2 5

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

17 Oct '20

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

3 6

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

16 Oct '20

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

2 1

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

16 Oct '20

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

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

15 Oct '20

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

2 2

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

15 Oct '20

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

5 5

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

15 Oct '20

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

1 0

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

15 Oct '20

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

1 0

To Header Manipulation
by Sam Ware 15 Oct '20

15 Oct '20

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

2 1

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

15 Oct '20

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

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

14 Oct '20

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

1 0

SRTP
by Andy Kama 14 Oct '20

14 Oct '20

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

3 3

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

14 Oct '20

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

2 2

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

14 Oct '20

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

3 7

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

14 Oct '20

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

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

14 Oct '20

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

1 0

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

13 Oct '20

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

2 2

Caseless regex
by Marat Gareev 13 Oct '20

13 Oct '20

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

3 5

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

13 Oct '20

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

2 2

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

13 Oct '20

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

4 9

Permissions address
by David Villasmil 13 Oct '20

13 Oct '20

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

3 3

Dispatcher stops working
by Sergio Charrua 12 Oct '20

12 Oct '20

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

4 4

Logging problem
by Jan Rozhon 12 Oct '20

12 Oct '20

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

4 7

Local cache registration
by David VILLAUME 11 Oct '20

11 Oct '20

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

2 1

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

09 Oct '20

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

2 2

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

09 Oct '20

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

3 3

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