- sr-users - kamailio.org

Memory allocation issue
by Logeshwaran G 26 Sep '17

26 Sep '17

Hi, I am having the below Error while executing the kamailio cfg file. Before an hour it executed successfully, There is no changes in cfg. loading modules under config path: /usr/local/lib64/kamailio/modules/ 0(27575) : <core> [core/mem/shm.c:105]: shm_getmem(): could not attach shared memory segment: Cannot allocate memory 0(27575) : <core> [core/mem/q_malloc.c:1203]: qm_malloc_init_shm_manager(): could not initialize qm shm memory pool Too much qm shm memory demanded: 67108864 bytes 0(27575) CRITICAL: <core> [core/cfg.y:3414]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/config/kamailio_final.cfg, line 302, column 39: Can't initialize shared memory ERROR: bad config file (1 errors) Input please? Thanks & Kind Regards, *Logeshwaran G* *"Be Hungry, Stay Foolish"*

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Docker image
by Sergey Safarov 25 Sep '17

25 Sep '17

Hello guys Yes i packaged kamailio to docker container <https://hub.docker.com/r/safarov/kamailio3/>. This container is ready to run kamailio in docker environment. Could you make review of this. Dockerfiles you can review at https://github.com/kamailio/kamailio/tree/master/pkg/kamailio/alpine_docker Build process logs you can review at https://hub.docker.com/r/safarov/kamailio3/builds/bl9rv6ygeubwnydh4an2tuc/ Sergey

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Astricon 2017
by Daniel-Constantin Mierla 25 Sep '17

25 Sep '17

Hello, Kamailio will be present at Astricon 2017 having a stand in the expo area (Orlando, FL, during October 3-5 -- more details at www.astricon.net). Should you participate at the event and have flyers/other materials with products/solutions/services that use Kamailio or giveaways, you are welcome to join the booth or just drop them there, we are more than happy to promote and spread them. Also, you are very welcome to just stop by and have a chat. Besides the activity in the exhibition area, I and Fred Posner will have presentations about using Kamailio together with Asterisk or other VoIP applications. Be sure you do not miss them, the agenda is available at: - http://www.asterisk.org/community/astricon-user-conference/schedule Expect a consistent group from Kamailio community, it's a great event to be at! Looking forward to meeting many of you next week in Orlando! Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - www.kamailioworld.com

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Re: [SR-Users] [sr-dev] SIP ping, sockets and multi homed
by Patrick Wakano 25 Sep '17

25 Sep '17

Hello list, After deep investigation I found out that the issue is happening because of the mechanism Kamailio uses to find the most appropriate socket combined with the virtual IP usage.... For socket selection, Kamailio first asks Linux which would be the routing for a certain destination, then matches this outcome with the list of listened sockets. By default Linux will always return the local IP instead of the virtual one. So if Kamailio only binds to the virtual IP, the Linux's returned IP won't match our socket list and then the socket selection fails.... So currently Kamailio active/stand-by HA setup with virtual IP in a multi-homed environment does not work properly... We can overcome this by listening to all addresses, but then Kamailio will decide to use the local IP instead of the virtual one, and then the HA idea end up broken.... We can also force socket for everything. For regular call routing, this options actually works fine if the script always force some socket, but it seems to be not always possible for local generated messages. By disabling mhomed and using the dispatcher socket attr to send the OPTIONS message it works for example. But for the LCR module we have no option because apparently the LCR module has only one socket option, and so I am not able have a different socket per gateway. We can also explicitly set the src address for a routing rule to be the virtual IP instead of the local one. This makes Linux to return the virtual IP instead of the local one. This solution seems fine, but since it is done at linux level, every other service on the box would be affected and every route on the box would needed to be changed.. Another problem I found out is that I am not able to force the socket for local generated messages. It actually crashes Kamailio, and I always get a "core was not generated" msg.... anyone aware of that? The actual fix for this, would be having Kamailio to search the most appropriate socket in a different manner. It could loop the sockets and see which one reaches the destination instead of asking Linux to do it, or it could have some routing directives which could be consulted before selecting a socket.... (just ideas, not sure how feasible they are...) I am considering using the Linux routing change option, but has anyone faced such situation before? Regards, Patrick Wakano On 24 August 2017 at 09:18, Patrick Wakano <pwakano(a)gmail.com> wrote: > Hello Daniel, > I still couldn't make this SIP probing work.... So currently my setup > cannot actively check for GW availability... > Any idea about what may be causing it? > > Thanks for your time, > Patrick Wakano > > On 16 August 2017 at 13:01, Patrick Wakano <pwakano(a)gmail.com> wrote: > >> Forgot to mention it is a centos7 box... >> >> On 16 Aug. 2017 11:02 am, "Patrick Wakano" <pwakano(a)gmail.com> wrote: >> >>> Hi Daniel! >>> Very grateful for your attention! >>> >>> Apologies for lacking important info, check below the version and >>> sockets. >>> It may be worth mentioning these IPs are virtual ones and I am using net.ipv4.ip_nonlocal_bind >>> = 1(in this case the virtual IP is assigned to the box) >>> Interesting that it has a "mhomed: no" for all sockets but I do have >>> mhomed=1 in my cfg file.... could that be the issue? >>> Also there is no message before this one: /usr/sbin/kamailio[6164]: >>> ERROR: <core> [core/forward.c:181]: get_out_socket(): no socket found >>> This is the first one that pops when dispatcher tries to ping.... >>> >>> Thanks, >>> Patrick >>> >>> kamailio -v >>> version: kamailio 5.0.2 (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 12:03:49 Jun 26 2017 with gcc 4.8.5 >>> >>> kamcmd> core.sockets_list >>> { >>> socket: { >>> proto: udp >>> address: 172.28.128.100 >>> ipaddress: 172.28.128.100 >>> port: 5060 >>> mcast: no >>> mhomed: no >>> } >>> socket: { >>> proto: udp >>> address: 192.168.33.100 >>> ipaddress: 192.168.33.100 >>> port: 5060 >>> mcast: no >>> mhomed: no >>> } >>> socket: { >>> proto: tcp >>> address: 172.28.128.100 >>> ipaddress: 172.28.128.100 >>> port: 5060 >>> mcast: no >>> mhomed: no >>> } >>> socket: { >>> proto: tcp >>> address: 192.168.33.100 >>> ipaddress: 192.168.33.100 >>> port: 5060 >>> mcast: no >>> mhomed: no >>> } >>> socket: { >>> proto: tcp >>> address: 172.28.128.100 >>> ipaddress: 172.28.128.100 >>> port: 8080 >>> mcast: no >>> mhomed: no >>> } >>> } >>> >>> >>> On 15 August 2017 at 17:24, Daniel-Constantin Mierla <miconda(a)gmail.com> >>> wrote: >>> >>>> Hello, >>>> >>>> what are the sockets kamailio is listening on? See `kamcmd help` for >>>> the command that lists the sockets. >>>> >>>> Also, what is the log messages before the first one you pasted, there >>>> should be another one printed from get_out_socket(). >>>> It is always important to provide kamailio version (kamailio -v) and >>>> the operating system you are using. >>>> >>>> Cheers, >>>> Daniel >>>> >>>> >>>> On 14.08.17 14:41, Patrick Wakano wrote: >>>> >>>> Hello Kamailio list, >>>> >>>> Hope you all doing well! >>>> >>>> I am stuck with one problem that should be incredibly easy to be done, >>>> but apparently it is not.... I want my dispatcher, LCR and nathelper >>>> modules to always SIP ping my SIP clients. I have a Kamailio based SBC that >>>> is separating two networks, and so I am using the 'mhomed' flag, but none >>>> of these SIP pings are working.... for now I will only ask about the >>>> dispatcher.... >>>> My dispatcher module has to talk with SIP servers in both networks, the >>>> thing is when I enable SIP ping for it (modparam("dispatcher", >>>> "ds_probing_mode", 1)), it gives me this type of error: >>>> /usr/sbin/kamailio[6164]: ERROR: <core> [core/forward.c:181]: >>>> get_out_socket(): no socket found >>>> /usr/sbin/kamailio[6164]: ERROR: <core> [core/forward.c:183]: >>>> get_out_socket(): no corresponding socket found for(udp: >>>> 192.168.33.110:6060) >>>> /usr/sbin/kamailio[6164]: ERROR: tm [ut.h:317]: uri2dst2(): no >>>> corresponding socket found for "192.168.33.110" af 2 (udp: >>>> 192.168.33.110:6060) >>>> /usr/sbin/kamailio[6164]: ERROR: tm [uac.c:443]: t_uac_prepare(): no >>>> socket found >>>> /usr/sbin/kamailio[6164]: ERROR: dispatcher [dispatch.c:2652]: >>>> ds_ping_set(): unable to ping [sip:192.168.33.110:6060] >>>> >>>> After research it seems I need to manually set a "socket" attribute for >>>> each of my GWs in dispatcher list. After putting the socket parameter for >>>> the GW, I got another error: >>>> /usr/sbin/kamailio[14190]: ERROR: <core> [core/socket_info.c:2046]: >>>> parse_protohostport(): bad port number in udp:192.168.33.100:5060 >>>> /usr/sbin/kamailio[14190]: ERROR: <core> [core/forward.c:181]: >>>> get_out_socket(): no socket found >>>> /usr/sbin/kamailio[14190]: ERROR: <core> [core/forward.c:183]: >>>> get_out_socket(): no corresponding socket found for(udp: >>>> 192.168.33.110:6060) >>>> /usr/sbin/kamailio[14190]: ERROR: tm [ut.h:317]: uri2dst2(): no >>>> corresponding socket found for "192.168.33.110" af 2 (udp: >>>> 192.168.33.110:6060) >>>> /usr/sbin/kamailio[14190]: ERROR: tm [uac.c:443]: t_uac_prepare(): no >>>> socket found >>>> /usr/sbin/kamailio[14190]: ERROR: dispatcher [dispatch.c:2652]: >>>> ds_ping_set(): unable to ping [sip:192.168.33.110:6060] >>>> >>>> What can be wrong in my port info in the socket config? (I tried the >>>> socket with and without quotes but had no luck...) >>>> select * from dispatcher where id=2; >>>> id | setid | destination | flags | priority >>>> | attrs | description >>>> ----+-------+-------------------------+-------+----------+-- >>>> -------------------------------------------+--------------------- >>>> 2 | 2 | sip:192.168.33.110:6060 | 0 | 0 | socket="udp: >>>> 192.168.33.100:5060";rweight=50 | >>>> (1 row) >>>> >>>> Anyhow, I was expecting the 'mhomed' parameter to do the socket >>>> selection for me. Regular call routing either by the dispatcher or by LCR, >>>> also seems to require me to explicitly set the socket otherwise Kamailio >>>> (or the OS) may end up using the wrong socket. So, another question is why >>>> 'mhomed' does not dictate the whole socket selection Kamailio has to do >>>> regardless the module it is using to send SIP requests? >>>> >>>> Sorry for the long e-mail.... >>>> Best regards, >>>> Patrick Wakano >>>> >>>> >>>> _______________________________________________ >>>> Kamailio (SER) - Development Mailing Listsr-dev@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-dev >>>> >>>> >>>> -- >>>> Daniel-Constantin Mierlawww.twitter.com/miconda -- www.linkedin.com/in/miconda >>>> Kamailio Advanced Training - www.asipto.com >>>> Kamailio World Conference - www.kamailioworld.com >>>> >>>> >>> >

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LCR Ping GWs
by Patrick Wakano 25 Sep '17

25 Sep '17

Hello list, I was trying to activate continuous SIP ping from the LCR module, which apparently was available in the past, but currently it is not possible anymore. As I could check It can only ping inactive GWs. Question is, why such change was promoted? We have to send a call to the GW to detect it has died instead of actively monitoring them and so avoid this useless call attempt. It is not that big deal, but it does gives me a better monitoring of my providers (specially the ones not used so frequently) health. Regards, Patrick

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Enabling Redirection in Kamailio SIP Server
by Anas Rasheed 23 Sep '17

23 Sep '17

How can we enable redirection in Kamilio SIP Server using Siremis User Interface for it? Can anyone provide us with the tutorial of it?

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Memory configuration
by Joe Baran 23 Sep '17

23 Sep '17

I feel fairly confident I’m running into a memory tuning issue, but I’m struggling to determine the next step. Can someone give me some guidance on how to adjust settings to allow the system to stop crashing kamailio? Sep 14 18:25:15 server1 kamailio[29468]: ERROR: <core> [core/mem/q_malloc.c:292]: qm_find_free(): qm_find_free(0x7fa9eeeea000, 144); Free fragment not found! Sep 14 18:25:15 server1 kamailio[29468]: ERROR: <core> [core/mem/q_malloc.c:425]: qm_malloc(): qm_malloc(0x7fa9eeeea000, 144) called from kazoo: kz_amqp.c: kz_amqp_async_query(1357), module: kazoo; Free fragment not found! Sep 14 18:25:15 server1 kamailio[29468]: ERROR: kazoo [kz_amqp.c:1359]: kz_amqp_async_query(): failed to allocate kz_amqp_cmd in process 29468 Sep 14 18:25:15 server1 kamailio[29468]: INFO: <script>: kGddp3V0bA6nitnk25dB4w..|log|failed to send Kazoo query for authentication xxxxxxxxxxxxxxxxxx Sep 14 18:25:15 server1 kamailio[29468]: ERROR: tm [t_reply.c:494]: _reply_light(): cannot allocate shmem buffer Sep 14 18:25:15 server1 kamailio[29468]: ERROR: sl [sl.c:271]: send_reply(): failed to reply stateful (tm) Sep 14 18:25:16 server1 kamailio[29483]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 108 Sep 14 18:25:16 server1 kamailio[29360]: ALERT: <core> [main.c:743]: handle_sigs(): child process 29438 exited by a signal 11 Sep 14 18:25:16 server1 kamailio[29360]: ALERT: <core> [main.c:746]: handle_sigs(): core was not generated Thanks, Joe Baran

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Shared htable between servers
by Iskren Hadzhinedev 22 Sep '17

22 Sep '17

Hi list, Is it possible to have a htable shared between servers on different machines? I have two kamailio proxies on different servers for resilience and I'm storing call information in a hash table. Then I can route calls for the same user to the respective PBX. The reason for this is to have attended transfers working reliably (attended transfer between PBXes are a mess). Thank you! Kind regards, -- /Iskren Hadzhinedev/

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Call transfer between pbx servers behind kamailio proxy
by Iskren Hadzhinedev 22 Sep '17

22 Sep '17

Hi everyone, I'm having a hard time transferring calls when the users are on different FreeSWITCH servers behind kamailio. The users are distributed as follows: user1(a)sip.test.com -> Kamailio - ds_select_dst(1,1) -> FreeSWITCH1 user2(a)sip.test.com-> Kamailio - ds_select_dst(1,1) -> FreeSWITCH1 user3(a)sip.test.com -> Kamailio - ds_select_dst(1,1) -> FreeSWITCH2 When user1 calls user2 and user2 does an attended transfer to user3, the following happens: 285 67.220453 USER2_IP -> KAMAILIO_IP SIP 726 Request: REFER sip:mod_sofia@FreeSWITCH1_IP:5060, in-dialog | 286 67.220674 KAMAILIO_IP -> FreeSWITCH1_IP SIP 877 Request: REFER sip:mod_sofia@FreeSWITCH1_IP:5060, in-dialog | 287 67.223651 FreeSWITCH1_IP -> KAMAILIO_IP SIP 925 Status: 202 Accepted | 288 67.223753 KAMAILIO_IP -> USER2_IP SIP 838 Status: 202 Accepted | 289 67.225502 FreeSWITCH1_IP -> KAMAILIO_IP SIP/SDP 1225 Request: INVITE sip:user3@sip.test.com | 290 67.387323 KAMAILIO_IP -> 69.172.200.109 SIP/SDP 1420 Request: INVITE sip:user3@sip.test.com | Turns out 69.172.200.109 is the actual IP address of sip.test.com (I've probably should have gone with sip.example.com or something else). What is the proper way to route that INVITE back to the other FreeSWITCH server so that the transfer would work? Thanks! Kind regards, -- /Iskren Hadzhinedev/

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how to play ring tune when callee decline s
by 赵国杰 22 Sep '17

22 Sep '17

Hi guys, I want the caller to play a short audio(like "the number your are calling is busy") when the callee declines the call. How can i do that? Best Regards Jesse

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Typos in Kamctl Stats results
by Noah Mehl 21 Sep '17

21 Sep '17

Hey, I’m noticing that the results from the websockets module when running # kamctl stats there are some odd results that may just be typos? "websocket:ws_msrp_max_concurrent_connections = 0", "websocket:ws_msrp_max_concurrent_connectons = 0”, "websocket:ws_sip_max_concurrent_connections = 0", "websocket:ws_sip_max_concurrent_connectons = 0", Thanks! ~Noah

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Encryption of voice stream
by Vasan VS 21 Sep '17

21 Sep '17

Hello all, Kindly let us know if we can have a mechanism to have the voice stream in using the SIP server completely encrypted as well as prove that it is encrypted using any speific mechanism. Thanks in advance Vasan

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COALESCE Transformation?
by Brandon Armstead 20 Sep '17

20 Sep '17

Hello all, I've checked the docs and did some brief searching, curious if this transformation or method exists currently? Such as: https://dev.mysql.com/doc/refman/5.7/en/comparison-operators.html#function_… Thanks! -- Sincerely, Brandon Armstead CTO / CRYY.com

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private memory leak - but where?
by Sebastian Damm 20 Sep '17

20 Sep '17

Hi, I'm trying to debug a Kamailio which runs out of memory after a while. But following the hints at https://www.kamailio.org/wiki/tutorials/troubleshooting/memory doesn't help. I know it is the private memory that I'm missing. I already increased the pkg size to 16M, but somehow that seems to be not enough. The Kamailio receives HEP messages, sends them to a lua script where it is processed. Overall there are not more than 10 custom private variables used. In the log I'm seeing these messages: Sep 19 17:04:42 salm /usr/sbin/kamailio[19134]: ERROR: app_lua [app_lua_api.c:726]: app_lua_run_ex(): error executing: handle_packet (err: 2) Sep 19 17:04:42 salm /usr/sbin/kamailio[19134]: ERROR: <script>: SCRIPT: failed to execute lua function! Sep 19 17:04:42 salm /usr/sbin/kamailio[19134]: ERROR: <core> [core/mem/q_malloc.c:292]: qm_find_free(): qm_find_free(0x7f9744eea010, 232); Free fragment not found! Sep 19 17:04:42 salm /usr/sbin/kamailio[19134]: ERROR: <core> [core/mem/q_malloc.c:425]: qm_malloc(): qm_malloc(0x7f9744eea010, 232) called from core: core/parser/msg_parser.c: get_ hdr_field(116), module: core; Free fragment not found! Sep 19 17:04:42 salm /usr/sbin/kamailio[19134]: ERROR: <core> [core/parser/msg_parser.c:118]: get_hdr_field(): out of memory Sep 19 17:04:42 salm /usr/sbin/kamailio[19134]: ERROR: <core> [core/parser/msg_parser.c:330]: parse_headers(): bad header field [Via: SIP/2.0/WSS o6m] Sep 19 17:04:42 salm /usr/sbin/kamailio[19134]: ERROR: pv [pv_core.c:647]: pv_get_callid(): cannot parse Call-Id header kamcmd pkg.stats for this process gives me this output. { entry: 61 pid: 19134 rank: 61 used: 8748080 free: 4384 real_used: 16772832 total_size: 16777216 total_frags: 30 } Now I tried to get a dump of what is in use, following the guidelines on the wiki page mentioned above, but without success. I can easily start and stop the traffic coming in to Kamailio (by starting or stopping the captagent sending it there), but I can't get any output. Here's what I did: captagent stop kamcmd cfg.set_now_int core memlog 1 kamcmd cfg.set_now_int core mem_dump_pkg 19134 captagent start ; sleep 1 ; captagent stop (I started captagent so that the process actually had something to do because that's what the wiki page says it needs for printing memory information.) During that second, there were quite a lot of packets running through this process, but I did not get a memory dump. A shm_summary dump however works as expected. What am I missing? How do I get more information about what this process is using its memory for? TIA Sebastian

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sipp audio file
by Diego Nadares 19 Sep '17

19 Sep '17

Hi guys, Any body know how to create pcap (with audio) file to use with sipp?  <nop> <action> <exec play_pcap_audio="../scenarios/pcap/demo.cap"/> </action> </nop> I tried with wireshark capturing rtp but I'm doing something wrong i think. Thanks! Diego

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jsonrpcc problem
by Adam Kuśmirek 19 Sep '17

19 Sep '17

Hello, I have a problem with jsonrpcc module. OS is Debian Jessie. I tried with packeged kamailio 4.4 and compiled kamailio 5.0. Kamailio starts properly. Routing block is very simple: route { jsonrpc_request("get_source", "{'ipv4': '$si', 'port': $sp}", "RESPONSE", "ERROR", "$var(result)"); exit; } route[RESPONSE] { xlog("Result received: $var(result)"); } route[ERROR] { xlog("Error received: $var(result)"); } If any message arrives, kamailio sends it to my rpc_server (netstring encoded): 106:{'id': 2, 'jsonrpc': '2.0', 'method': 'get_source', 'params': {'ipv4': '10.20.121.231', 'port': 5060}}, My server decodes the message and immediately answers with netstring encoded response: 47:{"jsonrpc": "2.0", "result": "200 OK", "id": 2}, At this moment kamailio logs errors and disconnects from rpc server: root@kamailio-test:/etc/kamailio# 4(925) DEBUG: <core> [parser/msg_parser.c:595]: parse_msg(): SIP Request: 4(925) DEBUG: <core> [parser/msg_parser.c:597]: parse_msg(): method: <REGISTER> 4(925) DEBUG: <core> [parser/msg_parser.c:599]: parse_msg(): uri: <sip:kamailio.conpeek.com> 4(925) DEBUG: <core> [parser/msg_parser.c:601]: parse_msg(): version: <SIP/2.0> 4(925) DEBUG: <core> [parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK1033201528>; state=16 4(925) DEBUG: <core> [parser/parse_via.c:2639]: parse_via(): end of header reached, state=5 4(925) DEBUG: <core> [parser/msg_parser.c:486]: parse_headers(): Via found, flags=2 4(925) DEBUG: <core> [parser/msg_parser.c:488]: parse_headers(): this is the first via 4(925) DEBUG: <core> [receive.c:178]: receive_msg(): After parse_msg... 4(925) DEBUG: <core> [receive.c:221]: receive_msg(): preparing to run routing scripts... 4(925) DEBUG: tm [t_lookup.c:1312]: t_newtran(): DEBUG: t_newtran: msg id=1 , global msg id=0 , T on entrance=0xffffffffffffffff 4(925) DEBUG: <core> [parser/parse_addr_spec.c:864]: parse_addr_spec(): end of header reached, state=10 4(925) DEBUG: <core> [parser/msg_parser.c:171]: get_hdr_field(): <To> [40]; uri=[sip:1000@kamailio.conpeek.com] 4(925) DEBUG: <core> [parser/msg_parser.c:172]: get_hdr_field(): to body ["1000" <sip:1000@kamailio.conpeek.com> ] 4(925) DEBUG: <core> [parser/msg_parser.c:152]: get_hdr_field(): cseq <CSeq>: <1> <REGISTER> 4(925) DEBUG: <core> [parser/msg_parser.c:183]: get_hdr_field(): content_length=0 4(925) DEBUG: <core> [parser/msg_parser.c:89]: get_hdr_field(): found end of header 4(925) DEBUG: tm [t_lookup.c:466]: t_lookup_request(): t_lookup_request: start searching: hash=43020, isACK=0 4(925) DEBUG: tm [t_lookup.c:424]: matching_3261(): DEBUG: RFC3261 transaction matching failed 4(925) DEBUG: tm [t_lookup.c:648]: t_lookup_request(): DEBUG: t_lookup_request: no transaction found 4(925) DEBUG: <core> [md5utils.c:67]: MD5StringArray(): MD5 calculated: 920bfe28c581e1bf18554e3a9b234ceb 4(925) DEBUG: <core> [usr_avp.c:631]: destroy_avp_list(): destroying list (nil) 4(925) DEBUG: <core> [usr_avp.c:631]: destroy_avp_list(): destroying list (nil) 4(925) DEBUG: <core> [usr_avp.c:631]: destroy_avp_list(): destroying list (nil) 4(925) DEBUG: <core> [usr_avp.c:631]: destroy_avp_list(): destroying list (nil) 4(925) DEBUG: <core> [usr_avp.c:631]: destroy_avp_list(): destroying list (nil) 4(925) DEBUG: <core> [usr_avp.c:631]: destroy_avp_list(): destroying list (nil) 4(925) DEBUG: <core> [xavp.c:446]: xavp_destroy_list(): destroying xavp list (nil) 4(925) DEBUG: <core> [receive.c:318]: receive_msg(): cleaning up *0(921) ALERT: <core> [main.c:740]: handle_sigs(): child process 929 exited by a signal 11* * 0(921) ALERT: <core> [main.c:743]: handle_sigs(): core was generated* 0(921) INFO: <core> [main.c:755]: handle_sigs(): terminating due to SIGCHLD 0(921) DEBUG: <core> [main.c:757]: handle_sigs(): terminating due to SIGCHLD 7(928) INFO: <core> [main.c:810]: sig_usr(): signal 15 received 6(927) INFO: <core> [main.c:810]: sig_usr(): signal 15 received 5(926) INFO: <core> [main.c:810]: sig_usr(): signal 15 received 4(925) INFO: <core> [main.c:810]: sig_usr(): signal 15 received 3(924) INFO: <core> [main.c:810]: sig_usr(): signal 15 received 2(923) INFO: <core> [main.c:810]: sig_usr(): signal 15 received 1(922) INFO: <core> [main.c:810]: sig_usr(): signal 15 received 0(921) DEBUG: tm [t_funcs.c:86]: tm_shutdown(): DEBUG: tm_shutdown : start 0(921) DEBUG: tm [t_funcs.c:89]: tm_shutdown(): DEBUG: tm_shutdown : emptying hash table 0(921) DEBUG: tm [h_table.c:127]: free_cell_helper(): freeing transaction 0x7fb99a822350 from h_table.c:449 0(921) DEBUG: tm [t_funcs.c:91]: tm_shutdown(): DEBUG: tm_shutdown : removing semaphores 0(921) DEBUG: tm [t_funcs.c:93]: tm_shutdown(): DEBUG: tm_shutdown : destroying tmcb lists 0(921) DEBUG: tm [t_funcs.c:96]: tm_shutdown(): DEBUG: tm_shutdown : done 0(921) INFO: <core> [sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized 0(921) DEBUG: <core> [mem/shm.c:174]: shm_core_lock_destroy(): destroying the shared memory lock 0(921) DEBUG: <core> [mem/shm.c:266]: shm_destroy_manager(): destroying memory manager: f_malloc 0(921) DEBUG: <core> [mem/pkg.c:89]: pkg_destroy_manager(): destroying memory manager: f_malloc The problem occures also if rpc server does not reply, i think when trying to start ERROR route. If I try with jsonrpc_notification, everything is OK. Any help ? Regards Adam

2 2

See how many workers are actually processing packets?
by Sebastian Damm 19 Sep '17

19 Sep '17

Hi, I think this has been answered before, but I don't know the answer. Is there a way to find out, how many workers are busy right now processing packets? For a new setup I don't know whether I need 16 or 96 UDP listeners, but I'd like to scale it correctly. The setup just inspects HEP messages and sends some of them to a RabbitMQ. So TM or SL stats probably won't help. Best Regards Sebastian

1 0

Proxy Invite to registered natted user
by eskornev 19 Sep '17

19 Sep '17

Hi All Trying to set Kamailio as proxy in front of Asterisk. When natted user registers is remembered in Kamailio via save("location"). Kamailio forwards Register to * with new contact field (pointing to Kamailio). But when server originates Invite to user, kamailio transfer request to "fake" address that was in original Register contact field (getting it from lookup("location")). How can i set Kamailio to send Invite to source ip and port of user?

2 1

kamailio jsonrpc client
by Aidar Kamalov 19 Sep '17

19 Sep '17

Hello, cannot get working rpc client n kamailio. [root@sipchel kamailio]# grep 'json\|jansson' kamailio-default.cfg loadmodule "jsonrpcs.so" loadmodule "jansson.so" loadmodule "janssonrpcc.so" # jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) modparam("jsonrpcs", "transport", 3) # jansson modparam("janssonrpcc", "server", "conn=user_db;addr=127.0.0.1;port=5060;priority=10;weight=10"); .... request_route { ... janssonrpc_request("user_db", "core.version", '', "route=RESPONSE;retry=1"); ... } .... event_route[xhttp:request] { if ($hu =~ "^/jsonrpc") { jsonrpc_dispatch(); } else { xhttp_reply("200", "OK", "text/html", "<html><body>Wrong URL $hu</body></html>"); } return; } With this config i got error Sep 19 08:52:15 sipchel /usr/sbin/kamailio[39891]: INFO: janssonrpcc [janssonrpc_connect.c:294]: bev_connect(): Connecting to server 127.0.0.1:5060 for conn user_db. Sep 19 08:52:15 sipchel /usr/sbin/kamailio[39891]: INFO: janssonrpcc [janssonrpc_connect.c:277]: bev_connect_cb(): Connected to host 127.0.0.1:5060 ... Sep 19 08:52:20 sipchel /usr/sbin/kamailio[39891]: WARNING: janssonrpcc [janssonrpc_request.c:242]: schedule_retry(): Number of retries exceeded. Failing request. Sep 19 08:52:20 sipchel /usr/sbin/kamailio[39891]: WARNING: janssonrpcc [janssonrpc_io.c:218]: fail_request(): Request timeout: #012{"id":3,"jsonrpc":"2.0","method":"core.version"} But [root@sipchel kamailio]# curl -s -H 'Content-Type: application/json' --data-binary '{"jsonrpc":"2.0", "method": "core.version"}' http://localhost:5060/jsonrpc { "jsonrpc": "2.0", "result": "kamailio 5.0.3 (x86_64/linux) " } How to get working rpc client in kamailio? Is it possible to define url for janssonrpcc module? -- Aydar A. Kamalov

1 0

rtpproxy sdp
by Isravel Raja Thangamani 18 Sep '17

18 Sep '17

Hi Thanks in advance if anyone can point me in the correct direction . I have kamailio running in front of some asterisk. SIP endpoint register to their asterisk PBX via Kamailio dispatcher module. I'm running rtpengine with a Wan and private interface to bridge audio stream between these endpoints on the WAN to asterisk PBX running on LAN IP behind Kamailio. Calls from ext to ext work fine audio both directions , calls outbound to PSTN via SIP trunk to SIP provider via trunk on asterisk work fine audio both directions. But incoming calls via SIP provider no audio from external caller to the asterisk ext neither asterisk to external caller I reckon I have something wrong in my Kamailio.cfg . if I register an ext direct to asterisk I get audio both ways on incoming calls. And rtp logs I think my mistake in somewhere in the cfg below. Do I need to handle invites from the backend asterisk servers different that invites from sip endpoints? This e-mail message and any documents attached to it are confidential and may contain information that is protected from disclosure by various federal and state laws, including the HIPAA privacy rule (45 C.F.R., Part 164). This information is intended to be used solely by the entity or individual to whom this message is addressed. If you are not the intended recipient, be advised that any use, dissemination, forwarding, printing, or copying of this message without the sender's written permission is strictly prohibited and may be unlawful. Accordingly, if you have received this message in error, please notify the sender immediately by return e-mail or call (insert phone number), and then delete this message.

3 2

ringing calls dialogs
by Aidar Kamalov 18 Sep '17

18 Sep '17

Hello, is it possible from route get info about all calls in ringing state? (for call pickup implementation) Is it better to do via dialogs or presence modules? -- Aydar A. Kamalov

3 4

Issue in ACC_CDR
by Logeshwaran G 18 Sep '17

18 Sep '17

Hi, I have enabled both cnxcc and ACC cdr flags. The call is dropped when the credit is got over (its handling by cnxcc correctly.) In this case the details are not storing in the ACC_cdr table. I am not sure what I am missing. in cfg: if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails Error: 2(19484) ERROR: acc [acc_cdr.c:373]: write_cdr(): message is empty! 2(19484) ERROR: acc [acc_cdr.c:656]: cdr_on_end_confirmed(): failed to write cdr! Please Guid me. Thanks & Kind Regards, *Logeshwaran G* *"Be Hungry, Stay Foolish"*

2 1

Siremis - version from git don't work
by Yu Boot 18 Sep '17

18 Sep '17

Downloaded new Siremis version via "git clone" and replaced /var/www/ content by new files. Now any menu entry gives the following error: https://pastebin.com/hV8ngRVx If if copy old Siremis files back, everything works fine.

2 8

Polycom Presence Routing for PUBLISH Messages
by John Janke 18 Sep '17

18 Sep '17

So I'm using a Polycom VVX 310 with Presence enabled (both on Phone and Kamailio) and the phone is sending SUBSCRIBE messages but not PUBLISH. I have tried various phone configurations without success. The closest thread to my situation I found online said this: "The thing is Polycom phones perform end-to-end subscription to presence event. It worked when I changed the routing configuration to simply relay the SUBSCRIBE request to the monitored device. (I'm using OpenSER as SIP proxy.)" I'm not sure if this is correct as it's from 2012, but in the case that it is I don't quite understand how to implement this. Any help on how to edit the config file? John Janke Progressive Innovation and Automation, Inc. PO BOX 309 Plainfield, IL 60544 Direct: 847.316.9028 Office: 800.688.9051 x 55328 Fax: 630.860.1017

2 1

Re: [SR-Users] Siremis graph time format is a bug!?
by Humberto Rodríguez 18 Sep '17

18 Sep '17

Thanks Daniel-Constantin Mierla, I followed your recommendation, and began to use a date column as INT. Then, I started to insert the date as unix_timestamp, like this (unix_timestamp(now())), which was the only way the graphic showed the date correctly (Active_Failed_Dialogs_4_hours - From 2017-09-15 12:04:01 To 2017-09-15 16:00:01), I failed inserting (now()+0), siremis didn't understand this format too. Cheers, Hto Now No Matter How Hard The Past, You Can Always Begin Again ~ -------------------------------------------- On Thu, 9/14/17, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: Subject: Re: [SR-Users] Siremis graph time format is a bug!? To: "Humberto Rodríguez" <humberj77(a)yahoo.es>, "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org>, sr-users(a)lists.sip-router.org Date: Thursday, September 14, 2017, 4:09 AM Hello, try to set the type of the time column to int, so it is retrieved as numeric timestam, not as formated datetime. Cheers, Daniel On 13.09.17 05:39, Humberto Rodríguez wrote: Hello: I added a graphic to Siremis of my actives dialogs. But, it only shows unix time (From 1969-12-31 19:33:37 To1969-12-31 19:33:37) as Xaxis (19:33). I had siremis 4.2 and never saw it correctly, I just updated to Siremis 4.4 and I see the same problem.Please see the attachment file with the graph, and my configuration files and table bellow Thanks in advance. My /var/www/siremis-4.4.0/siremis/modules/ser/view/xxxx.xml file <?xml version="1.0" standalone="no"?> <EasyView Name="ChartsSumdialogsView" Description="Charts" class="EasyView" Tab="" TemplateEngine="Smarty" Access="SERAdmin.Administer_Modules" TemplateFile="view.tpl"> <FormReferences> <Reference Name="ser.cms.form.ChartsForm"/> </FormReferences> </EasyView> ----------------------------------------------------------- My siremisCharts.xml ---------------------------------------------------------- <ChartGroup name="sumdialogs"> <Chart name="sumdialogs" table="active_dialogs" title="Active_Failed_Dialogs" type="area" order="reverse" orderby="ORDER BY id DESC" limit="LIMIT 10080"> <XAxis>  <Item name="date_created" data="date_created" title="date_created" type="timestamp"/> </XAxis> <YAxis> <Item name="active_dialogs" data="active_dialogs" title="active_dialogs" type="dataset" color="#D02020"/> <Item name="failed_dialogs" data="failed_dialogs" title="failed_dialogs" type="dataset" color="#DFC329"/> </YAxis> </Chart> </ChartGroup> --------------------------------------------------------------------------------------- See bellow, the portion of the table used to build the graphic, the column "date_created" is the problem, and the format is "TIMESTAMP" ------------------------------------------------------------------------------------------ # id, date_created, active_dialogs, early_dialogs, expired_dialogs, failed_dialogs, processed_dialogs, '1', '2017-04-11 14:40:40', '220', '0', '1219', '37', '6879491' '2', '2017-04-11 14:42:02', '206', '0', '1219', '37', '6879604' '3', '2017-04-11 14:43:01', '215', '0', '1219', '37', '6879685' '4', '2017-04-11 14:44:01', '191', '0', '1220', '37', '6879744' '5', '2017-04-11 14:45:02', '203', '0', '1220', '37', '6879837' '9', '2017-04-11 14:46:01', '211', '0', '1220', '37', '6879929' '11', '2017-04-11 14:47:01', '234', '0', '1220', '37', '6880019' '12', '2017-04-11 14:48:01', '222', '0', '1220', '37', '6880087' ---------------------------------------------------------------------------------------------------------------------------------- No Matter How Hard The Past, You Can Always Begin Again ~ _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - www.kamailioworld.com

2 1

Kamailio Offline Messages
by Jango 17 Sep '17

17 Sep '17

I need help about offline messages this is my full working kamailio.cfg with audio/video call & instant messaging but i need features to receive user offline messages. kamailio.cfg #!KAMAILIO #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_PRESENCE #!define WITH_NAT #!define WITH_TLS #!define WITH_ACCDB # # Kamailio (OpenSER) SIP Server v4.2 - default configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: <sr-users(a)lists.sip-router.org> # # Refer to the Core CookBook at http://www.kamailio.org/wiki/ # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # - option for NAT SIP OPTIONS keepalives: WITH_NATSIPPING # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Include Local Config If Exists ######### import_file "kamailio-local.cfg" ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!endif #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ #auto_aliases=no /* add local domain aliases */ alias="jsrver.tk" /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:10.0.0.10:5060 /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 #!ifdef WITH_TLS enable_tls=yes #!endif # life time of TCP connection when there is no traffic # - a bit higher than registration expires to cope with UA behind NAT tcp_connection_lifetime=3605 ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" pstn.gw_port = "" desc "PSTN GW Port" #!endif #!ifdef WITH_VOICEMAIL # VoiceMail Routing on offline, busy or no answer # # - by default Voicemail server IP is empty to avoid misrouting voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules/" #!else mpath="/usr/lib64/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "mi_fifo.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 "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "mi_rpc.so" loadmodule "acc.so" # -- for siremis CDRs -------------- loadmodule "rtimer.so" loadmodule "sqlops.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 0) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # set it to 1 to enable GRUU modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "//etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) modparam("debugger", "log_level_name", "exec") #!endif modparam("rtimer", "timer", "name=cdr;interval=300;mode=1;") modparam("rtimer", "exec", "timer=cdr;route=CDRS") modparam("sqlops", "sqlcon", "cb=>mysql://kamailio:kamailiorw@localhost/kamailio") ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # handle retransmissions if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } if($ua =~ "friendly-scanner") { sl_send_reply("200", "OK"); exit; } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS") && uri==myself && $rU==$null) { sl_send_reply("200","Keepalive"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (!has_totag()) return; # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); exit; } # Handle SIP registrations route[REGISTRAR] { if (!is_method("REGISTER")) return; if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); #!ifdef WITH_NATSIPPING # do SIP NAT pinging setbflag(FLB_NATSIPPING); #!endif } if (!save("location")) sl_reply_error(); exit; } # User location service route[LOCATION] { #!ifdef WITH_SPEEDDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") if(sd_lookup("speed_dial")) route(SIPOUT); #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) route(SIPOUT); #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } route(RELAY); exit; } # Presence server processing route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") { route(TOVOICEMAIL); # returns here if no voicemail server is configured sl_send_reply("404", "No voicemail service"); exit; } #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; } if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if(is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # IP authorization and user authentication route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { if(is_first_hop()) set_contact_alias(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control and singaling updates for NAT traversal route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage("co"); if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (uri==myself) return; append_hf("P-hint: outbound\r\n"); route(RELAY); exit; } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } if (strempty($sel(cfg_get.pstn.gw_port))) { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); } else { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # Routing to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE|SUBSCRIBE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n"); return; } if(is_method("INVITE")) { if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } else { if($rU==$null) return; $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } route(RELAY); exit; #!endif return; } # ====================================================== # Populate CDRs Table of Siremis # ====================================================== route[CDRS] { sql_query("cb","call kamailio_cdrs()","rb"); sql_query("cb","call kamailio_rating('default')","rb"); } # Manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # Manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); } # Manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { $du = $null; route(TOVOICEMAIL); exit; } #!endif }

2 1

Dispatcher Module with ds_is_from_list function
by Ilie Soltanici 17 Sep '17

17 Sep '17

Hello, "ds_is_from_list" function from Dispatcher module give false result - when actually it should be true. modparam("dispatcher", "db_url",DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "setid_col", "setid") modparam("dispatcher", "destination_col", "destination") modparam("dispatcher", "flags", 2) modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") modparam("dispatcher", "sock_avp", "$avp(AVP_SOCK)") request_route { xlog("L_INFO","[=Start Routing=]") if(ds_is_from_list()) { xlog("L_INFO","[=FROM ASTERISK=]"); } else { xlog("L_INFO","[=TO ASTERISK=]"); } if($si=="192.168.1.10") { xlog("L_INFO","[=FROM ASTERISK BY IP=]"); } else { xlog("L_INFO","[=TO ASTERISK BY IP=]"); } root@kamailio:/var/log# kamctl dispatcher show dispatcher gateways +----+-------+---------------------+-------+----------+-------+-----------------+ | id | setid | destination | flags | priority | attrs | description | +----+-------+---------------------+-------+----------+-------+-----------------+ | 1 | 1 | sip:192.168.1.10:5060 | 0 | 3 | | Main Asterisk | | 2 | 2 | sip:192.168.1.11:5060 | 0 | 2 | | Backup Asterisk | | 3 | 3 | sip:192.168.1.12:5060 | 0 | 1 | | Test Asterisk | +----+-------+---------------------+-------+----------+-------+-----------------+ root@kamailio:/var/log# cat /var/log/kamailio.log [=Start Routing=] [=TO ASTERISK=] [=FROM ASTERISK BY IP=] Additional Information - SIP Packet 2017/09/15 19:21:57.589451 192.168.1.50:5061 -> 192.168.1.1:5060 REGISTER sip:192.168.1.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.50:5061;branch=z9hG4bK-5219-1-0 From: sip:10@192.168.1.50;tag=1 To: sip:10@192.168.1.50 Call-ID: 1-5219(a)192.168.1.50 CSeq: 1 REGISTER Contact: sip:10@192.168.1.50:5061 Max-Forwards: 10 Expires: 120 User-Agent: SIPp/Win32 Content-Length: 0 - Kamailio Version - output of kamailio -v root@kamailio:/var/log# kamailio -v version: kamailio 5.0.3 (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 12:12:14 Sep 7 2017 with gcc 6.3.0 root@kamailio:/var/log# - Operating System: Debian 9.1/ Linux kamailio1 4.9.0-3-amd64 #1 SMP Debian 4.9.30-2+deb9u3 (2017-08-06) x86_64 GNU/Linux Is this a bug or something wrong with configuration? Thank you

2 1

Implementing C or Java file in kamailio
by Logeshwaran G 15 Sep '17

15 Sep '17

Hi, Is there any way to implement my own C or Java program in kamailio.cfg to Validate my own scenarios? Thanks & Kind Regards, *Logeshwaran G* *"Be Hungry, Stay Foolish"*

3 2

Replacing the From header user part.
by vivek＠advaitamtech.com 15 Sep '17

15 Sep '17

Hi All, Am trying to replace the user part of From header with own user name in kamailio.cfg I tried to assign directly, Eg: $fU = "abcd" Sometime when I receive From header as ' "12345" <12345(a)1.1.1.1>;tag=xxxx '. In this case it works as ' "12345" <abcd(a)1.1.1.1>;tag=xxxx ' . So i tried using replace_all() function. But this ended up adding extra characters(in my case "abcd") infront of content_type header. I dont know why. Can someone please help me out in replacing the from header in this case? Thanks, Vivek

2 2

Maximum Registration on Kamailo??
by amit Kumar 14 Sep '17

14 Sep '17

Hi All, kamctl ul show => show below result, does that mean max 1024 contact can be registered, if that is the case how to increase the max registration limit. Domain:: location table=*1024 *records=2 max_slot=1 Thanks

3 2

kamailio - multiple asterisk and chanspy
by Жан Базаров 14 Sep '17

14 Sep '17

Hello! how i can use asterisk's chanspy_app in kamailio-loadbalance-asterisk system?

2 1

Killing Calls for a Profile
by Sunil More 14 Sep '17

14 Sep '17

Hello All, Is there a method / command to kill calls related to one dialog profile. Right now i have to find all dialogs for a customer and loop through all the dialog hashes to run kamcmd dlg.end_dlg till i can kill all calls. Is there a easier way to do this. Thanking You, Sunil More Ph : 9503338275

2 4

Siremis "Chat Service" time format has bug or is only config issue??
by Humberto Rodríguez 14 Sep '17

14 Sep '17

Hello: I added a graphic to Siremis of my actives dialogs. But, it only shows unix time (From 1969-12-31 19:33:37 To1969-12-31 19:33:37) as Xaxis (19:33). I had version 4.2 and never saw it correctly, I just updated to version 4.4 and I see the same problem.Please see the attachment file with the graph, and my configuration files and table bellow Thanks in advance. My /var/www/siremis-4.4.0/siremis/modules/ser/view/xxxx.xml file ?xml version="1.0" standalone="no"?> <EasyView Name="ChartsSumdialogsView" Description="Charts" class="EasyView" Tab="" TemplateEngine="Smarty" Access="SERAdmin.Administer_Modules" TemplateFile="view.tpl"> <FormReferences> <Reference Name="ser.cms.form.ChartsForm"/> </FormReferences> </EasyView> ----------------------------------------------------------- My siremisCharts.xml ---------------------------------------------------------- > <ChartGroup name="sumdialogs"> > > <Chart name="sumdialogs" table="active_dialogs" > title="Active_Failed_Dialogs" type="area" order="reverse" > > > orderby="ORDER BY id DESC" limit="LIMIT 10080"> > > <XAxis> >  > > <Item > name="date_created" data="date_created" title="date_created" > type="timestamp"/> > > </XAxis> > > <YAxis> > > <Item > name="active_dialogs" data="active_dialogs" > title="active_dialogs" type="dataset" color="#D02020"/> > > > <Item > name="failed_dialogs" data="failed_dialogs" > title="failed_dialogs" type="dataset" color="#DFC329"/> > > </YAxis> > </Chart> > </ChartGroup> > --------------------------------------------------------------------------------------- See bellow, the portion of the table used to build the graphic, the column "date_created" is the problem, and the format is "TIMESTAMP" > ------------------------------------------------------------------------------------------ > > > # id, date_created, active_dialogs, > early_dialogs, expired_dialogs, failed_dialogs, > processed_dialogs, > '1', '2017-04-11 14:40:40', '220', '0', > '1219', '37', '6879491' > '2', '2017-04-11 14:42:02', '206', '0', > '1219', '37', '6879604' > '3', '2017-04-11 14:43:01', '215', '0', > '1219', '37', '6879685' > '4', '2017-04-11 14:44:01', '191', '0', > '1220', '37', '6879744' > '5', '2017-04-11 14:45:02', '203', '0', > '1220', '37', '6879837' > '9', '2017-04-11 14:46:01', '211', '0', > '1220', '37', '6879929' > '11', '2017-04-11 14:47:01', '234', > '0', '1220', '37', '6880019' > '12', '2017-04-11 14:48:01', '222', > '0', '1220', '37', '6880087' > > ----------------------------------------------------------------------------------------------------------------------------------

2 1

Siremis graph time format is a bug!?
by Humberto Rodríguez 14 Sep '17

14 Sep '17

Hello: I added a graphic to Siremis of my actives dialogs. But, it only shows unix time (From 1969-12-31 19:33:37 To1969-12-31 19:33:37) as Xaxis (19:33). I had siremis 4.2 and never saw it correctly, I just updated to Siremis 4.4 and I see the same problem.Please see the attachment file with the graph, and my configuration files and table bellow Thanks in advance. My /var/www/siremis-4.4.0/siremis/modules/ser/view/xxxx.xml file <?xml version="1.0" standalone="no"?> <EasyView Name="ChartsSumdialogsView" Description="Charts" class="EasyView" Tab="" TemplateEngine="Smarty" Access="SERAdmin.Administer_Modules" TemplateFile="view.tpl"> <FormReferences> <Reference Name="ser.cms.form.ChartsForm"/> </FormReferences> </EasyView> ----------------------------------------------------------- My siremisCharts.xml ---------------------------------------------------------- <ChartGroup name="sumdialogs"> <Chart name="sumdialogs" table="active_dialogs" title="Active_Failed_Dialogs" type="area" order="reverse" orderby="ORDER BY id DESC" limit="LIMIT 10080"> <XAxis>  <Item name="date_created" data="date_created" title="date_created" type="timestamp"/> </XAxis> <YAxis> <Item name="active_dialogs" data="active_dialogs" title="active_dialogs" type="dataset" color="#D02020"/> <Item name="failed_dialogs" data="failed_dialogs" title="failed_dialogs" type="dataset" color="#DFC329"/> </YAxis> </Chart> </ChartGroup> --------------------------------------------------------------------------------------- See bellow, the portion of the table used to build the graphic, the column "date_created" is the problem, and the format is "TIMESTAMP" ------------------------------------------------------------------------------------------ # id, date_created, active_dialogs, early_dialogs, expired_dialogs, failed_dialogs, processed_dialogs, '1', '2017-04-11 14:40:40', '220', '0', '1219', '37', '6879491' '2', '2017-04-11 14:42:02', '206', '0', '1219', '37', '6879604' '3', '2017-04-11 14:43:01', '215', '0', '1219', '37', '6879685' '4', '2017-04-11 14:44:01', '191', '0', '1220', '37', '6879744' '5', '2017-04-11 14:45:02', '203', '0', '1220', '37', '6879837' '9', '2017-04-11 14:46:01', '211', '0', '1220', '37', '6879929' '11', '2017-04-11 14:47:01', '234', '0', '1220', '37', '6880019' '12', '2017-04-11 14:48:01', '222', '0', '1220', '37', '6880087' ---------------------------------------------------------------------------------------------------------------------------------- No Matter How Hard The Past, You Can Always Begin Again ~

2 1

Cannot msg or call
by Костик 14 Sep '17

14 Sep '17

Hello, I have setup Kamailio 5.0.3 under Ubuntu 16.04. I can connect to the server but cannot call/text my contacts and don't see them online. When I try to call them, I get errors like ERROR: <core> [core/tcp_main.c:4264]: tcpconn_main_timeout(): connect 10.8.8.218:51079 failed (timeout) For texts, the error is 405 Method not allowed What is the problem and how to fix it? Thanks Serj

2 2

kamailio 4.4 registering to a SIP trunk provider using UAC then outbound routing to it.
by Jonathan Hunter 13 Sep '17

13 Sep '17

Hi Guys, I have a SIP provider who only accept registrations from users, so I have registered my kamailio Server with them using the uac module. It is registered fine, and I can get inbound calls on the trunk. My query is the best way to send outbound calls out via the registered sip gateway? I would normally load balance using dispatcher if it was to IP authenticating providers, but as they are using registration, what would be best practice to send outbound calls via the registration which is establised with the uac module? Is it just a question of doing it within the configuration file, as again they will look to auth requests as well with a 401/407 so I assume I can use uac_auth() for that. Just wondering what other people have done if anything, is my other option is to register using a B2Bua on the same server, and leave kamailio just as a proxy. Many thanks Jon

2 2

dialog keep-alive don't work
by Yu Boot 13 Sep '17

13 Sep '17

Hello. I'm using following dialog module settings: modparam("dialog", "db_url", DBURL) modparam("dialog", "db_mode", 1) modparam("dialog", "dlg_flag", 4) modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "profiles_with_value", "limithash ; caller") modparam("dialog", "ka_timer", 10) modparam("dialog", "ka_interval", 300) modparam("dialog", "send_bye", 1) and I expect some OPTIONS within dialogs. But that don't happen. I don't see any OPTIONS messages in sngrep in established calls. What's wrong? I call dlg_manage() right before main t_relay, everything including CDRs are working fine...

2 3

Issue in Acc module
by Logeshwaran G 13 Sep '17

13 Sep '17

Hi, I am getting the below Error while Call Ended. 2(28504) ERROR: acc [acc_cdr.c:373]: write_cdr(): message is empty! 2(28504) ERROR: acc [acc_cdr.c:656]: cdr_on_end_confirmed(): failed to write cdr! Am I missing something? Please Guide me Thanks & Kind Regards, *Logeshwaran G* *"Be Hungry, Stay Foolish"*

1 1

kill the call using callid
by Logeshwaran G 13 Sep '17

13 Sep '17

Hi, Is it possible to kill the call using callid, I have referred the dialog module, I found the command: kamcmd dlg.end_dlg 342 56 But I need to kill the call with Callid, Input Please? Thanks & Kind Regards, *Logeshwaran G* *"Be Hungry, Stay Foolish"*

2 3

Can not register on kamailio
by Adesh Pandey 13 Sep '17

13 Sep '17

I am new to kamailio and I want to setup Kamailio as dispatcher for two Asterisk boxes but SIP registrations must be happening on the Kamailio itself. I have enabled mysql module and used auth module but when I try to register it says unauthorized access. In subscribers list I have added an entry for sip user 1 with domain as kamailio IP. Please help me to understand where can I improve so that I can implement register on Kamailio. -- Adesh Pandey Sr. Software Developer [image: phone] +91 92129 92129 Ext: 56 [image: moible] +91 8527384897 [image: Profile Pic] [image: Facebook] <https://www.facebook.com/VoiceTree?_rdr>[image: Twitter] <https://twitter.com/MyOperator>[image: Twitter] <https://www.youtube.com/user/MyOperatorCo>[image: Linkedin] <https://www.linkedin.com/company/voicetree-technologies>[image: Google+] <https://plus.google.com/+MyoperatorCo/posts> [image: Logo] <http://myoperator.co/>

2 4

Problem implementing usrloc via dmq
by Aleksandar Sosic 13 Sep '17

13 Sep '17

Hi everyone, we're trying to make work usrloc over dmq without a db on our nodes. Our `kamailio.cfg` code to make it work on the nodes is like this: ``` ... loadmodule "usrloc.so" ... loadmodule "dmq.so" loadmodule "dmq_usrloc.so" ... modparam("dmq", "server_address", DMQ_ADDRESS) modparam("dmq", "notification_address", "sip:routers.ourcloud.com:5060") modparam("dmq", "multi_notify", 1) modparam("dmq", "num_workers", 4) modparam("dmq_usrloc", "enable", 1) modparam("dmq_usrloc", "sync", 1) ``` Where `DMQ_ADDRESS="sip:<some_address>:5060"` is the public IP address of the running node. `nslookup` of `routers.ourcloud.com`: ```➜ ~ nslookup routers.ourcloud.com Server: 8.8.8.8 Address: 8.8.8.8#53 Non-authoritative answer: Name: rrouters.ourcloud.com Address: <IP1> Name: routers.ourcloud.com Address: <IP2> ``` and obviously IP1 and IP2 are pubblic IPs of kamailio nodes that respond to my pings. When i register a client to node IP1 it's ul dump is: ``` { "jsonrpc": "2.0", "result": { "Domain": "location", "Size": 1024, "AoRs": [{ "Info": { "AoR": "0409828030", "HashID": 1970914268, "Contacts": [{ "Contact": { "Address": "sip:0409828030@172.19.0.255:56328;ob", "Expires": 289, "Q": -1, "Call-ID": "5mclwiX4oYAVK0z-iT7M0p9RBX5-02g6", "CSeq": 59427, "User-Agent": "Telephone 1.2.6", "Received": "sip:<PROXY_IP>:5060", "Path": "[not set]", "State": "CS_NEW", "Flags": 0, "CFlags": 192, "Socket": "udp:<IP1>:5060", "Methods": 8159, "Ruid": "uloc-599d8fa8-f-1", "Instance": "[not set]", "Reg-Id": 0, "Last-Keepalive": 1503498176, "Last-Modified": 1503498176 } }] } }], "Stats": { "Records": 1, "Max-Slots": 1 } }, "id": 1 } ``` On node with IP2 the ul dump is: ``` { "jsonrpc": "2.0", "result": { "Domain": "location", "Size": 1024, "AoRs": [], "Stats": { "Records": 0, "Max-Slots": 0 } }, "id": 1 } ``` What are we doing wrong? Are we missing something? -- Aleksandar Sosic mail: alex.sosic(a)timenet.it skype: alex.sosic cell: +385 91 2505 146

3 7

Help to understand kamailio error msgs
by José Seabra 12 Sep '17

12 Sep '17

Hello there, I'm facing some error msg on my kamailio server in production that i would like to understand what they means in order to avoid them. The msg are: - *ERROR: tm [t_reply.c:1270]: t_should_relay_response(): ERROR: t_should_relay_response: status rewrite by UAS: stored: 408, received: 200* - *dispatcher [dispatch.c:1509]: ds_load_replace(): cannot find load for (16720.mydomain.com <http://16720.mydomain.com>)* - *dispatcher [dispatch.c:2270]: ds_next_dst(): cannot update load distribution* - *tm [t_reply.c:533]: _reply_light(): ERROR: _reply_light: can't generate 487 reply when a final 603 was sent out* - *tm [t_reply.c:533]: _reply_light(): ERROR: _reply_light: can't generate 487 reply when a final 480 was sent out* - *WARNING: tm [t_lookup.c:245]: ack_matching(): WARNING: ack_matching() attempted on a transaction with no E2EACK callbacks => the results are not completely reliable when forking is involved* Thank you Best regards -- José Seabra

2 3

jsonrpc execution error
by David Villasmil 12 Sep '17

12 Sep '17

Hello guys, i configured jsonrpc on kamailio.cfg and RPCFIFOPATH on kamctlrc to the same file, seems to work for some commands, but not for others: root@ip-172-31-30-160:/home/admin# kamctl trusted show +----+--------------+-------+--------------+--------------+- -----+----------+ | id | src_ip | proto | from_pattern | ruri_pattern | tag | priority | +----+--------------+-------+--------------+--------------+- -----+----------+ | 1 | 1.2.3.4 | udp | | NULL | | 0 | | 2 | 5.4.3.2 | udp | | NULL | | 0 | +----+--------------+-------+--------------+--------------+- -----+----------+ root@ip-172-31-30-160:/home/admin# kamctl trusted dump { "jsonrpc": "2.0", "error": { "code": -32000, "message": "Execution Error" }, "id": 32022 } root@ip-172-31-30-160:/home/admin# kamctl trusted reload { "jsonrpc": "2.0", "error": { "code": -32000, "message": "Execution Error" }, "id": 32041 } Any help? Thanks! Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

2 1

Multiple Htable Reload
by Logeshwaran G 12 Sep '17

12 Sep '17

Hi, I have multiple Htables, Is it possible to Reload all the Htables at one time via jsonrpc: As of Now I am using the below command to Reload one htable at a time. {"jsonrpc": "2.0","method": "htable.reload","params": ["b"],"id": 123} Input Please? Thanks & Kind Regards, *Logeshwaran G* *"Be Hungry, Stay Foolish"*

2 1

Map Technique(Key value pair) in kamailio
by Logeshwaran G 12 Sep '17

12 Sep '17

Hi, Is there any way to use the Mapping technique in kamailio. For Example: ========== Key = 91, Value = India Key = 92, Value = Srilanka How to store Key Value Pairs into kamailio and how to Retrieve? If I pass 91 it should Retrieve India, Is it possible? Input Please Thanks & Kind Regards, *Logeshwaran G* *"Be Hungry, Stay Foolish"*

3 8

Passive Open TCP sessions increases and kamailio runs out of memory
by Niklas Larsson 11 Sep '17

11 Sep '17

Hi, I have a kamailio 4.4.5 (x86_64/linux), running as a proxy in front of asterisk, handling udp/tcp/tls registration, using udp towards asterisk. Over time memory is consumed and the Passive Open TCP sessions is increasing until we run out of memory and a restart is needed. Sep 11 06:25:03 ERROR: <core> [mem/f_malloc.c:415]: fm_search_defrag(): fm_search_defrag(0x7f9d55184000, 34120); Free fragment not found! Sep 11 06:25:03 ERROR: <core> [mem/f_malloc.c:498]: fm_malloc(): fm_malloc(0x7f9d55184000, 34120) called from tls: tls_init.c: ser_malloc(275), module: tls; Free fragment not found! This image contains graphs that show the behaviour: https://imgur.com/68PvmyA Should it behave like this? Is there a knob to twist to change behaviour? tcp / tls stuff from kamailio.cfg: enable_tls=yes tcp_max_connections=8196 tls_max_connections=8196 tcp_no_connect=yes # Don't try to connect - drop if there is no session tcp_connection_lifetime=3605 modparam("tcpops", "closed_event", 1) modparam("usrloc", "handle_lost_tcp", 1) modparam("usrloc", "close_expired_tcp", 1) in the registrar route I have: tcp_keepalive_enable("60", "5", "5"); /niklas

1 0

kamailio asterisk queue
by Жан Базаров 11 Sep '17

11 Sep '17

hello! how i can forward presence peer state to asterisk queue? Thank's

2 1

Q: purpose of t_newtran() in kamailio-basic.cfg route[LOCATION]?
by Anthony Alba 09 Sep '17

09 Sep '17

In kamailio-basic.cfg route[LOCATION] we have the following: --snip-- $var(rc) = $rc; t_newtran(); switch ($var(rc)) { case -1: case -3: --snip-- What is the purpose of t_newtran() here? Since we have not hit t_relay() yet (and don't have a transaction) is this to ensure the caller sees/acks the 404/405? Earlier in the mailing list https://lists.kamailio.org/pipermail/sr-users/2015-April/088062.html I read, although this is in reply to a different situation: "Why do you need to create new transactions yourself with t_newtran()? Nonexotic applications of Kamailio usually don't require this. ..." # User location service route[LOCATION] { if (!lookup("location")) { $var(rc) = $rc; t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } route(RELAY); exit; } Thanks AAlba

2 1

Re: [SR-Users] append_branch + dispatcher
by Aidar Kamalov 08 Sep '17

08 Sep '17

After I add route(DISPATCH) to branch_route branch calls are going throught asterisk # Manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); xnotice("BRANCH: $rm $ru [$ci]"); route(DISPATCH); route(NATMANAGE); } but if I set $ru = "sip:34200@domain.org"; append_branch("sip:34201@domain.org"); first main ($ru) INVITE resolve to contact of ip phone INVITE sip:34200@172.16.3.7:5060 but branch doesnt' resolve INVITE sip:34201@domain.org Is there any way to resolve it? -- Aydar A. Kamalov

2 1

append_branch + dispatcher
by Aidar Kamalov 08 Sep '17

08 Sep '17

Hello! I have a simple route route[HUNT] { $ru = "sip:34200@172.16.3.7:5060"; append_branch("sip:34201@192.168.2.71"); } 172.16.3.7 and 192.168.2.71 ip phones 192.168.10.54 is kamailio 192.168.10.47 is asterisk If I called to this route kamailio will call to 34200 through dispatcher via asterisk. But branch going directly to the ip phone. Is it possible to call all branches through dispatcher and asterisks? U 192.168.10.54:5060 -> 192.168.10.47:5060 INVITE sip:34200@172.16.3.7:5060 SIP/2.0. Record-Route: <sip:192.168.10.54;lr=on;ftag=1dc09a643aa66d85o0;did=12f.336>. Via: SIP/2.0/UDP 192.168.10.54;branch=z9hG4bK2fd3.b2bbf3cfc7962b4cb7a1c6a174668c0d.0. Via: SIP/2.0/UDP 172.16.3.4:5060;branch=z9hG4bK-8b2e7bed. From: ".............. .......... ................" <sip:53003@192.168.10.54 >;tag=1dc09a643aa66d85o0. To: <sip:1111@192.168.10.54>. Call-ID: 7c5b5710-57b4f8c9(a)172.16.3.4. U 192.168.10.54:5060 -> 192.168.2.71:5060 INVITE sip:34201@192.168.2.71 SIP/2.0. Record-Route: <sip:192.168.10.54;lr=on;ftag=1dc09a643aa66d85o0;did=12f.336>. Via: SIP/2.0/UDP 192.168.10.54;branch=z9hG4bK2fd3.b2bbf3cfc7962b4cb7a1c6a174668c0d.1. Via: SIP/2.0/UDP 172.16.3.4:5060;branch=z9hG4bK-8b2e7bed. From: ".............. .......... ................" <sip:53003@192.168.10.54 >;tag=1dc09a643aa66d85o0. To: <sip:1111@192.168.10.54>. Call-ID: 7c5b5710-57b4f8c9(a)172.16.3.4. -- Aydar A. Kamalov

2 1

need some help about acc module
by Vidja Prashant 08 Sep '17

08 Sep '17

Hello, I have some feature required changes in acc module. In acc module we getting cdrs in database only which have METHOD=INVITE METHOD=BYE but in my setup I also get METHOD=OPTIONS so I need this entry also in acc table in kamailio database. so which changes i need to do in kamailio acc module.... or other...... my cdrs log I paste below which have mehod=options "Sep 7 07:16:53 sip /usr/local/sbin/kamailio[29264]: NOTICE: acc [acc.c:317]: acc_log_request(): ACC: request accounted: timestamp=1504783013;method=OPTIONS;from_tag=eUC9UgGrD;to_tag=j90H6yy5pU6Xc;call_id=2xJd0KCt-P;code=200;reason=OK;calling=46566464645;called=you;egressIP= kamailio.org;src_ip=1.0.0.127;to=sip:+918866213320@sip.call.org ;tag=j90H6yy5pU6Xc;from=sip:918888888888@localhost;contact=;pai=;uri=sip: you(a)kamailio.org;onnet=0;rate=0.000333333" Thanks.. # #

2 1

crash after using async_route()
by Vitaliy Aleksandrov 07 Sep '17

07 Sep '17

Hello kamailio list, Recently found a problem in my configuration that uses async_route() functionality. It crashes after several calls when wait_timer fires. #0 0xb74a8556 in raise () from /lib/libc.so.6 #1 0xb74a9d78 in abort () from /lib/libc.so.6 #2 0x08293ae2 in qm_free (qmp=0xad65d000, p=0x3d64692d, file=0xb6216a16 "tm: h_table.c", func=0xb621663c <__FUNCTION__.18751> "free_cell_helper", line=187, mname=0xb621664d "tm") at core/mem/q_malloc.c:471 #3 0xb613f103 in free_cell_helper (dead_cell=0xae2cd210, silent=0, fname=0xb6239ea5 "timer.c", fline=655) at h_table.c:187 #4 0xb61e7758 in wait_handler (ti=557858937, wait_tl=0xae2cd258, data=0xae2cd210) at timer.c:655 #5 0x0826a2cc in timer_list_expire (t=557858937, h=0xad6b9668, slow_l=0xad6ba144, slow_mark=312) at core/timer.c:874 #6 0x08267cb1 in timer_handler () at core/timer.c:939 #7 0x0826a4d3 in timer_main () at core/timer.c:978 #8 0x08069575 in main_loop () at main.c:1721 #9 0x080707ca in main (argc=11, argv=0xbf85f044) at main.c:2723 When crash happens, kamailio prints the following message: Sep 4 16:15:38 [18938]: : <core> [core/mem/q_malloc.c:469]: qm_free(): BUG: qm_free: bad pointer 0x70707553 (out of memory block!) called from tm: h_table.c: free_cell_helper(187) - aborting Also had a few crashes in retransmission_handler(): #0 0xb750b556 in raise () from /lib/libc.so.6 #1 0xb750cd78 in abort () from /lib/libc.so.6 #2 0xb6249b5a in retransmission_handler (r_buf=0xae036674) at timer.c:367 #3 0xb6247558 in retr_buf_handler (ticks=1234464444, tl=0xae036688, p=0x1f40) at timer.c:594 #4 0x0826a2cc in timer_list_expire (t=1234464444, h=0xad71c668, slow_l=0xad71cd44, slow_mark=2232) at core/timer.c:874 #5 0x08267cb1 in timer_handler () at core/timer.c:939 #6 0x0826a4d3 in timer_main () at core/timer.c:978 #7 0x08069575 in main_loop () at main.c:1721 #8 0x080707ca in main (argc=11, argv=0xbff64134) at main.c:2723 ERROR: tm [timer.c:366]: retransmission_handler(): transaction 0xae0365e0 scheduled for deletion and called from RETR timer (flags 6d) Both timers fired for an INVITE transaction that was previously suspended by async_route(), then resumed, sent out and received a 4xx reply (407). This configuration worked fine with kamailio 4.2.x and problem appeared after upgrading to 5.0.2. Trying to figure out how to narrow down the problem. Any input is appreciated.

2 6

alias_db muliple usernames for alias
by Aidar Kamalov 07 Sep '17

07 Sep '17

Hello, I have modparam("alias_db", "append_branches", 1) and table dbaliases +----+----------------+-----------------+----------+-----------------+ | id | alias_username | alias_domain | username | domain | +----+----------------+-----------------+----------+-----------------+ | 1 | 1111 | sipchel | 34200 | sipchel | | 2 | 1111 | sipchel | 34201 | sipchel | +----+----------------+-----------------+----------+-----------------+ If I call to 1111 ringing only 34200, because first invite is right (resolving to username contact), but branch inivite is wrong. Is it bug or it is my my fault? First: INVITE sip:34200@172.16.3.7:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.54;branch=z9hG4bKa277.42eb85466191923b73c9234c5c342b67.0 Via: SIP/2.0/UDP 172.16.3.4:5060;branch=z9hG4bK-c5fef8cc From: "" <sip:53003@192.168.10.54>;tag=1898fba2a5729040o0 To: <sip:1111@192.168.10.54> Second: INVITE sip:34201@sipchel SIP/2.0 Record-Route: <sip:192.168.10.54;lr=on;ftag=1898fba2a5729040o0;did=958.fd11> Via: SIP/2.0/UDP 192.168.10.54;branch=z9hG4bKa277.42eb85466191923b73c9234c5c342b67.1 Via: SIP/2.0/UDP 172.16.3.4:5060;branch=z9hG4bK-c5fef8cc From: "" <sip:53003@192.168.10.54>;tag=1898fba2a5729040o0 To: <sip:1111@192.168.10.54> -- Aydar A. Kamalov

1 0

handle_tcpconn_ev(): connect failed
by Donat Zenichev 07 Sep '17

07 Sep '17

Hi. Recently I've come across with TCP connection problem. The topology is as following: DNS srv load balancer - two kamailio proxy servers - one routing server. Client appeals to NAPTR record like: sip.domain.com So dns returns one of the proxy servers to client (depending on weight/priority). Now both kamailio have the same priority and weight (the goal is load balancing). Routing server (now it is asterisk) working with chan_pjsip.so, that supports NAPTR/SRV records. He is able to resolve Record-Route / Route headers with value - sip.domain.com (that proxy servers add to record-route headers while relaying requests to him). This topology is done to support present dialogs, even if proxy that recently processed it, is dead. But the problem comes, when routing server (asterisk) sends in-dialog requests to the proxy, that wasn't used to establish the dialog. Example, routing server obtains 200 OK from endpoint (relayed by kamailio1 to him) and he sends back ACK, but not to the kamailio1, he sends it to kamailio2 (because he resolves NAPTR sip.domain.com and gets ip of second kamailio). Kamailio2 processes the request as usual, because both kamailio have the same db for dialog module, but when he tries to relay the request to endpoint, he gots the error: ERROR: <core> [tcp_main.c:4070]: handle_tcpconn_ev(): connect XXX.XXX.XXX.XXX:52185 failed The port that kamailio2 tries to use to relay the ACK, is port that endpoint used to establish the dialog with kamailio1 and actually his TCP connection is now established with kamailio1. So kamailio2 tries to use the same port and gets the error. And this is proper behavior I think. There is no problem with UDP transport. Has anyone seen the similar problem? That indeed is not a problem, but proper behavior. -- -- BR, Donat Zenichev Wnet VoIP team Tel: +380(44) 5-900-808 http://wnet.ua

2 4

cr failure routes
by Nelson Migliaro 07 Sep '17

07 Sep '17

Hello, I am using carrierroute in Kamailio (great module btw). All calls go the same gateway (rewrite_host). I am routing over differents vendor using a prefix. Prefix: 21 dnis: 999999999 $rU after rewrite_host is 21999999999 My problem is when trying to use a failure route, next domain use a different prefix but the host. So I get the: Candidate destination <gateway> was previously used Is there a flag or something to prevent this verification ? Thank you Nelson.-

2 1

How can client get registered users list from Kamailio server or server may broadcast this actively?
by Jack Wang 07 Sep '17

07 Sep '17

Hey guys, As indicated by the title, is there any module providing the related function? My server version is 4.2. Thanks. :)

2 1

Cost deduction in kamailio
by Logeshwaran G 07 Sep '17

07 Sep '17

Hi, How to deduct the call cost or credits for the subscribers with the cnxcc module? Input Please Thanks & Kind Regards, Logeshwaran G

3 15

Query in LCR
by Logeshwaran G 07 Sep '17

07 Sep '17

Hi, I am handling Two Different Account holders and Two Different Gateways with the different lcr_ids: id :1 lcr_id:1 gw_name:car1 ip_addr : 1.2.3.4 id :2 lcr_id:2 gw_name:car2 ip_addr : 1.2.5.3 In kamailio config file I am using load_gws(1), Here Comes my doubt, I need to Pass the lcr_id value inside the load_gws(lcr_id) Dynamically, Is it Possible , Please Guide me. Thanks & Kind Regards, *Logeshwaran G* *"Be Hunger, Stay Foolish"*

2 2

Doubt in Lcr_table
by Logeshwaran G 07 Sep '17

07 Sep '17

Hi, May I know the Difference between the "prefix" column in Lcr_gw table and lcr_rule table. I am using the prefix value in lcr_rule table, like 00091. But What is the use of Prefix Column in lcr_gw table. Input Please Thanks & Kind Regards, *Logeshwaran G* *"Be Hunger, Stay Foolish"*

2 2

Loading keys with key_type="1" from database for htable - Module params help needed
by George Diamantopoulos 06 Sep '17

06 Sep '17

3 3

Call pickup with feature code
by Cyrille Demaret 06 Sep '17

06 Sep '17

Hi, I'm trying to implement call pickup with a feature code for phones that doesn't support "Replaces" header. User has to dial for example *8 to pickup a call who is in the same group pickup. My idea is to loop for each members of the group (coming from a SQL table using sqlops) and check for calls in early state. I was thinking to use the dialog module to search for active calls but I don't find any function allowing to do this. Does anyone know a module with a function allowing to find an active calls by callee? Thank you, Cyrille

1 0

SQL for ordered LCR rules
by Patrick Wakano 06 Sep '17

06 Sep '17

Hello list, Hope you all doing well! I am trying to ease the management of LCR routing rules, since once we begin to have multiple prefixes, multiple GWs and so on, the visualization and management of the rules priorities becomes exponentially hard to do. So first thing I am trying to achieve is an easy way of retrieving the rules in an ordered manner. I couldn't find any tool to do such thing and source code was not very friendly.... so I've come up with this Postgresql query that I think retrieves all rules in the same order I expect LCR to select the GWs. SELECT lr.lcr_id, lr.prefix, lrt.priority, lg.gw_name, lg.ip_addr FROM lcr_rule lr JOIN lcr_rule_target lrt ON lrt.lcr_id = lr.lcr_id AND lrt.rule_id = lr.id JOIN lcr_gw lg ON lg.lcr_id = lr.lcr_id AND lg.id = lrt.gw_id WHERE lr.enabled = 1 AND lg.defunct = 0 AND lr.lcr_id = ID AND lr.prefix SIMILAR TO '(|PREFIX%)' ORDER BY lr.lcr_id, LENGTH(lr.prefix) DESC, lrt.priority; It is missing the weights calculation, but it is rather complex and I am not using it anyway.... Other than that does anyone did something similar to check if my query really matches what LCR engine does? Thanks, Patrick Wakano

4 3

Re: [SR-Users] SQL for ordered LCR rules
by Patrick Wakano 06 Sep '17

06 Sep '17

Thanks for the response guys! The link https://github.com/dimitri/prefixIt is returning 404.... Regarding the performance itself I am not worried since this select it is just for management and I don't expect having millions of rules. The idea is just to have an easy way to have a picture of how the LCR will order and select the gateways based on a given prefix. The three LCR tables are not so easy to handle and manage from command line so my idea was to have a single SELECT or VIEW to return me all I need at once! >From what I could check, I think the select I sent pretty much translates what LCR module does internally, I am just trying to verify if it has some flaw, which could mislead me in the rules management. Cheers, Patrick Wakano On 6 September 2017 at 00:32, Dmitry Sinina <dmitry.sinina(a)onat.edu.ua> wrote: > https://yeti-switch.org/demo.html > > > On 9/5/17 5:29 PM, Dmitry Sinina wrote: > >> And you can try our opensource LCR engine. We use kamailio as load >> balancer and SEMS as SBC. >> >> On 9/5/17 3:02 AM, Patrick Wakano wrote: >> >>> Hello list, >>> >>> Hope you all doing well! >>> I am trying to ease the management of LCR routing rules, since once we >>> begin to have multiple prefixes, multiple GWs and so on, the visualization >>> and management of the rules priorities becomes exponentially hard to do. >>> So first thing I am trying to achieve is an easy way of retrieving the >>> rules in an ordered manner. I couldn't find any tool to do such thing and >>> source code was not very friendly.... so I've come up with this Postgresql >>> query that I think retrieves all rules in the same order I expect LCR to >>> select the GWs. >>> >>> SELECT lr.lcr_id, lr.prefix, lrt.priority, lg.gw_name, lg.ip_addr >>> FROM lcr_rule lr >>> JOIN lcr_rule_target lrt ON lrt.lcr_id = lr.lcr_id AND lrt.rule_id = >>> lr.id <http://lr.id> >>> JOIN lcr_gw lg ON lg.lcr_id = lr.lcr_id AND lg.id <http://lg.id> = >>> lrt.gw_id >>> WHERE lr.enabled = 1 AND lg.defunct = 0 AND lr.lcr_id = ID AND lr.prefix >>> SIMILAR TO '(|PREFIX%)' >>> ORDER BY lr.lcr_id, LENGTH(lr.prefix) DESC, lrt.priority; >>> >>> It is missing the weights calculation, but it is rather complex and I am >>> not using it anyway.... Other than that does anyone did something similar >>> to check if my query really matches what LCR engine does? >>> >>> Thanks, >>> Patrick Wakano >>> >>> _______________________________________________ >>> Kamailio (SER) - Users Mailing List >>> sr-users(a)lists.kamailio.org >>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>> >>> >> >

2 4

real-time update question
by Stent, Bob 05 Sep '17

05 Sep '17

I am just starting out with Kamailio. One of the things I want to be able to do is add a domain to the configuration and have Kamailio pick it up without having to restart. I see there is a runtime update framework but I am having trouble navigating the documentation to find out what this actually does for me. Can somebody point me to the right set of docs to read to figure this out? Do I simply add an alias to the kamalio.cfg file and somehow send an RPC command to Kamalio to re-read configuration? Bob Stent Aspect Software 5 Technology Park Drive Suite #9 Westford, MA 01886 This email (including any attachments) is proprietary to Aspect Software, Inc. and may contain information that is confidential. If you have received this message in error, please do not read, copy or forward this message. Please notify the sender immediately, delete it from your system and destroy any copies. You may not further disclose or distribute this email or its attachments.

3 2

5.0 Version compiling gives warning
by Yasin CANER 05 Sep '17

05 Sep '17

3 3

Dialplan Module
by Nelson Migliaro 05 Sep '17

05 Sep '17

Hello, Do you know if its possible to

2 2

Re: [SR-Users] Incorrect value when setting number like values in Htable using Kamcmd
by Tony - Telecomy 05 Sep '17

05 Sep '17

Prefixing the numbers with "s:" fixed the issue. Thank you for your help. Tony

1 0

Incorrect value when setting number like values in Htable using Kamcmd
by Tony - Telecomy 05 Sep '17

05 Sep '17

Hello, I'm having an issue setting phone numbers values in htable using kamcmd when the phone number is above the integer range 2147483647. Initially I planned to set the values as strings but this is not working as expected as well. Please find below some examples: root@tst:~/tst/test# kamcmd htable.seti tst_sim integer_limit "2147483647" root@tst:~/tst/test# kamcmd htable.seti tst_sim integer_exceeded_limit "2147483648" root@tst:~/tst/test# kamcmd htable.sets tst_sim integer_as_string_valid "2147483647" root@tst:~/tst/test# kamcmd htable.sets tst_sim integer_as_string_invalid "2147483648" root@tst:~/tst/test# kamcmd htable.dump tst_sim | grep -A 1 integer name: integer_as_string_valid value: 2147483647 -- name: integer_limit value: 2147483647 -- name: integer_as_string_invalid value: 18446744071562067968 -- name: integer_exceeded_limit value: -2147483648 my ctl params are as follow: modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl") modparam("ctl", "binrpc_max_body_size", 1024) modparam("ctl", "binrpc_struct_max_body_size", 1024) Thanks, Tony

3 2

telcosuite3-crm sql
by Kordován Szabolcs 05 Sep '17

05 Sep '17

Hi, I would like to try the IMS according to the Kamailio World 2017: Kamailio For IMS And VoLTE presentation. Unfortunately I have problem with telcosuite, the sql structure is missing. Where can I find the whole create.sql? Thanks for help. Regards, Szabolcs

3 7

Kamailio SIP protocol over WSS
by Washington Costa 05 Sep '17

05 Sep '17

Hi, I solved my problem. I was missing load some modules and routers. I saw this file as example: https://gist.github.com/jesusprubio/4066845 but is important to know that every module the begin "mi" they aren't supporte on Kamailio 5.0. You will need replace for the module relative the version 5.0. Washington L da Costa Analista de Sistemas (82) 9 9823-4782

2 1

Question about fix_nated_contact.
by Володимир Іванець 04 Sep '17

04 Sep '17

Hello everyone, I have system setup where there are few Asterisk servers behind Kamailio that is behind the firewall. There is a problem with call initiation. When phone from outside sends INVITE to Kamailio, it then passes it to one of Asterisk servers. Asterisk server eventually replies to Kamailio with 200 OK. Contact header contains Asterisk's local IP address and port. Kamailio executes function *fix_nated_contact* before sending reply to the phone. However Contact header will still have Asterisk's local socket. *fix_nated_contact* is being executed with no doubt. I can see xlog output that I set before and after it. Can someone tell how exactly *fix_nated_contact* gets "*request's source address:port pair*" and if it can be printed to the log? Thank you!

1 0

Homer confusion ....
by Mark Boyce 04 Sep '17

04 Sep '17

Hi All Trying to get a basic version of homer working on Kamailio V5. So far I have sipcapture throwing everything in to a single mysql table. So first question; Should I be using sipcapture or siptrace? I’m not sure I can see the difference. Second question; On what I have working so far using sipcapture, I’m seeing the invites, subscribes, cancels, act’s all end up in the database. What I’m missing is any responses from Kamailio back to the handset (AUTh Challenges etc) modparam("sipcapture", "db_url", "mysql://sipcapture:secretpassword@localhost/homer_data") modparam("sipcapture", "capture_on", 1) modparam("sipcapture", "hep_capture_on", 1) modparam("sipcapture", "insert_retries", 5) modparam("sipcapture", "insert_retry_timeout", 10) plus a sprinkling of "sip_capture(); “ in the “xxx_route {“ blocks What am I missing? Mark

4 6

presence_xml mod_init(): Can't bind xcap_client
by Borislav Gergov 04 Sep '17

04 Sep '17

*Hi I have presence_xml parameter force_active set to 1 and still i get **error** presence_xml:mod_init: Can't bind xcap_client* *kamailio 5.1.0-dev6 (x86_64/linux) *

2 3

MWI forwarding between Kamailio and Freeswitch
by Wang, Wallace 04 Sep '17

04 Sep '17

Hello We're using Kamailio & freeswitch as my company voice system. So far, everything is good, except the MWI. I tried to find out how to forward MWI between Kamailio and Freeswitch, or let the Presence_MWI module can work with the freeswitch behind Kamailio. If it was possiable, could anyone give me an idea or some links or demo configure for this. I really appreciate your help. Best Regards Wallace Wang

2 1

uac +kamailio + dispatcher + asterisk
by Жан Базаров 04 Sep '17

04 Sep '17

Hello! provider trunks registration on kamailio UAC?! it's clear. but how does asterisk find out through which trunk the call should be made? I tried to set the header from asterisk dialplan. for example: if (is_method("INVITE")) { #record_route_preset("109.195.102.122"); route(DIRECTION); setflag(FLT_ACC); # do accounting } # ------ LOADBALANCE ROUTE ------------ # if(!ds_is_from_list()) { route(DISPATCH); } route[DISPATCH] { #round robin dispatching on gateways group '1' if(!ds_select_dst("1", "4")) { send_reply("404", "No destination"); exit; } xlog("L_DBG", "--- SCRIPT: going to <$ru> via <$du>\n"); t_on_failure("RTF_DISPATCH"); route(NATMANAGE); route(DRELAY); exit; } route[DIRECTION] { if ($hdr(x-trunk) != $null) { if (!is_method("BYE")){ $fu=""; t_on_failure("MANAGE_FAILURE"); $dlg_ctx(timeout_route) = "DIALOG_END"; $avp(i:10)=43200; $dlg_ctx(timeout_bye) = 0; sql_pvquery("ca", "select l_uuid, auth_username, auth_password, realm, l_domain, r_domain from uacreg where id='$hdr(x-trunk)'", "$avp(uuid), $avp(uname), $avp(passwd), $avp(realm), $avp(src_ipaddr), $avp(dst_ipaddr)"); t_on_failure("MANAGE_FAILURE"); $dlg_ctx(timeout_route) = "DIALOG_END"; $avp(i:10)=43200; $dlg_ctx(timeout_bye) = 0; $fu=""; uac_replace_from("sip:$avp(uname)@$avp(dst_ipaddr)"); $tu="sip:"+$tU+"@"+$avp(dst_ipaddr); $ru="sip:"+$tU+"@"+$avp(dst_ipaddr); remove_hf("Contact"); $var(contact)="sip:"+$avp(uname)+"@10.49.9.2:5060"; insert_hf("Contact: <$var(contact)>\r\n"); #insert_hf("Contact: "); msg_apply_changes(); fix_nated_register(); xlog("L_INFO","Contact header $var(contact) 111111111111111111111111111111111111111 is $ct {$ct}\n"); route(RELAY); } #### BYE } ### XTRUNK But in this configuration I do not come bye but when I register providers trunks on asterisk - problem with BYE not visible. but I can not register provider-trunks on all the asterisks, because incoming invite arrives at the link + address, and all the asteriscs ring. All my asterisk's behind nat

2 1

jsonrpcs execution error
by David Villasmil 04 Sep '17

04 Sep '17

Hello guys, i configured jsonrpc on kamailio.cfg and RPCFIFOPATH on kamctlrc to the same file, seems to work for some commands, but not for others: root@ip-172-31-30-160:/home/admin# kamctl trusted show +----+--------------+-------+--------------+--------------+------+----------+ | id | src_ip | proto | from_pattern | ruri_pattern | tag | priority | +----+--------------+-------+--------------+--------------+------+----------+ | 1 | 1.2.3.4 | udp | | NULL | | 0 | | 2 | 5.4.3.2 | udp | | NULL | | 0 | +----+--------------+-------+--------------+--------------+------+----------+ root@ip-172-31-30-160:/home/admin# kamctl trusted dump { "jsonrpc": "2.0", "error": { "code": -32000, "message": "Execution Error" }, "id": 32022 } root@ip-172-31-30-160:/home/admin# kamctl trusted reload { "jsonrpc": "2.0", "error": { "code": -32000, "message": "Execution Error" }, "id": 32041 } Any help? Thanks! Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

2 2

Carrierroute, uac_*, failure_route.
by yu＠yu-boot.ru 04 Sep '17

04 Sep '17

Hello. Have a following task. Select carrier by carrierroute module from db, try to route (dial, relay, bridge - you name it) call there and if call is failed/rejected due to any reason, go to failure_route(?), perform carrierroute lookup again for another carrier and route call there. Main problem is that different carriers require different From/To format. And Kamailio don't allow to use uac* functions in failure_route. How to resolve this? Maybe entire routing method is wrong for my task? Any advice? -- sent from myMail for Android

3 3

Re: [SR-Users] [Freeswitch-users] Fwd: Call drops after 32 seconds with ACK Timeout
by Karsten Horsmann 02 Sep '17

02 Sep '17

Hi Mykyta, since you are plain sip without encryption you can see your sip-traffic on the wire, use sngrep on the kamailio box to figure out whats with the sip-dialogs - it helps a lot. https://github.com/irontec/sngrep and if its more an kamailio config problem, it would be better to stay at sr-users@. Or you can setup an webhomer for debugging. Kind Regards 2017-09-01 17:49 GMT+02:00 Sean Ingham <sean(a)missionlabs.co.uk>: > Once a call is answered FreeSWITCH sends a 200 OK SIP message to your > Kamailio loadbalancer. As per the SIP spec your kamailio box should respond > to a 200 OK with an ACK. The FS logs show that it never receives an ACK, > and it repeatedly tries sending the 200 OK with no response. After 30 > seconds without an ACK FS will assume the call has failed and hangup. > > Looking at your Kamailio SIP logs you can see the 200 OKs getting through > okay, and no attempt to send an ACK back to FS. I think this indicates an > issue with your kamailio.cfg rather than FreeSWITCH. > > > Sean > www.wirex-precision.co.uk > > On Fri, Sep 1, 2017 at 3:40 PM, Микита Нікітенко < > nikitanikitenko(a)gmail.com> wrote: > >> >> Hello! I have server at digital ocean with a floating ip address and I >> have routed sip traffic to this floating ip. At this server installed >> kamailio which balancing between freeswitch servers. I have a problem that >> call drops after 32 seconds with error ACK Timeout. Traffic comming throug >> the kamailio proxy. >> >> kamailio.cfg - https://pastebin.com/x0CL3TEZ >> kamailio sip messages log - https://pastebin.com/wyT3cBbD >> freeswitch logs with sip debug - https://pastebin.com/zPysfQkV >> >> I'm struggling for a week and have no idea how to fix it. >> Thank you very much. >> Mykyta Nikitenko >> >> >> >> -- Mit freundlichen Grüßen *Karsten Horsmann*

2 1

Call drops after 32 seconds with ACK Timeout
by Микита Нікітенко 01 Sep '17

01 Sep '17

Hello! I have server at digital ocean with a floating ip address and I have routed sip traffic to this floating ip. At this server installed kamailio which balancing between freeswitch servers. I have a problem that call drops after 32 seconds with error ACK Timeout. kamailio.cfg - https://pastebin.com/x0CL3TEZ kamailio sip messages log - https://pastebin.com/wyT3cBbD freeswitch logs with sip debug - https://pastebin.com/zPysfQkV I'm struggling for a week and have no idea how to fix it. Thank you very much. Mykyta Nikitenko

2 1

Change on the fly logging verbosity - Kamailio v5.0.2
by Karsten Horsmann 01 Sep '17

01 Sep '17

Hello, at this moment i try to figure out the best way to add an "switch" for debug/log output. I use many xlog, like everyone that starts with kamailio too. Then i go further and add #!ifdef cases around to switch this via #!define. Is there any other way to turn on/off my "debug"-logging on the fly, while kamailio is still running? #!define TESTBED_MODE request_route{ ... #!ifdef TESTBED_MODE route(DEBUG); #!endif ... #!ifdef TESTBED_MODE xlog("L_INFO","---FIX_SIPS-after: ru:($ru)"); #!endif ... } -- Kind Regards *Karsten Horsmann*

3 3

Kamailio v5.0.3 Released
by Daniel-Constantin Mierla 01 Sep '17

01 Sep '17

Hello, Kamailio SIP Server v5.0.3 stable release is out! This is a maintenance release of the latest stable branch, 5.0, that includes fixes since the release of v5.0.2. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.0.x. Deployments running previous 5.0.x versions are strongly recommended to be upgraded to v5.0.3. For more details about version 5.0.3 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2017/09/kamailio-v5-0-3-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda

2 2

last pkg.stats entry
by Juha Heinanen 01 Sep '17

01 Sep '17

When I make pkg.stats query (latest master), I get a weird last entry. What is it? -- Juha ... { entry: 28 pid: 14675 rank: -4 used: 9450336 free: 2354944 real_used: 10227968 total_size: 12582912 total_frags: 310 } { entry: 29 pid: 0 rank: -128 used: 0 free: 0 real_used: 0 total_size: 12582912 total_frags: 0 }

2 5

Planning Kamailio v5.0.3
by Daniel-Constantin Mierla 01 Sep '17

01 Sep '17

Hello, I am considering to release Kamailio v5.0.3 sometime next week, therefore it is the time to backport patches with fixes from master that apply to 5.0 branch. If you are aware of such fix that has to be backported, reply here or make a pull request on github project. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - www.kamailioworld.com

2 3

Kamailio SIP protocol over WSS
by Washington Costa 31 Aug '17

31 Aug '17

Hello, I'm using kamailio 5.0.2 (x86_64/linux). And it works with WS. Regards, Washington Costa

1 0

Kamailio SIP protocol over WSS
by Washington Costa 31 Aug '17

31 Aug '17

Hi. I'm trying use the JSSIP cliente and Kamailio with WSS. On these sites I generated my certificates: https://letsencrypt.org/ and https://certbot.eff.org/ My browser identify the certificates but SIP client just say: failed: WebSocket opening handshake was canceled Do you have any idea how to fix it? Do you can say a checklist with everything I need to do for use Kamailio with WSS? Here is my log on Kamailio: Aug 31 09:10:10 ip-172-31-39-220 kamailio[13183]: 33(13239) DEBUG: <core> [core/ip_addr.c:229]: print_ip(): tcpconn_new: new tcp connection: 187.41.60.10 Aug 31 09:10:10 ip-172-31-39-220 kamailio[13183]: 33(13239) DEBUG: <core> [core/tcp_main.c:985]: tcpconn_new(): on port 44630, type 3 Aug 31 09:10:10 ip-172-31-39-220 kamailio[13183]: 33(13239) DEBUG: <core> [core/tcp_main.c:1291]: tcpconn_add(): hashes: 3062:1194:1491, 2 Aug 31 09:10:10 ip-172-31-39-220 kamailio[13183]: 33(13239) DEBUG: <core> [core/io_wait.h:380]: io_watch_add(): DBG: io_watch_add(0xa4ac60, 53, 2, 0x7f259df9cd80), fd_no=44 Aug 31 09:10:10 ip-172-31-39-220 kamailio[13183]: 33(13239) DEBUG: <core> [core/io_wait.h:602]: io_watch_del(): DBG: io_watch_del (0xa4ac60, 53, -1, 0x0) fd_no=45 called Aug 31 09:10:10 ip-172-31-39-220 kamailio[13183]: 33(13239) DEBUG: <core> [core/tcp_main.c:4139]: handle_tcpconn_ev(): sending to child, events 1 Aug 31 09:10:10 ip-172-31-39-220 kamailio[13183]: 33(13239) DEBUG: <core> [core/tcp_main.c:3818]: send2child(): selected tcp worker 1 26(13225) for activity on [tls:172.31.39.220:4443], 0x7f259df9cd80 Aug 31 09:10:10 ip-172-31-39-220 kamailio[13183]: 26(13225) DEBUG: <core> [core/tcp_read.c:1574]: handle_io(): received n=8 con=0x7f259df9cd80, fd=9 Aug 31 09:10:10 ip-172-31-39-220 kamailio[13183]: 26(13225) DEBUG: tls [tls_server.c:197]: tls_complete_init(): completing tls connection initialization Aug 31 09:10:10 ip-172-31-39-220 kamailio[13183]: 26(13225) DEBUG: tls [tls_server.c:226]: tls_complete_init(): Using initial TLS domain TLSs<default> (dom 0x7f259ddc4a80 ctx 0x7f259de03bc0 sn []) Aug 31 09:10:10 ip-172-31-39-220 kamailio[13183]: 26(13225) DEBUG: tls [tls_domain.c:715]: sr_ssl_ctx_info_callback(): SSL handshake started Aug 31 09:10:10 ip-172-31-39-220 kamailio[13183]: 26(13225) DEBUG: <core> [core/tcp_main.c:2430]: tcpconn_do_send(): sending... Aug 31 09:10:10 ip-172-31-39-220 kamailio[13183]: 26(13225) DEBUG: <core> [core/tcp_main.c:2464]: tcpconn_do_send(): after real write: c= 0x7f259df9cd80 n=2568 fd=9 Aug 31 09:10:10 ip-172-31-39-220 kamailio[13183]: 26(13225) DEBUG: <core> [core/tcp_main.c:2465]: tcpconn_do_send(): buf= Aug 31 09:10:10 ip-172-31-39-220 kamailio[13183]: #026#003#001 Aug 31 09:10:10 ip-172-31-39-220 kamailio[13183]: 26(13225) DEBUG: <core> [core/io_wait.h:380]: io_watch_add(): DBG: io_watch_add(0xa93f00, 9, 2, 0x7f259df9cd80), fd_no=1 Aug 31 09:10:10 ip-172-31-39-220 kamailio[13183]: 26(13225) DEBUG: <core> [core/tcp_read.c:195]: tcp_emit_closed_event(): TCP closed event creation triggered Aug 31 09:10:10 ip-172-31-39-220 kamailio[13183]: 26(13225) DEBUG: <core> [core/tcp_read.c:202]: tcp_emit_closed_event(): no callback registering for handling TCP closed event - dropping! Aug 31 09:10:10 ip-172-31-39-220 kamailio[13183]: 26(13225) DEBUG: <core> [core/tcp_read.c:305]: tcp_read_data(): EOF on 0x7f259df9cd80, FD 9 ([187.41.60.10]:44630 ->26(13225) DEBUG: <core> [core/tcp_read.c:306]: tcp_read_data(): -> [172.31.39.220]:4443) Aug 31 09:10:10 ip-172-31-39-220 kamailio[13183]: 26(13225) DEBUG: <core> [core/tcp_read.c:1341]: tcp_read_req(): EOF Aug 31 09:10:10 ip-172-31-39-220 kamailio[13183]: 26(13225) DEBUG: <core> [core/io_wait.h:602]: io_watch_del(): DBG: io_watch_del (0xa93f00, 9, -1, 0x10) fd_no=2 called Aug 31 09:10:10 ip-172-31-39-220 kamailio[13183]: 26(13225) DEBUG: <core> [core/tcp_read.c:1495]: release_tcpconn(): releasing con 0x7f259df9cd80, state -1, fd=9, id=2 ([187.41.60.10]:44630 -> [187.41.60.10]:4443) Aug 31 09:10:10 ip-172-31-39-220 kamailio[13183]: 26(13225) DEBUG: <core> [core/tcp_read.c:1499]: release_tcpconn(): extra_data 0x7f259dfb6488 Aug 31 09:10:10 ip-172-31-39-220 kamailio[13183]: 33(13239) DEBUG: <core> [core/tcp_main.c:3250]: handle_tcp_child(): reader response= 7f259df9cd80, -1 from 1 Aug 31 09:10:10 ip-172-31-39-220 kamailio[13183]: 33(13239) DEBUG: tls [tls_server.c:665]: tls_h_close(): Closing SSL connection 0x7f259dfb6488 Aug 31 09:10:17 ip-172-31-39-220 kamailio[13183]: 33(13239) DEBUG: <core> [core/ip_addr.c:229]: print_ip(): tcpconn_new: new tcp connection: 187.41.60.10 Aug 31 09:10:17 ip-172-31-39-220 kamailio[13183]: 33(13239) DEBUG: <core> [core/tcp_main.c:985]: tcpconn_new(): on port 44692, type 3 Aug 31 09:10:17 ip-172-31-39-220 kamailio[13183]: 33(13239) DEBUG: <core> [core/tcp_main.c:1291]: tcpconn_add(): hashes: 2869:1129:1296, 3 Aug 31 09:10:17 ip-172-31-39-220 kamailio[13183]: 33(13239) DEBUG: <core> [core/io_wait.h:380]: io_watch_add(): DBG: io_watch_add(0xa4ac60, 53, 2, 0x7f259df9cd80), fd_no=44 Aug 31 09:10:17 ip-172-31-39-220 kamailio[13183]: 33(13239) DEBUG: <core> [core/io_wait.h:602]: io_watch_del(): DBG: io_watch_del (0xa4ac60, 53, -1, 0x0) fd_no=45 called Aug 31 09:10:17 ip-172-31-39-220 kamailio[13183]: 33(13239) DEBUG: <core> [core/tcp_main.c:4139]: handle_tcpconn_ev(): sending to child, events 1 Aug 31 09:10:17 ip-172-31-39-220 kamailio[13183]: 33(13239) DEBUG: <core> [core/tcp_main.c:3818]: send2child(): selected tcp worker 2 27(13226) for activity on [tls:172.31.39.220:4443], 0x7f259df9cd80 Aug 31 09:10:17 ip-172-31-39-220 kamailio[13183]: 27(13226) DEBUG: <core> [core/tcp_read.c:1574]: handle_io(): received n=8 con=0x7f259df9cd80, fd=9 Aug 31 09:10:17 ip-172-31-39-220 kamailio[13183]: 27(13226) DEBUG: tls [tls_server.c:197]: tls_complete_init(): completing tls connection initialization Aug 31 09:10:17 ip-172-31-39-220 kamailio[13183]: 27(13226) DEBUG: tls [tls_server.c:226]: tls_complete_init(): Using initial TLS domain TLSs<default> (dom 0x7f259ddc4a80 ctx 0x7f259de05708 sn []) Aug 31 09:10:17 ip-172-31-39-220 kamailio[13183]: 27(13226) DEBUG: tls [tls_domain.c:715]: sr_ssl_ctx_info_callback(): SSL handshake started Aug 31 09:10:17 ip-172-31-39-220 kamailio[13183]: 27(13226) DEBUG: <core> [core/tcp_main.c:2430]: tcpconn_do_send(): sending... Aug 31 09:10:17 ip-172-31-39-220 kamailio[13183]: 27(13226) DEBUG: <core> [core/tcp_main.c:2464]: tcpconn_do_send(): after real write: c= 0x7f259df9cd80 n=2568 fd=9 Aug 31 09:10:17 ip-172-31-39-220 kamailio[13183]: 27(13226) DEBUG: <core> [core/tcp_main.c:2465]: tcpconn_do_send(): buf= Aug 31 09:10:17 ip-172-31-39-220 kamailio[13183]: #026#003#001 Aug 31 09:10:17 ip-172-31-39-220 kamailio[13183]: 27(13226) DEBUG: <core> [core/io_wait.h:380]: io_watch_add(): DBG: io_watch_add(0xa93f00, 9, 2, 0x7f259df9cd80), fd_no=1 Aug 31 09:10:17 ip-172-31-39-220 kamailio[13183]: 27(13226) DEBUG: <core> [core/tcp_read.c:195]: tcp_emit_closed_event(): TCP closed event creation triggered Aug 31 09:10:17 ip-172-31-39-220 kamailio[13183]: 27(13226) DEBUG: <core> [core/tcp_read.c:202]: tcp_emit_closed_event(): no callback registering for handling TCP closed event - dropping! Aug 31 09:10:17 ip-172-31-39-220 kamailio[13183]: 27(13226) DEBUG: <core> [core/tcp_read.c:305]: tcp_read_data(): EOF on 0x7f259df9cd80, FD 9 ([187.41.60.10]:44692 ->27(13226) DEBUG: <core> [core/tcp_read.c:306]: tcp_read_data(): -> [172.31.39.220]:4443) Aug 31 09:10:17 ip-172-31-39-220 kamailio[13183]: 27(13226) DEBUG: <core> [core/tcp_read.c:1341]: tcp_read_req(): EOF Aug 31 09:10:17 ip-172-31-39-220 kamailio[13183]: 27(13226) DEBUG: <core> [core/io_wait.h:602]: io_watch_del(): DBG: io_watch_del (0xa93f00, 9, -1, 0x10) fd_no=2 called Aug 31 09:10:17 ip-172-31-39-220 kamailio[13183]: 27(13226) DEBUG: <core> [core/tcp_read.c:1495]: release_tcpconn(): releasing con 0x7f259df9cd80, state -1, fd=9, id=3 ([187.41.60.10]:44692 -> [187.41.60.10]:4443) Aug 31 09:10:17 ip-172-31-39-220 kamailio[13183]: 27(13226) DEBUG: <core> [core/tcp_read.c:1499]: release_tcpconn(): extra_data 0x7f259dfb6488 Aug 31 09:10:17 ip-172-31-39-220 kamailio[13183]: 33(13239) DEBUG: <core> [core/tcp_main.c:3250]: handle_tcp_child(): reader response= 7f259df9cd80, -1 from 2 Aug 31 09:10:17 ip-172-31-39-220 kamailio[13183]: 33(13239) DEBUG: tls [tls_server.c:665]: tls_h_close(): Closing SSL connection 0x7f259dfb6488

2 1

question for best way to remove sips uris - TLS and sips in $ru - get_send_socket2(): protocol/port mismatch
by Karsten Horsmann 31 Aug '17

31 Aug '17

Hello List, of course - best way is not to use sips: uri scheme. But i have to deal with that. I try to configure an multihomed kamailio (public/private IP) in front of my SIP-Servers. One of my softphones variants used TLS for connection but the INVITES of this softphone use the "sips" URI scheme. As i saw in other posts, this seems to be an issue for many people. In my case it breaks the routing to my internal sip-server (only plain udp). Most of my config is similar to havfos example + TLS. https://github.com/havfo/WEBRTC-to-SIP/blob/master/etc/kamailio/kamailio.cfg I tried to fixed that with this extra route - but this breaks the dialog somewhere - after 60 seconds i have no audio and the BYE becomes another CSeq: in the request_route right after route(AUTH) i add my route(FIX_SIPS) if (is_method("INVITE|SUBSCRIBE")) { if ($ru =~ "^sips:") { route(FIX_SIPS); } else { record_route(); } } route[FIX_SIPS] { if ($ru =~ "^sips:") { xlog("L_INFO","---FIX_SIPS-before: ru:($ru)"); $var(orig_uri) = $ru; $ru = "sip:" + $rU + "@" + $td; xlog("L_INFO","---FIX_SIPS-now: $rU $td ru:($ru) orig_uri:($var(orig_uri))"); record_route(); #$ru = $var(orig_uri); xlog("L_INFO","---FIX_SIPS-after: ru:($ru)"); } } 172.20.120.81 = Kamailio private IP 172.20.120.79 = My internal SIP-Server (B2BUA) I used kamailio with CentOS 7 and RTPENGINE 63bdd615 from master kamailio-5.0.2-3.1.x86_64 kamailio-tls-5.0.2-3.1.x86_64 kamailio-mysql-5.0.2-3.1.x86_64 kamailio-websocket-5.0.2-3.1.x86_64 -- Kind Regards *Karsten Horsmann*

1 2

sql_do_query_async(): the db driver module doesn't support async query
by Juha Heinanen 31 Aug '17

31 Aug '17

sqlops README tells: 4.5. sql_query_async(connection, query) Make an async SQL query using 'connection', if implemented by db driver module (e.g., db_mysql) my sqlops sqlcon parameter specifies mysql driver, but still i get to syslog: Aug 24 08:02:05 rox1 /usr/bin/sip-proxy[13886]: ERROR: sqlops [sql_api.c:427]: sql_do_query_async(): the db driver module doesn'tsupport async query any idea why? -- juha

2 7

kamailio asterisk + provider sip trunks
by Жан Базаров 31 Aug '17

31 Aug '17

Good morning! kamailio + dispatcher + provider trunks. How to make RR for incoming calls from pstn? my trunks are authorized on an asterisk, but i have problem with multiple calls on each asterisk. If I do through uac then I have a problem with BYE, and in general the module is uncomfortable. Please tell me how to implement a bunch: kamailio + asterisk + provider trunks..

2 1

Kamailio failover - dialog (transaction) that is available on different proxies
by Donat Zenichev 30 Aug '17

30 Aug '17

Hi. I came up with idea to set up stand with two kamailio and one b2bua server (for routing). The idea consists of failover for dialogs, transactions. So if one of kamailio nodes is down, another one is able to catch up the dialog and let users to properly end up the session. For better realizing of it, I will try to describe the idea step by step: 1. UAC invites UAS, they've done three-way-handshake, media stream is up. 2. Kamailio that processed this dialog is down. 3. Users decided to end the session with BYE method, but proxy that processed their three-way-handshake recently is down, so one of ua sends BYE to the destination route that contains domain name (that both kamailio serve), BYE achieves the second kamailio to let him properly end the dialog. But, there is a big but, this second kamailio hasn't ever known about this dialog, he doesn't support any transactions for it and furthermore he doesn't know anything about this call-id. So the solution for it, as I think, is hidden in db mode for user location (columns that contain call-ids, branches etc. But I need to be sure, if I'm on the right way. For purpose, where one ip is served by two nodes, I have two solutions: -First one. I want to create heartbeat cluster with two kamailio nodes, they will have one shared ip address, so when one node gets down, another one brings up shared ip interface and implements the same actions that master does. -Another method is to assign a few ip addresses to one domain name (ip addresses of different kamailio proxies). So the goal looks simple, if someone has ever done something like that, I will be glad to read the ideas. -- -- BR, Donat Zenichev Wnet VoIP team Tel: +380(44) 5-900-808 http://wnet.ua

3 5

Simultaneous calls in kamailio
by Logeshwaran G 30 Aug '17

30 Aug '17

Hi, Is there any way(tools) to place Simultaneous Calls in kamailio: For Example: 1000 calls simultaneously in Kamailio. Input Please?? Thanks & Kind Regards, Logeshwaran G

4 6

Fwd: sdp extra \r\n in multipart body
by Diego Nadares 29 Aug '17

29 Aug '17

Hi Daniel! Thanks for your reply. The rtpengine module I think it's expecting two \r\n like: ...pcma/8000\r\n \r\n --uniqueboundary .... cpc,9\r\n \r\n --uniqueboundary-- I saw the code and I modified it but Can any one cofirm if multipart has to be like ...pcma/8000\r\n \r\n --uniqueboundary Or ...pcma/8000\r\n --uniqueboundary Or any one of those? Thanks again! Diego El El lun, 21 de ago. de 2017 a las 02:33, Daniel-Constantin Mierla < miconda(a)gmail.com> escribió: > Hello, > > there is \r\n after > > a=rtpmap:8 PCMA/8000 > > because > > --uniqueBoundary-- > > is on next line. > > Maybe is the fault of the code extracting sdp from multi-part body. Open > an issue on github bug tracker not to forget about it. If nobody looks at > it, I will try to do it, but it may take few days given traveling. > > Cheers, > Daniel > > On 18.08.17 22:21, Diego Nadares wrote: > > Hi guys, > > I have the following multipart body > > Content-Type: multipart/mixed;boundary=uniqueBoundary > Remote-Party-ID: <tel:1112222;phone-context=+54> > <1112222;phone-context=+54> > Content-Length: 364 > > --uniqueBoundary > Content-Type: application/sdp > > v=0 > o=user1 53655765 2353687637 IN IP4 111.111.111.111 > s=- > c=IN IP4 111.111.111.111 > t=0 0 > m=audio 6000 RTP/AVP 8 > *a=rtpmap:8 PCMA/8000* > *--uniqueBoundary--* > Content-Type: application/gtd > Content-Disposition: signal;handling=optional > > IAM, > CPC,20 > GCI,885sdfasdf1123 > > --uniqueBoundary-- > > > As you can see *--uniqueBoundary *before *a=rtpmap:8 PCMA/8000* has not > an extra '\r\n'. Is this correct? Or should have it? > > The thing is that rtpengine is receiving sdp with no final '\r'n' from > kamailio: > > Dump for 'offer' from 127.0.0.1:37388: { "sdp": "v=0#015#012o=user1 > 53655765 2353687637 IN IP4 111.111.111.111#015#012s=-#015#012c=IN IP4 > 111.111.111.111#015#012t=0 0#015#012m=audio 6000 RTP/AVP 8#015#012*a=rtpmap:8 > PCMA/8000"*, "ICE": "remove", "direction": [ "pub", "priv" ], "replace": > [ "origin", "session-connection" ], "transport-protocol": "RTP/AVP", > "call-id": "1-17755(a)111.111.111.111", "received-from": [ "IP4", > "111.111.111.111" ], "from-tag": "1", "command": "offer" } > > And the resulting sdp from rtpengine is this: > > --uniqueBoundary > Content-Type: application/sdp > > v=0 > o=user1 53655765 2353687637 IN IP4 172.16.213.16 > s=- > c=IN IP4 172.16.213.16 > t=0 0 > m=audio 35066 RTP/AVP 8 > *a=rtpmap:8 PCMA/8000a=sendrecv* > a=rtcp:35067 > > The problem is related with that '\r\n' but Is this a malformed message or > a possible bug? > > Thanks in advance! > > Diego > > > > > > > _______________________________________________ > Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > > -- > Daniel-Constantin Mierlawww.twitter.com/miconda -- www.linkedin.com/in/miconda > Kamailio Advanced Training - www.asipto.com > Kamailio World Conference - www.kamailioworld.com > >

2 3

Query in LCR Rule Target Table:
by Logeshwaran G 29 Aug '17

29 Aug '17

Hi, I need to know What is the Purpose of Priority and Weight column in Lcr_rule_target table: Example: | id | lcr_id | rule_id | gw_id | priority | weight | status | createdBy | createdDate | modifiedBy | modifiedDate | +----+--------+---------+-------+----------+--------+--------+-----------+-------------+------------+--------------+ | 1 | 1 | 1 | 1 | 1 | 1 | 1 | NULL | NULL | NULL | NULL | | 2 | 1 | 4 | 1 | 1 | 2 | 1 | NULL | NULL | NULL | NULL Kindly someone please Guide me. Thanks & Kind Regards, Logeshwaran G

2 4

LCR dont_strip_or_prefix_flag
by Patrick Wakano 29 Aug '17

29 Aug '17

Hello list, I am using Kamailio 5.0.2 on centos7 and I am trying to use the dont_strip_or_prefix_flag ( https://www.kamailio.org/docs/modules/5.0.x/modules/lcr.html#idp49131644) for LCR routing but I discovered that just by declaring the flag like this the module won't prefix my calls: #!define LCR_DONT_STRIP_OR_TAG 7 modparam("lcr", "dont_strip_or_prefix_flag", LCR_DONT_STRIP_OR_TAG) It doesn't matter if I set or reset the flag it never prefix the ReqUR if the declaration is present. I have to remove the declaration to make the module prefix it correctly. But then the whole purpose of the flag is lost....Anyone aware of this behaviour? Regards, Patrick Wakano

3 4

TSILO + NAT issues
by Emanuele Gambaro 28 Aug '17

28 Aug '17

Hi to all. I have the following setup-needs : *) Kamailio 5.0.1 with TSILO module and Apple push notification service. *) Rtpengine configured in order to force a "media-handover” for every call. *) NAT forced for all connection. I have a lot of problem with the INVRESUME route. When an INVITE is resumed (upon mobile client connection via APN message) the NAT does not work, and there is no rtp flow between devices. It works only if device is already online (so without the SUSPEND route). How can I setup TSILO in order to force NAT management for all calls? Thank you -- Emanuele Gambaro --- email: emanuele.gambaro(a)pynlab.com skype: sarbyn_work OpenPGP Key: https://goo.gl/fdeVnI

2 1

lcr gws
by Logeshwaran G 28 Aug '17

28 Aug '17

Hi, I am having two lcr gateways in lcr_gw table. But When executing the "kamcmd lcr.dump_gws", Its showing only one gateway list: [root@zeodialer ~]# kamcmd lcr.dump_gws { lcr_id: 1 gw_id: 1 gw_index: 1 gw_name: Carrier1 scheme: sip: ip_addr: 104.251.178.29 hostname: port: 5060 params: transport: ;transport=udp strip: 0 prefix: tag: flags: 1 state: 0 defunct_until: 0 } Did I missing something? Thanks & Kind Regards, Logeshwaran G

2 8

Fwd: Weird issue with kamailio relaying messages to itself
by George Diamantopoulos 26 Aug '17

26 Aug '17

Stupid gmail replying to sender only... ---------- Forwarded message ---------- From: George Diamantopoulos <georgediam(a)gmail.com> Date: 27 August 2017 at 02:29 Subject: Re: [SR-Users] Weird issue with kamailio relaying messages to itself To: Daniel-Constantin Mierla <miconda(a)gmail.com> Hello all, I've figured out what was going on, so I'm sharing here in case anyone else runs into this. I guess it should be a fairly common situation under certain circumstances when using the dispatcher module... The problem was that no destinations in the dispatcher set used were available for these requests. So $du was not set by ds_select_next. Which meant that when t_relay() was called later in the script, it would route based on R-URI, and the RURI's uri was kamailio itself. As to the reason why I was left with no dispatcher destinations available, well, I would mark destinations offline for 500 "server error" responses coming from them, and asterisk (which is the receiving application for all of dispatcher's destination sets) will send out a 500 to the B-leg when it receives a 480 to an A-leg. Getting a single 480 to one of the asterisk boxes would cause this to happen across all destinations, as kamailio would retry the next destination after a 500 failure and would receive a 500 from all of them in the end (because all of them would get the 480 in such small time frame). BR, George On 31 July 2017 at 21:52, George Diamantopoulos <georgediam(a)gmail.com> wrote: > Hello Daniel, > > Thanks for the reply. That is correct, looping was not my intention, I'm > trying to figure out what's causing it... > > BR, > George > > On 31 July 2017 at 17:29, Daniel-Constantin Mierla <miconda(a)gmail.com> > wrote: > >> Hello, >> >> usually you should not loop requests locally, unless a very special case >> -- do you do the loop routing because of a need or just happens but you >> don't know why? >> >> Cheers, >> Daniel >> >> On 31.07.17 13:46, George Diamantopoulos wrote: >> >> Hello all, >> >> I have been toying with kamailio lately, and I thought I had gotten to >> the point where I had a mostly working (tm) prototype. I gave it a test >> drive with some real calls, however, and an issue manifested at least once, >> where homer received packets originating from the kamailio host, and whose >> destination was also the kamailio host. >> >> The dialog this manifested in is a generally problematic one, with many >> retransmissions occurring because of slow database access (I haven't >> implemented htable caching yet). No packet capture over the network is >> actually taking place, I'm copying everything to homer with "trace_mode"set >> to 1. >> >> Homer shows these messages like in the screenshot: >> https://imagebin.ca/v/3VGJAmovRBmo. Here's an example of a packet: >> >> >> 2017-07-28 14:32:29 +0300 : 2.3.4.5:5060 -> 2.3.4.5:5060 >> INVITE sip:1234567890@kamailio-server.org SIP/2.0 >> Record-Route: <sip:2.3.4.5;lr;ftag=as491cec82> >> Via: SIP/2.0/UDP 2.3.4.5;branch=z9hG4bK0cf.0779 >> 9e3eb71f33a9ef91178ac760ebd2.0 >> Via: SIP/2.0/UDP 2.3.4.5;branch=z9hG4bKsr-BnCVy >> rWoUladI14pU7Pry-Pzy-ONy-VSQ74NU7HzeYazq2nFU2Oc5FIWiGIKq-HXe >> x1oONpam-C6IrIXkr4FQxZRh7sM >> Max-Forwards: 69 >> From: <sip:subscriber@5.4.3.2:5061>;tag=as491cec82 >> To: <sip:1234567890@kamailio-server.org> >> Contact: <sip:2.3.4.5;line=sr-eNC05xhKefQnON1EglFrOkF0OfpzV2s9UzM9UXM >> NQXMn5-B0Q-s*> >> Call-ID: 4e5d22c3369704b97d866c8d0f3798f8@192.168.201.2:5061 >> CSeq: 103 INVITE >> User-Agent: Asterisk PBX 11.13.1~dfsg-2~bpo70+1 >> Date: Fri, 28 Jul 2017 11:32:24 GMT >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, >> PUBLISH, MESSAGE >> Supported: replaces, timer >> Remote-Party-ID: "9876543210" <sip:9876543210@192.168.201.2> >> ;party=calling;privacy=off;screen=yes >> Content-Type: application/sdp >> Content-Length: 500 >> >> v=0 >> o=root 242468242 242468243 IN IP4 172.17.130.13 >> s=Asterisk PBX 11.13.1~dfsg-2~bpo70+1 >> c=IN IP4 172.17.130.13 >> t=0 0 >> m=audio 59426 RTP/AVP 18 101 >> a=rtpmap:18 G729/8000 >> a=fmtp:18 annexb=no >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=ptime:50 >> a=sendrecv >> a=rtcp:59427 >> a=ice-ufrag:hffIoy6x >> a=ice-pwd:nXB8ip7Qz8ZG5yyZRr97kGJbej >> a=candidate:igct4zWHEzhGCjWc 1 UDP 2130706431 <21%203070%206431> >> 172.17.130.13 59426 typ host >> a=candidate:igct4zWHEzhGCjWc 2 UDP 2130706430 <21%203070%206430> >> 172.17.130.13 59427 typ host >> >> Can something like this be triggered by misconfiguration in the routing >> scripts? Should it worry me and should I dig into it more? Could it be a >> bug of the siptrace module and nothing bad actually took place? I'm not >> sure where to start with this, so any input would be greatly appreciated. >> Thanks! >> >> >> _______________________________________________ >> Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> >> >> -- >> Daniel-Constantin Mierlawww.twitter.com/miconda -- www.linkedin.com/in/miconda >> Kamailio Advanced Training - www.asipto.com >> Kamailio World Conference - www.kamailioworld.com >> >> >

2 1

Kamailio fails to process ACK to 487 from Cisco-SIPGateway/IOS-12.x?
by George Diamantopoulos 26 Aug '17

26 Aug '17

Hello all, I'm having a weird issue with Kamailio failing to properly process an ACK received to a 487 it sent, resulting in retransmissions of the 487. I assume it's because it can't match the ACK to the transaction, but I could be wrong. I'm using a modified version of the default configuration, so ACKs should be handled correctly. I haven't editted the WITHINDLG route in any way that would affect this (or at least I think). In addition, ACKs to 487 from other UAs are processed correctly, and these transactions are handled by the same routes in kamailio configuration as the problematic one, so I'm inclined to believe it's UA-specific? Here's an example transaction of the failed kind (results in kamailio retransmitting the 487): myself:5060 -> peer:5060 ------------------------- SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP peer:5060 From: <sip:user@peer>;tag=116B5368-24D8 To: <sip:tel@myself>;tag=as655f6372 Call-ID: 84DC69F2-873811E7-8A639B5A-3D9194E8@peer CSeq: 101 INVITE Server: modCOM v2 SIP Server Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 peer:49590 -> myself:5060 ------------------------- ACK sip:tel@myself:5060 SIP/2.0 Via: SIP/2.0/UDP peer:5060 From: <sip:user@peer>;tag=116B5368-24D8 To: <sip:tel@myself>;tag=as655f6372 Date: Wed, 23 Aug 2017 12:50:47 GMT Call-ID: 84DC69F2-873811E7-8A639B5A-3D9194E8@peer Max-Forwards: 10 Content-Length: 0 CSeq: 101 ACK And here's another similar transaction which is successful (no retransmissions): myself:5060 -> peer:5060 ------------------------ SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP peer:5060;branch=z9hG4bKjbmvq4009gthskk0a6s1.1 From: <sip:user@anonymous.invalid;user=phone>;tag=599D7495-9ACE9E3-0A324A05 To: <sip:tel@anonymous.invalid:5060;user=phone>;tag=as65375e5d Call-ID: 599D7495-007A5832@fath3pcu238 CSeq: 1 INVITE Server: modCOM v2 SIP Server Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 peer:5060 -> myself:5060 ------------------------ ACK sip:tel@myself:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP peer:5060;branch=z9hG4bKjbmvq4009gthskk0a6s1.1 From: <sip:user@anonymous.invalid;user=phone>;tag=599D7495-9ACE9E3-0A324A05 To: <sip:tel@anonymous.invalid:5060;user=phone>;tag=as65375e5d Call-ID: 599D7495-007A5832@fath3pcu238 Max-Forwards: 69 Content-Length: 0 CSeq: 1 ACK I can't pinpoint anything wrong with the first exchange, other than the fact that for some reason, the "less than" (<) sign in the from and to URIs is escaped as &lt in homer's GUI (which also breaks CSS rendering in Firefox, I had to clear this code out). However, these escaping characters are not visible with sngrep when capturing traffic normally, and neither when doing a select in homer's database directly, so I guess it's a rendering bug in homer-ui and can be ignored (unless someone has reason to believe otherwise). Now the relevant portion of the debug log is: Aug 23 16:47:12 modcom-sbc-1 kamailio[9750]: {1 101 ACK 64AA4E6C-874011E7-9A729B5A-3D9194E8@peer} 7(9760) exec: *** cfgtrace:request_route=[WITHINDLG] c=[/etc/kamailio/kamailio.cfg] l=223 a=24 n=t_check_trans Aug 23 16:47:12 modcom-sbc-1 kamailio[9750]: {1 101 ACK 64AA4E6C-874011E7-9A729B5A-3D9194E8@peer} 7(9760) DEBUG: tm [t_lookup.c:1001]: t_check_msg(): msg id=104 global id=103 T start=(nil) Aug 23 16:47:12 modcom-sbc-1 kamailio[9750]: {1 101 ACK 64AA4E6C-874011E7-9A729B5A-3D9194E8@peer} 7(9760) DEBUG: tm [t_lookup.c:459]: t_lookup_request(): start searching: hash=54992, isACK=1 Aug 23 16:47:12 modcom-sbc-1 kamailio[9750]: {1 101 ACK 64AA4E6C-874011E7-9A729B5A-3D9194E8@peer} 7(9760) DEBUG: tm [t_lookup.c:494]: t_lookup_request(): proceeding to pre-RFC3261 transaction matching Aug 23 16:47:12 modcom-sbc-1 kamailio[9750]: {1 101 ACK 64AA4E6C-874011E7-9A729B5A-3D9194E8@peer} 7(9760) DEBUG: tm [t_lookup.c:641]: t_lookup_request(): no transaction found Aug 23 16:47:12 modcom-sbc-1 kamailio[9750]: {1 101 ACK 64AA4E6C-874011E7-9A729B5A-3D9194E8@peer} 7(9760) DEBUG: tm [t_lookup.c:1070]: t_check_msg(): msg id=104 global id=104 T end=(nil) Aug 23 16:47:12 modcom-sbc-1 kamailio[9750]: {1 101 ACK 64AA4E6C-874011E7-9A729B5A-3D9194E8@peer} 7(9760) exec: *** cfgtrace:request_route=[WITHINDLG] c=[/etc/kamailio/kamailio.cfg] l=231 a=2 n=exit It explicitly states that no transaction is found, after initiating pre-RFC3261 (why?) transaction matching. However, even pre-3261 matching should work, as the from and to headers as well as call-id in request and repy are the same. Any input would be greatly appreciated, thanks! George

2 3

One Kamailio and many RTPENGINEs
by Denys Pozniak 25 Aug '17

25 Aug '17

Hello! I want to build distributed topology with one KAMAILIO and 5x RTPENGINEs for each region. I will have dedicated links between these sites, so I can garantee somehow voice quality. If one customer from region1 calls customer region2, RTP should go: Customer1->RTEPNGINE1->RTPENGINE2->Customer2 So the main question: Can one Kamailio build "call" between two RTPENGINEs ? BR, Denys

4 4

Handeling Proxy Cancel event
by Abdul Basit 25 Aug '17

25 Aug '17

Hi, I am trying to capture and distinguish two types of cancels generated by kamailio. 1- Cancel when Invite timeout 408 2- Cancel generated when call lands on multiple devices and any one of them answer the call. This type of cancel comes with (rfc3326) Reason: SIP;cause=200;text="Call completed elsewhere" How can I capture and isolate these proxy generated CANCEL events? -- regards, abdul basit

3 5

Diaplan module
by Nelson Migliaro 24 Aug '17

24 Aug '17

Hello, I am trying to use the dialplan module but I am having this problem: First of all, I pull the "dpid" value from the two first digits of the dnis: $var(dialplan_vendor) = $(rU{s.substr,0,2}); Then if(!dp_translate("$var(dialplan_vendor)","$dlg_var(servicio)/$var(ani_dialplan)")) { xlog("L_NOTICE","Error al seleccionar el dialplan\n"); sl_send_reply("403", "Not allowed"); drop; } But when I test everything I get: ERROR: dialplan [dialplan.c:233]: dp_get_ivalue(): no AVP, XAVP or SCRIPTVAR found (error in scripts) I think the problem is related to variable $var(dialplan_vendor). It is set up as string due to the transformation and dialplan need a integer. If I set up the variable this way for testing: $var(dialplan_vendor) = 18, it works correctly. Question, how can I transform the sting into a integer? Thank you Nelson.

1 1

Consistency: -1 sometimes true, sometimes false
by Sebastian Damm 24 Aug '17

24 Aug '17

Hi, we recently stumbled upon the following problem: We had something like this in our route: $var(ret) = allow_address("2"); if ($var(ret)) { do_something; } Turns out, this always evaluates to true, no matter whether the IP address really is in address group 2 or not. The values in $var(ret) are 1 or -1, depending on the outcome of allow_address(). However, if we write it this way if (allow_address("2")) { do_something; } the code inside is only executed if the IP address is in address group 2. Is this some kind of inconsistency? Should negative values in variables evaluate to true or false? Best Regards, Sebastian

3 5

Feature request: USRLOC jsonrpc ul.add - set all entry fields
by Dmitry Sytchev 23 Aug '17

23 Aug '17

Hi Daniel! Looking into code it seems RPC functions of USRLOC don't allow setting all fields for USRLOC entry, for example, Received and Socket. It would be great if we can set these, because we need to have ability to transfer contacts from other server to Kamailio. I think at present this can be done sending specially crafted fake REGISTER packets or calling special route from HTTP with some tricks for saving contact and received, but as for me it would be great to set it from RPC because we already have ul.add method. It also would be cool if we can use named parameters in JSONRPC params: {} section with names corresponding to usrloc fields, like 'table', 'AOR', 'received' and so on. Maybe on specual key not to mix with current 'params'. Thanks for your great work! -- Best regards, Dmitry Sytchev, IT Engineer

2 1

Feature request: USRLOC jsonrpc ul.add - set all entry fields
by Dmitry Sytchev 23 Aug '17

23 Aug '17

Hi Daniel! Looking into code it seems RPC functions of USRLOC don't allow setting all fields for USRLOC entry, for example, Received and Socket. It would be great if we can set these, because we need to have ability to transfer contacts from other server to Kamailio. I think at present this can be done sending specially crafted fake REGISTER packets or calling special route from HTTP with some tricks for saving contact and received, but as for me it would be great to set it from RPC because we already have ul.add method. It also would be cool if we can use named parameters in JSONRPC params: {} section with names corresponding to usrloc fields, like 'table', 'AOR', 'received' and so on. Maybe on specual key not to mix with current 'params'. Thanks for your great work! -- Best regards, Dmitry Sytchev, IT Engineer

1 0

Kamailio APP_LUA
by santosh gupta 23 Aug '17

23 Aug '17

Hi Robert, Hope you are doing good. You have helped me in the past. I need to set LCR via app_lua, getting rates and destination from mysql. Link which i am following is https://www.kamailio.org/wiki/embeddedapi/4.0.x/lua The issue is I am not able to get connect via sr commands for example sr.sqlops.query("connection", "query", "result") What should be exactly is "connection", "query", "result" Error I am getting in syslog is /usr/local/kamailio/etc/kamailio/kamailio.lua:1: attempt to index field 'sqlops' (a nil value) I am using kamailio 5.0.2 A little bit of headsup will help me. Thanks a lot again. -- Regards Santosh Gupta +91 9323043570 <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_cam…> Virus-free. www.avg.com <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_cam…> <#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>

2 2

Query in Dialplan module
by Logeshwaran G 23 Aug '17

23 Aug '17

Hi, I have one query in Dialplan module, my scenario is If I call 22291+++++++++ the first three digit should remove and the call place to 91+++++++. But i dont get the exact result, the first three digits are not removing. I have used the below rules: Lcr_rule table: ============ id: 5 lcr_id: 2 Prefix:22291 Request uri: NULL Stopper: 1 Enabled: 1 LCR_rule_target: ================== id:3 lcr_id:2 rule_id:5 gw:id:2 priority:1 weight:1 Dialplan table: ============= id:3 dpid:2 pr:1 match_op:1 match_exp: ^222[0-9]+$ subst_exp: ^222[0-9]+$ repl_exp:\1 Is it correct or not, Please guide me. Thanks & Kind Regards, Logeshwaran G

2 3

Set up caller ID
by Nelson Migliaro 23 Aug '17

23 Aug '17

Hello, I have several Asterisk sending calls to a centralizad Kamailio. I would like this Kamailio to set up the caller id number of the calls. I am using the uac_replace_from function but I would like to pull the caller id from a database to set up the number based on the sip vendor the call is forwarded What would be the correct way to do that? I was thinking of: sql_query("ca", "select ddi from ddi_table where xxx= $rU limit 1", "ra"); $avp(dnis) = $dbr(ra=>[0,0]); sql_result_free("ra"); But I am afraid of causing a bottheneck on the database. As usual, thank you for the help. Regards, Nelson.-

2 1

rtpproxy rtp timeout
by yu＠yu-boot.ru 23 Aug '17

23 Aug '17

Hello. What version of rtpproxy app should be used for this feature to work with Kami? How to use it correctly? -- Отправлено из myMail для Android

3 4

snmpstats get calls that have ended in 5xx cause code
by Olli Attila 23 Aug '17

23 Aug '17

Hello, I have Kamailio 5.0 configured with snmpstats module. I can succesfully query for eg. the current num of dialogs on kamailio by issuing: snmpget -v 2c -c public localhost KAMAILIO-MIB::kamailioCurNumDialogs.0 KAMAILIO-MIB::kamailioCurNumDialogs.0 = Gauge32: 0 Is it possible to make query about calls that have ended with 5xx sip cause code on last 5min or 15min? I noticed that the following is available on the snmpstats module but it only holds a counter that keeps increasing until I restart kamailio process: snmpget -v 2c -c public localhost KAMAILIO-MIB::kamailioTotalNumFailedDialogSetups.0 Also I think the above includes all cause codes that are between 3xx-6xx. Cheers, --Olli

1 0

cr reload in xmlrpc
by Logeshwaran G 23 Aug '17

23 Aug '17

Hi, It is the correct way of using the cr reload command in xmlrpc through postman? <?xml version='1.0'?> <methodCall> *<methodName>cr.reload</methodName>* </methodCall> Thanks & Kind Regards, Logeshwaran G

2 6

Save original $fU, $rU
by yu＠yu-boot.ru 22 Aug '17

22 Aug '17

Have a problem with Asterisk. When callee behind Asterisk hangs up first, it sends back BYE with swapped From and To fields, which affects Kami CDRs rendering them useless. I've tried to save original From/To to AVPs on initial INVITE but they are empty when CDR is written. How to solve this? -- Отправлено из myMail для Android

3 2

dp_translate_cr_route, R-URI
by Yu Boot 22 Aug '17

22 Aug '17

Hello. #cut off prefix "9" dp_translate("1"); ... #searching in db by called number without prefix... if(!cr_route("default", "default", "$rU", "$rU", "call_id")){ ... and failing. Is it normal behavior that cr_route "sees" original $rU even after dp_translate manipulations?

1 0

Kamailio / Asterisk and multidomain
by Cyrille Demaret 22 Aug '17

22 Aug '17

Hi, I'm new to Kamailio and I'm trying to use it as a front SIP proxy to one or more asterisk. Unlike to the "Kamailio 4.0.x and Asterisk 11.3.0 Realtime Integration" tutorials, I would like to let Kamailio handle registrations, calls between users and other basic functionalities. Asterisk will only handle advanced features like voicemail, advanced dialplan configuration,. My problem is that It needs to be multidomain. My network will be as follow with 2 network interfaces for Kamailio and a private lan between Kamailio and Asterisk : Public interface ip 1.2.3.4 ----> [Kamailio] <----- Private interface 192.168.100.10 <----------------> 192.168.100.11 [Asterisk] When I send a call to asterisk, the domain is sent in the from field of the INVITE and I can do what I need on the Asterisk dialplan (I can get the SIP domain using the ${SIPDOMAIN} variable). My problem is when I need to send a call back to Kamailio for example to reach another user of the domain. I'm using Asterisk 14 with PJSIP with the following config : [kamailio] type=endpoint transport=transport-udp context=from-kamailio disallow=all allow=ulaw aors=kamailio [kamailio] type=aor contact=sip:192.168.100.10:5060 [kamailio] type=identify endpoint=kamailio match=192.168.100.10 If I use this dial string in my Asterisk dialplan "PJSIP/ kamailio /sip:200@testdomain.com", Asterisk contact directly testcomain.com without going through the local IP of my Kamailio. I can't send a domain to Kamailio in the INVITE request. Does anyone can help me on this or maybe simply tell me that I'm not going to the good direction? :) Thank you, Cyrille

3 3

Siremis - can't add carrier without prefix/suffix
by Yu Boot 22 Aug '17

22 Aug '17

Hello Suddenly revealed that I can't add carriers without rewrite_prefix/rewrite_suffix via Siremis. But I don't need these fields! And kamctl adds these rules fine. What's a problem? Screenshot with error http://tinypic.com/r/68v493/9

2 4

problems with use_dst_blacklist and ICMP "Destination Unreachable"
by Valentin Christoph 22 Aug '17

22 Aug '17

Hi all, Question: * Does the TM destination blacklist work, if destination host responds with ICMP "Destination Unreachable"? Discussion: I have got an IMS scenario, where I think the destination blacklist is not working and try to find the reason. I use UDP between two Kamailio SIP Proxies (P-CSCF and S-CSCF) and send a SIP request to the primary instance of the S-CSCF. However the destination Kamailio has been stopped with kamctl stop before and so the destination machine responds with ICMP "Destination Unreachable". DNS Failover towards secondary S-CSCF instance works perfectly. Destination Blacklist does not work. I.e. the next SIP Request is sent to the primary host again, before failover to secondary host applies. I would have expected the request would be sent to secondary host immediately. >From the description in doc/dst_blacklist.txt [...] A destination is added to the blacklist when an attempt to send to it fails (e.g. timeout while trying to send or connect on TCP), or when a SIP timeout occurs while trying to forward statefully an INVITE (using tm) and the remote side doesn't send back any response.[...] I think the destination blacklist does not work, because an ICMP Response "Destination Unreachable" is received. Second question: * How could I avoid the problem? Thanks, Christoph The information contained in this e-mail message is privileged and confidential and is for the exclusive use of the addressee. The person who receives this message and who is not the addressee, one of his employees or an agent entitled to hand it over to the addressee, is informed that he may not use, disclose or reproduce the contents thereof, and is kindly asked to notify the sender and delete the e-mail immediately.

1 0

rtpproxy: One direction rtp not being relayed
by Jon Bonilla (Manwe) 22 Aug '17

22 Aug '17

Hi all After a migration, a customer is reporting one way audio on incoming calls. A tcpdump capture shows 65 rtp packets in both directions but rtpproxy log shows RTP stats: 1 in from callee, 64 in from caller, 65 relayed, 0 dropped Any thoughts? cheers, Jon

2 1

sercmd dispatcher.list command giving an error: 500 - No Destination Sets
by Deepak Jaiswal 22 Aug '17

22 Aug '17

Hi, I am new to Kamailio and trying to debug an issue. We have two servers. On one server, I am getting destination like below. sercmd dispatcher.list { SET_NO: 1 SET: { SET_ID: 1 DEST: { URI: sip:xxx.xxx.xxx.139:5060 FLAGS: AX PRIORITY: 0 ATTRS: reg-time=1503397149 } DEST: { URI: sip:xxx.xxx.xxx.:5060 FLAGS: AX PRIORITY: 0 ATTRS: reg-time=1503397150 } } } However, on another server, I am getting below error. sercmd dispatcher.list error: 500 - No Destination Sets sercmd -s unixs:/tmp/kamailio_ctl dispatcher.list error: 500 - No Destination Sets [root@uk-vap-002 ~]# sercmd system.listMethods cfg.add_group_inst cfg.commit cfg.del cfg.del_delayed cfg.del_group_inst cfg.diff cfg.get cfg.help cfg.list cfg.rollback cfg.set cfg.set_delayed cfg.set_delayed_int cfg.set_delayed_string cfg.set_now_int cfg.set_now_string cfg.seti cfg.sets core.arg core.echo core.flags core.info core.kill core.printi core.prints core.ps core.psx core.pwd core.sctp_info core.sctp_options core.shmmem core.tcp_info core.tcp_options core.udp4_raw_info core.uptime core.version ctl.connections ctl.listen ctl.who dispatcher.list List going on.................................... I tried to find dispatcher.list file on both server. However, it is not there on both the server. Could you please help me on this? Thanks and Regards Deepak K. Jaiswal Tech support Engineer My Phone : +91- 95351-23352, +91- 88675-01385 My Movius work line: +1 (470) 298-7210 [ovius]<http://www.moviuscorp.com/> www.moviuscorp.com<http://www.moviuscorp.com/>

1 0

append_branch and tls
by James Cloos 21 Aug '17

21 Aug '17

[I see I posted this to -dev instead of -users; fixing that -JimC] When using rewrite_ruri() and append_branch() to set a list of next hops to try, what is the proper syntax to specify that a given route should use tcp or tls/tcp? I've mostly used a single next hop via t_relay_to_tcp() or t_relay_to_tls(), but need route stacks now. -JimC -- James Cloos <cloos(a)jhcloos.com> OpenPGP: 0x997A9F17ED7DAEA6

2 2

Kamailio+SEMs conference call setting
by leslie mekiyess 21 Aug '17

21 Aug '17

Hi, I have a Kamailio server , a SEM one, an SQL server and Siremis installed and working. I am able to perform p2p calls with a softmobile. But I didn't succeed to perform conference / group / speed dial calls. Besides defining some records in the database, I don't know how to complete the necessary (Kamailio/SEM ...) configuration in order to perform conference call, group and speed dial . Any assitance would very appreciated. Leslie :-)

1 0

siptrace: how to duplicate the complete dialog?
by Daniel Tryba 21 Aug '17

21 Aug '17

I'm trying to trace a complete dialog on a loadbalancer/proxy: UAC <-> Proxy <-> backend 109.235.34.a <-> 109.235.32.b <-> 185.61.68.c Proxy runs 4.3.x (but with 4.4.x I appear to have the same results). siptrace config: ============================================= modparam("siptrace", "duplicate_uri", "sip:10.11.12.13:9060") modparam("siptrace", "hep_mode_on", 1) modparam("siptrace", "trace_to_database", 0) modparam("siptrace", "trace_flag", 22) modparam("siptrace", "trace_on", 1) ============================================= After REQINIT (which does pike and handles OPTIONS), I setflag(22) the message. ============================================= route(REQINIT); setflag(22); route(NATDETECT); route(WITHINDLG); ============================================= Next depending in where the message comes from: ============================================= if(!is_in_subnet($si, backends)) { ds_select_dst(1, 0); t_on_failure("RTF_DISPATCH"); } else { remove_hf("Route"); loose_route(); } route(RELAY); ============================================= sip_trace() is being called in the routes: WITHINDLG MANAGE_BRANCH MANAGE_REPLY MANAGE_FAILURE RTF_DISPATCH The resulting duplicate packets are: 11:20:21.037 INVITE 109.235.34.a:39621 109.235.32.b:5060 tcp 11:20:21.040 INVITE 109.235.34.a:39621 109.235.32.b:5060 tcp 11:20:21.047 401 185.61.68.c:5060 109.235.32.b:5060 udp 11:20:21.073 ACK 109.235.34.a:39621 109.235.32.b:5060 tcp 11:20:21.117 INVITE 109.235.34.a:39621 109.235.32.b:5060 tcp 11:20:21.135 100 185.61.68.c:5060 109.235.32.b:5060 udp 11:20:21.450 180 185.61.68.c:5060 109.235.32.b:5060 udp 11:20:27.929 200 185.61.68.c:5060 109.235.32.b:5060 udp 11:20:28.106 ACK 109.235.34.a:39621 109.235.32.b:5060 tcp 11:20:30.625 BYE 185.61.68.c:5060 109.235.32.b:5060 udp 11:20:30.659 BYE 185.61.68.c:5060 109.235.32.b:5060 udp 11:20:30.663 BYE 109.235.32.b:5060 109.235.34.a:39621 tcp 11:20:30.664 200 109.235.34.a 39621 109.235.32.b:5060 tcp 11:20:30.664 200 109.235.32.b:5060 185.61.68.c:5060 udp What is missing are the forwarded replies, there is an ACK that matches the 401, but I have no clue what message was send to the UAC. Adding a sip_trace after REQINIT shows nothing extra. How (if possible) do I get all messages being send via the proxy? I could use sngrep/sipgrep, but that is an other process to monitor for failure.

2 4

allow inbound calls from any ip
by Logeshwaran G 21 Aug '17

21 Aug '17

Hi, I need to allow the inbound calls from any ip i kamailio. Input please Thanks & Kind Regards, Logeshwaran G

2 1

Take call counts from kamailio
by Logeshwaran G 21 Aug '17

21 Aug '17

Hi, How to take call counts from the kamailio? Thanks & Kind Regards, Logeshwaran G

2 12

sdp extra \r\n in multipart body
by Diego Nadares 21 Aug '17

21 Aug '17

Hi guys, I have the following multipart body Content-Type: multipart/mixed;boundary=uniqueBoundary Remote-Party-ID: <tel:1112222;phone-context=+54> Content-Length: 364 --uniqueBoundary Content-Type: application/sdp v=0 o=user1 53655765 2353687637 IN IP4 111.111.111.111 s=- c=IN IP4 111.111.111.111 t=0 0 m=audio 6000 RTP/AVP 8 *a=rtpmap:8 PCMA/8000* *--uniqueBoundary--* Content-Type: application/gtd Content-Disposition: signal;handling=optional IAM, CPC,20 GCI,885sdfasdf1123 --uniqueBoundary-- As you can see *--uniqueBoundary *before *a=rtpmap:8 PCMA/8000* has not an extra '\r\n'. Is this correct? Or should have it? The thing is that rtpengine is receiving sdp with no final '\r'n' from kamailio: Dump for 'offer' from 127.0.0.1:37388: { "sdp": "v=0#015#012o=user1 53655765 2353687637 IN IP4 111.111.111.111#015#012s=-#015#012c=IN IP4 111.111.111.111#015#012t=0 0#015#012m=audio 6000 RTP/AVP 8#015#012*a=rtpmap:8 PCMA/8000"*, "ICE": "remove", "direction": [ "pub", "priv" ], "replace": [ "origin", "session-connection" ], "transport-protocol": "RTP/AVP", "call-id": "1-17755(a)111.111.111.111", "received-from": [ "IP4", "111.111.111.111" ], "from-tag": "1", "command": "offer" } And the resulting sdp from rtpengine is this: --uniqueBoundary Content-Type: application/sdp v=0 o=user1 53655765 2353687637 IN IP4 172.16.213.16 s=- c=IN IP4 172.16.213.16 t=0 0 m=audio 35066 RTP/AVP 8 *a=rtpmap:8 PCMA/8000a=sendrecv* a=rtcp:35067 The problem is related with that '\r\n' but Is this a malformed message or a possible bug? Thanks in advance! Diego

2 1

Documenation of functions exported for KEMI
by Sebastian Damm 19 Aug '17

19 Aug '17

Hi, I was wondering whether there is some kind of documentation of all functions currently exported for KEMI by Kamailio modules. I know I can find them somehow via grep, but I could not find a page in the Kamailio wiki listing all of them. Is there any? Best Regards, Sebastian

2 1

Converting to kemi
by James Cloos 19 Aug '17

19 Aug '17

[I see I posted this to -dev instead of -users; fixing that -JimC] I'm working on converting some configs to kemi lua and have a couple of questions: First: Does KSR.pv.get() support multiple pseudos in a single call? The examples such as misc/examples/kemi/kamailio-basic-kemi-lua.lua make several calls, but also do things like KSR.pv.get("$T_branch_idx]") to avoid having the append "]" to the string. That makes me wonder why that file has code like this: KSR.dbg("new branch [".. KSR.pv.get("$T_branch_idx]") .. " to ".. KSR.pv.get("$ru") .. "\n"); instead of like: KSR.pv.get("[$T_branch_idx] to $ru\n") Second: Is it still the case that, eg, event_route[xhttp:request] cannot be in the kemi lua code? There was another. Perhaps I'll recall what it was... :( -JimC -- James Cloos <cloos(a)jhcloos.com> OpenPGP: 0x997A9F17ED7DAEA6

2 1

Kamailio_428 memory usage
by Evgeniy G. 18 Aug '17

18 Aug '17

Good afternoon! Can you please tell us how the Kamailio works with shared memory. The fact is that there was an accident in which Kamailio reported an error - little common memory. This happened a day later as I added the dialplan module. After that, I began to monitor the values: used_size real_used_size max_used_size free And I saw that I can not properly assess. I note that the server does not process calls, it is idle. After the server is started, the indicators are as follows: used_size: 63 real_used_size: 146 max_used_size: 146 free 106 After rebooting the module (for example, drouting, in which the data has not changed): used_size: 63 real_used_size: 193 max_used_size: 231 free 62 I found, a description, the reasons for the increase in memory after rebooting the modules, I quote "As suggested by Daniel, for carrierroute reload, we need the same amount of memory as it used to be. Block when reloading the table. " But why: 1. after increasing the memory the server frees only real_used_size 2. That that is specified as real_used_size on my observations can not be released in current of several days (probably weeks). 3.If in this state again to reboot the module (the same one that led to an increase in memory), the memory does not increase. Question: during what time should kamailio release the memory specified as real_used_size? Also what is considered as real_used_size? -- View this message in context: http://sip-router.1086192.n5.nabble.com/Kamailio-428-memory-usage-tp161327.… Sent from the Users mailing list archive at Nabble.com.

1 0

syslog CDR time format
by Yu Boot 18 Aug '17

18 Aug '17

Hello. Can't get 'date-time' format timestamps in syslog CDRs, while SQL CDRs are OK. According to acc module documentation: modparam("acc", "cdr_enable", 1) modparam("acc", "time_format", "%Y-%m-%d %H:%M:%S") modparam("acc", "time_mode", 3) modparam("acc", "cdr_facility", "LOG_DAEMON") syslog: NOTICE: acc [acc_cdr.c:352]: log_write_cdr(): start_time=1503068984.627; end_time=1503068991.250; duration=6.623; src_user=100; src_domain=a.b.c.d; src_ip=e.f.g.h; dst_ouser=200; dst_user=200; dst_domain=k.l.m.n

1 0

On the fly Change
by Logeshwaran G 18 Aug '17

18 Aug '17

Hi, Is there any possibilities of "on the fly" option in kamailio.cfg. For Example: I need to change my carrier IP address in the lcr_gw table, kamailio.cfg should takes it on the fly. Without restarting the kamailio. Thanks & Kind Regards, Logeshwaran G

4 18

Acc cdr duration
by yu＠yu-boot.ru 18 Aug '17

18 Aug '17

Hello. How to remove milliseconds from duration field in CDR? -- Отправлено из myMail для Android

1 0

dispatch_rpc
by Logeshwaran G 18 Aug '17

18 Aug '17

Hi, Is there any other module need to be added for xmlrpc for using dispatch_rpc(). Because I am getting an below error: 0(22485) ERROR: <core> [core/cfg.y:3274]: yyparse(): cfg. parser: failed to find command dispatch_rpc (params 0) 0(22485) CRITICAL: <core> [core/cfg.y:3414]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio_final.cfg, line 640, column 17: unknown command, missing loadmodule? I have loaded xmlrpc as well as xhttp modules. Do I need to enable some other modules as well?? Please Guide me. Thanks & Kind Regards, Logeshwaran G

1 0

SIP_DOMAIN in /etc/kamailio/kamctlrc
by Trent 17 Aug '17

17 Aug '17

Hello. I have not been able to find anything in the wiki, or documentation which covers this. The only item I have found is related to a module. >From the information, I have gleaned on this, it is clear an IP address, or a domain name is to be set here. What is not clear, is if the server is on multiple IP address whether they are on a WAN, LAN, or both, if multiple IP addresses are need, or can be added. I have a situation where I have a public WAN address, and a LAN address. Thanks a bunch!

1 0

kamailio restart:errors
by Uzma Anjum 17 Aug '17

17 Aug '17

Hi,I am new to kamailio. when i restart kamailio,I found errors on log.please help through this. 6(5633) ERROR: mymodule [my_mod.c:373]: outbound_call_process(): [outbound_call_process] URI : sip:11111111112@107.170.69.191:1008 received ip 183.82.115.246 in network ips:107.170.69.191:6090|c1|c2 6(5633) ERROR: mymodule [my_mod.c:379]: outbound_call_process(): [outbound_call_process] Request URI : 11111111112 6(5633) ERROR: mymodule [my_mod.c:395]: outbound_call_process(): outbound_call_process, obcalls privilges :select u.ob_call_privileges, a.phone_number, replace(a.name,'.','-') from account_info a, users u, subscriber s where a.id=u.account_id and u.account_id=s.account_id and u.extension=s.extension and s.username='dyan-dayan' 6(5633) ERROR: mymodule [my_mod.c:425]: outbound_call_process(): outbound_call_process query from user:dyan-dayan, phone_type:1 acc_name:dayan 6(5633) ERROR: mymodule [my_mod.c:501]: outbound_call_process(): [outbound_call_process]NEW URI : sip:ivr-1-dayan@107.170.69.191:6090 6(5633) ERROR: <core> [udp_server.c:448]: udp_rcv_loop(): recvfrom:Received encrypted packet len:4, v▒..

1 0

ClueCon Slides: Industrial Grade FreeSWITCH, Scaling, Balancing, High Availability, SIP, WebRTC
by Giovanni Maruzzelli 17 Aug '17

17 Aug '17

*Industrial Grade FreeSWITCH Scaling, Balancing and High Availability for SIP and WebRTC* Scaling your FreeSWITCH platform to serve a growing user base is a critical challenge. We'll go through the best techniques, practices, and implementations for Voice and Video Calls, Conferencing, WebRTC, SIP, Chatting, Presence and Instant Messaging Slides can be downloaded from http://opentelecom.it/cluecon2017.pdf Enjoy ClueCon! -giovanni -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18

4 4

Re: [SR-Users] DMQ Not Discovering all Nodes
by Charles Chance 16 Aug '17

16 Aug '17

Hello, Can you share the rest of your config (at least, any dmq modparams and the main route block)? Cheers, Charles On 16 Aug 2017 22:48, "John Petrini" <jpetrini(a)coredial.com> wrote: Hello, I've just started doing some testing with dmq but I'm having trouble getting it to discover all of the nodes in my cluster. There are 7 kamailio instances with the following in their configs: Example from the notification server (10.0.10.211): listen=udp:10.0.10.211:5080 loadmodule "dmq.so" # ----- dmq params ----- modparam("dmq", "server_address", "sip:10.0.10.211:5080") modparam("dmq", "notification_address", "sip:10.0.10.211:5080") Example from one of the nodes (10.0.10.216). (Only the listen and server_address is changed to match the local ip of each node.) listen=udp:10.0.10.216:5080 loadmodule "dmq.so" # ----- dmq params ----- modparam("dmq", "server_address", "sip:10.0.10.216:5080"); modparam("dmq", "notification_address", "sip:10.0.10.211:5080") Output from kamcmd dmq.list_nodes { host: 10.0.10.211 port: 5080 resolved_ip: 10.0.10.211 status: 2 last_notification: 0 local: 0 } { host: 10.0.10.216 port: 5080 resolved_ip: 10.0.10.216 status: 2 last_notification: 0 local: 1 } I'm expecting to see all 7 nodes in the output but only see the local node and the notification node. I imagine I could use an SRV record for the notication address to add all of the nodes but I thought the notification node was supposed to share the rest of the nodes in the cluster. Can anyone explain what it is I'm doing wrong? Thank You, John Petrini _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Sipcentric Ltd. Company registered in England & Wales no. 7365592. Registered office: Faraday Wharf, Innovation Birmingham Campus, Holt Street, Birmingham Science Park, Birmingham B7 4BB.

2 1

DMQ Not Discovering all Nodes
by John Petrini 16 Aug '17

16 Aug '17

Hello, I've just started doing some testing with dmq but I'm having trouble getting it to discover all of the nodes in my cluster. There are 7 kamailio instances with the following in their configs: Example from the notification server (10.0.10.211): listen=udp:10.0.10.211:5080 loadmodule "dmq.so" # ----- dmq params ----- modparam("dmq", "server_address", "sip:10.0.10.211:5080") modparam("dmq", "notification_address", "sip:10.0.10.211:5080") Example from one of the nodes (10.0.10.216). (Only the listen and server_address is changed to match the local ip of each node.) listen=udp:10.0.10.216:5080 loadmodule "dmq.so" # ----- dmq params ----- modparam("dmq", "server_address", "sip:10.0.10.216:5080"); modparam("dmq", "notification_address", "sip:10.0.10.211:5080") Output from kamcmd dmq.list_nodes { host: 10.0.10.211 port: 5080 resolved_ip: 10.0.10.211 status: 2 last_notification: 0 local: 0 } { host: 10.0.10.216 port: 5080 resolved_ip: 10.0.10.216 status: 2 last_notification: 0 local: 1 } I'm expecting to see all 7 nodes in the output but only see the local node and the notification node. I imagine I could use an SRV record for the notication address to add all of the nodes but I thought the notification node was supposed to share the rest of the nodes in the cluster. Can anyone explain what it is I'm doing wrong? Thank You, John Petrini

1 0

Kamailio not forces TLS socket
by Yuriy Gorlichenko 16 Aug '17

16 Aug '17

Hi! I using uac for send MESSAGE request from kamailio to remote destination endpoint Before sending i setting up $fs seudovariable as $fs = "tls:12.3.4.5:443"; kamailio listeinig this interface and succesfully receives requests and sending responses on it When i call uac_req_send();There is error in log file: ERROR: tm [ut.h:327]: uri2dst2(): no corresponding socket found for "2.3.26.11" af 2 (udp:2.3.26.11:57050) ERROR: tm [uac.c:272]: t_uac_prepare(): t_uac: no socket found If I setting up udp interface for listening then kamailio ognores my $fs set up and sending message through udp socket. kamailio version 4.4.6 Now i trying to roll back to 4.4.4 version but not sure if it will be helpfull.

1 0

Permission denied for /var/run/kamailio
by Ryan Wagoner 16 Aug '17

16 Aug '17

I'm running Kamailio 4.4.6-2.1 installed from RPM on CentOS 7, where /var/run is a tmpfs (by default). After every reboot Kamailio fails to start with the following error: Aug 16 00:27:39 sbc1 /usr/sbin/kamailio[7135]: ERROR: mi_fifo [fifo_fnc.c:72]: mi_init_fifo_server(): Can't create FIFO: Permission denied (mode=432) I have the following line in my kamailio.cfg modparam("mi_fifo", "fifo_name", "/var/run/kamailio/kamailio_fifo") At this point the /var/run/kamailio directory looks like this drwx------. 2 root root 80 Aug 16 00:30 . drwxr-xr-x. 26 root root 880 Aug 16 00:30 .. srw-------. 1 kamailio kamailio 0 Aug 16 00:30 kamailio_ctl If I run "chown kamailio /var/run/kamailio" and "systemctl start kamailio" I am good to go. Am I missing something in my config? It appears Kamailio is creating the /var/run/kamailio folder as root, writing the kamailio_ctl file, dropping root privileges, and then trying to write the kamailio_fifo file. Thanks, Ryan

2 1

sipcapture/HEP msg sizes>1500
by Daniel Tryba 16 Aug '17

16 Aug '17

While looking into HOMER I saw that large INVITEs got truncated in the webinterface, looking at the HEP traffic these (UDP) messages got fragmented in transport (asterisk/res_hep) and I started to think these were not getting reassembled correctly. Only to find out the database schemas have set the msg size set to varchar(1500). Is there any good reason for not changing this to a bigger default?

1 0

RTP loops
by Mauro Costantini 16 Aug '17

16 Aug '17

Hi everyone, we have an installation with: - 1 kamailio instance - with 2 interface, one with public and one with private IP (for local communication to the db and asterisk ). We use keepalived service for failover and so, every interface has 2 IPs: the real one and the “virtual” one for the keepalived service - some asterisk instances for transcoding and billing - with one interface with private IP - 2 RTP (rtpengine) instances - with 2 interfaces, one with public and one with private IP A few times happens a loop on the RTPengine and we could see a lot of lines like this in the log: “Too many packets in UDP receive queue (more than 50), aborting loop. Dropped packets possible” This loop doesn't stop until all the resources are exhausted and it results in the freeze of the machine. We don’t understand the reason why these loops are generated. We have 2 cases: 1) we send an INVITE to the CPE (a Bria on iOS) The CPE answers with a "100 Trying", a "180 Ringing" and a few seconds later sends us 3 packets of "200 ok" with SDP like this: SIP/2.0 200 OK Via: SIP/2.0/UDP 178.250.x.y;branch=z9hG4bKb842.29952898c30d6d06b75b916618357ccd.2 Via: SIP/2.0/UDP 192.168.x.y:5090;rport=5090;branch=z9hG4bK7fd100fb Record-Route: <sip:178.250.x.y;r2=on;lr=on;ftag=as26b82ac7;did=cb5.0ce;nat=yes> Record-Route: <sip:192.168.x.z;r2=on;lr=on;ftag=as26b82ac7;did=cb5.0ce;nat=yes> Contact: <sip:6051018****@158.148.x.y:3180;rinstance=ffdff5e0142dfd2d> To: <sip:6051018****@192.168.x.z>;tag=29859d3c From: <sip:051438****@192.168.x.y>;tag=as26b82ac7 Call-ID: 698200851849894c5874186163abe668@192.168.x.y:5090 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, BYE, REFER, INFO, NOTIFY, OPTIONS, UPDATE, PRACK, MESSAGE, SUBSCRIBE Content-Type: application/sdp Supported: replaces User-Agent: Bria iOS release 3.9.7 stamp 38887.38893 Content-Length: 230 v=0 o=- 1085831961 3 IN IP4 100.81.231.195 s=Cpc session c=IN IP4 100.81.231.195 t=0 0 m=audio 65454 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv Could the IP of the media part (100.81.231.195 in the CGNAT 100.64.0.0/10) be the cause of the loop? If that’s the case, how can we prevent the loop? 2) in the second case the CPE is an Avaya The Avaya send us an INVITE. (Also in this case in the SDP there is a private IP) INVITE sip:04818****@178.250.x.y SIP/2.0 From: <sip:0432415453@185.55.x.y>;tag=4500808-c0a8077d-13c4-50022-313081f-200d750c-313081f To: <sip:04818****@178.250.x.y> Call-ID: 44e8f20-c0a8077d-13c4-50022-313081f-31e99586-313081f CSeq: 1 INVITE Via: SIP/2.0/UDP 185.55.x.y:5060;rport;branch=z9hG4bK-313081f-257bc48-1472e4a5 Privacy: none Max-Forwards: 70 User-Agent: OfficeServ 7200 Contact: <sip:0432415453@185.55.z.y:5060;transport=udp> Allow: REGISTER,INVITE,ACK,BYE,REFER,NOTIFY,CANCEL,INFO,OPTIONS,PRACK,SUBSCRIBE,UPDATE Supported: 100rel Content-Type: application/sdp Content-Length: 289 v=0 o=SAMSUNG_SIP_GATEWAY 39304265 0 IN IP4 185.55.x.y s=SIP_CALL c=IN IP4 192.168.7.126 t=0 0 m=audio 30012 RTP/AVP 18 8 0 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv We reply with a “183 Session Progress” and the Avaya send US an “INFO” packet (dtmf) we replay with a “200 OK”; and then, after another “INFO” packet we replay with a “200 OK” with SDP. After that the Avaya replies with an ACK 2017/08/02 10:17:32.101121 185.55.x.y:5060 -> 178.250.x.y:5060 ACK sip:04818****@178.250.x.y SIP/2.0 From: <sip:0432415453@185.55.x.y>;tag=4500808-c0a8077d-13c4-50022-313081f-200d750c-313081f To: <sip:04818****@178.250.x.y>;tag=as783106e4 Call-ID: 44e8f20-c0a8077d-13c4-50022-313081f-31e99586-313081f CSeq: 1 ACK Via: SIP/2.0/UDP 185.55.x.y:5060;rport;branch=z9hG4bK-3130831-257ffea-84906e4 Max-Forwards: 70 User-Agent: OfficeServ 7200 Route: <sip:178.250.x.y;lr=on;r2=on;ftag=4500808-c0a8077d-13c4-50022-313081f-200d750c-313081f;did=adc.2f61> Route: <sip:192.168.x.y;lr=on;r2=on;ftag=4500808-c0a8077d-13c4-50022-313081f-200d750c-313081f;did=adc.2f61> Contact: <sip:043241****@185.55.x.y:5060;transport=udp> Content-Length: 0 And after that the chaos… Has anyone had a problem like this? Can somebody help us in order to prevent this loop? In the attached images you can see: [0] a call, our machines IPs, an ACK from public interface of our SBC to its private one; and than a lot of ACKs from the keepalived IP to the private IP of our SBC. [1] one of these strange ACKs, they are all different in content, but similar in structure: multiple Record-Route / Via, truncating too long message; [2] a "never ending" BYEs. After modifying the configuration adding some "listen=IP_ADDR:PORT" (and of course excluding the keepalive IP) we had no loops but is this enough? Why Kamailio generated the loops? Is there something else we should review in our config? Any advice would help a lot. Thanks. [0] https://goo.gl/rWh5db [1] https://goo.gl/MNShq9 [2] https://goo.gl/6mxepG

1 0

UAC reg_random_delay issue
by Ivo Vastert 16 Aug '17

16 Aug '17

Hi, I'm using the UAC module in Kamailio to setup about 500 outbound registration towards a single host. The issue I'm currently experiencing is that when we start Kamailio all this 500 registrations are initiated at once towards the host which can't handle all the requests at the same time. Therefor I tried to spread the registrations using the reg_random_delay function: modparam("uac", "reg_random_delay", 60) I would expect that this would spread the registrations over about a minute but in reality the registrations are spread as follow: Starttime / Registrations 21:59:14 151 21:59:15 166 21:59:16 127 21:59:18 60 21:59:22 35 So Kamailio is started at 21:59:14 and initiates 151 registrations, 1 second after start another 166 2 seconds after start another 127 4 seconds after start another 60 8 seconds after start the last 35 In reality it looks like some increment is applied instead of calculating a random starting offset for each outbound registration? Is there any solutions to this issue available? Best regards, Ivo Vastert

2 6

rtpengine : question about RTP/RTCP multiplexing (RFC 5761) and demultiplexing
by Abdoul Osséni 16 Aug '17

16 Aug '17

Hello All, I read rtpengine can multiplex and demultiplex RTP/RTCP. Cf.https://github.com/sipwise/rtpengine Example: UAC -> Kamailio + rtpengine -> Asterisk UAC and Asterisk don't support RFC 5761. Only rtpengine does. Does this mean that uac must be compliance with RFC 5761 if I want multiplex and demultiplex RTP/RTCP between UAC and rtpengine? Thanks. Abdoul.

2 2

log file
by Uzma Anjum 16 Aug '17

16 Aug '17

the log statements written in acc.c accounting module source code are not entered into syslog file.what can be wrong?

1 0

functions in accounting module
by Uzma Anjum 16 Aug '17

16 Aug '17

when the call answered,which methods are executed from acc.c source code and which method is inserted into database while call answer and call terminate.

1 0

Kamailio not listening on TLS port (5061)
by Deepak Jaiswal 16 Aug '17

16 Aug '17

Hi, I am new to kamailio and unable to debug the issue. Kamailio is not listening on TLS port. Could you please suggest some workaround? [root@uk-vap-002 ~]# ss -tnlp | grep kamailio LISTEN 0 1024 127.0.0.1:5060 *:* users:(("kamailio",29400,18)) LISTEN 0 1024 xxx.xxx.xxx.134:5060 *:* users:(("kamailio",29400,17)) LISTEN 0 1024 xxx.xxx.xxx.135:5060 *:* users:(("kamailio",29400,16)) LISTEN 0 1024 xxx.xxx.xxx:5060 *:* users:(("kamailio",29400,15)) Kamailio.cfg config done: listen=tls:xxx.xxx.xxx.133:5061 listen=tls:xxx.xxx.xxx.135:5061 listen=tls:xxx.xxx.xxx.134:5061 listen=tls:127.0.0.1:5061 2. #!ifdef WITH_TLS enable_tls=yes 3. #!ifdef WITH_TLS loadmodule "tls.so" #!endif 4. #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg") #!endif Regards Deepak Jaiswal

2 1

Siremis Install: Unable to write in the session save path (CentOS 6)
by Trent 16 Aug '17

16 Aug '17

Hello. Doing a new install, and I am in the stage of accessing the web interface for Sermis which is located at /var/www/html/siremis Full error: Unable to write in the session save path [/var/www/html/siremis/session]Unable to write in the session save path [/var/www/html/siremis/session] I would think this is a single access problem, but I have checked ownership in httpd.conf, and the files, and directories. All have the apache ownership assigned to them. I have even given the "session" folder access of 777 with still no change in the error. -rw-r--r--. 1 apache apache 10176 May 18 05:18 ChangeLog drwxr-xr-x. 4 apache apache 4096 May 18 05:18 misc drwxr-xr-x. 7 apache apache 4096 May 18 05:18 openbiz -rw-r--r--. 1 apache apache 1019 May 18 05:18 README.md drwxr-xr-x. 14 apache apache 4096 May 18 05:18 siremis In addition, there is nothing about this appearing in the httpd error_log, nor access_log for an access problem. Any suggestions? Thanks!

3 3

UAC random_delay not that random
by Ivo Vastert 15 Aug '17

15 Aug '17

Hi, I'm using the UAC module in Kamailio to setup about 500 outbound registration towards a single host. The issue I'm currently experiencing is that when we start Kamailio all this 500 registrations are initiated at once towards the host which can't handle all the requests at the same time. Therefor I tried to spread the registrations using the reg_random_delay function: modparam("uac", "reg_random_delay", 60) I would expect that this would spread the registrations over about a minute but in reality the numbers are sprad as follow: 2017/08/09 21:59:14: 301 2017/08/09 21:59:15: 332 2017/08/09 21:59:16: 253 2017/08/09 21:59:18: 120 2017/08/09 21:59:22: 69

1 0

KSR.pv.get("$(hdr(P-Asserted-Identity){uri.user})") returns unable to get pv value
by Muhammad Zaka 15 Aug '17

15 Aug '17

Hi We are experiencing issue with python script KSR.pv.get using branch 5.0.2 as it fails "unable to get pv value" Example Code: if KSR.pv.get("$ai"): fromid = KSR.pv.get("$(hdr(P-Asserted-Identity){uri.user})") Error: Aug 14 18:38:23 sbc99-kam-dev /usr/sbin/kamailio[30459]: PYT {INVITE}: ERROR: app_python [apy_kemi.c:637]: sr_apy_kemi_f_pv_get(): unable to get pv value for [$(hdr(P-Asserted-Identity){uri.user})] Is anybody have any idea how to workaround? Many Thanks Regards Muhammad Zaka

2 4

Kamailio 4.3.4: receive_msg(): no via found in reply
by Grant Bagdasarian 15 Aug '17

15 Aug '17

Hello, We're testing kamailio 4.3.4 with two kamailio's communicating with each other. One of them is a gateway with rtpengine at the carrier side, the other is a basic sip router. When calling a destination and cancelling the INVITE a 487 is received from the carrier and directly ACK'd by the first kamailio and in turn is forwarded to the second. But the 487 response received by the 2nd kamailio doesn't have a Via header which results in kamailio printing the error: receive_msg(): no via found in reply. Any ideas why the forwarded 487 doesn't contain a Via header and how to fix this? Note: IP's are changed. Carrier: 10.0.0.1 Kamailio 1: 10.14.0.1 Kamailio 2: 10.14.0.2 U 10.0.0.1:5060 -> 10.14.0.1:5060 SIP/2.0 487 Request Cancelled. Via: SIP/2.0/UDP 10.14.0.1;branch=z9hG4bK8e28.642c68037d059634308d4be8fe3f0cdc.0. From: "+31612345678" <sip:+31612345678@10.14.0.3:5060>;tag=109ac722. To: <sip:003112345678@10.14.0.3:5060>;tag=sbcsipuas_1_C22196_20160314080307791_b59sb10. Call-ID: 78100MjY3OGExZmMyODU2MzAyMzM5ODA3MmNhMDdlMzg0MzU. Contact: <sip:10.0.0.1:5060>. CSeq: 2 INVITE. Server: sbc_5. Content-Length: 0. . # U 10.14.0.1:5060 -> 10.0.0.1:5060 ACK sip:003112345678@10.0.0.1:5060 SIP/2.0. Via: SIP/2.0/UDP 10.14.0.1;branch=z9hG4bK8e28.642c68037d059634308d4be8fe3f0cdc.0. Max-Forwards: 67. To: <sip:003112345678@10.14.0.3:5060>;tag=sbcsipuas_1_C22196_20160314080307791_b59sb10. From: "+31612345678"<sip:+31612345678@10.14.0.3:5060>;tag=109ac722. Call-ID: 78100MjY3OGExZmMyODU2MzAyMzM5ODA3MmNhMDdlMzg0MzU. CSeq: 2 ACK. Content-Length: 0. . # U 10.14.0.1:5060 -> 10.14.0.2:5060 SIP/2.0 487 Request Cancelled. From: "+31612345678" <sip:+31612345678@10.14.0.3:5060>;tag=109ac722. To: <sip:003112345678@10.14.0.3:5060>;tag=sbcsipuas_1_C22196_20160314080307791_b59sb10. Call-ID: 78100MjY3OGExZmMyODU2MzAyMzM5ODA3MmNhMDdlMzg0MzU. Contact: <sip:10.0.0.1:5060>. CSeq: 2 INVITE. Server: sbc_5. Content-Length: 0.

4 16

Reinvite, rewrite reply SDP?
by Yu Boot 15 Aug '17

15 Aug '17

Finally I find a root of a problem with my T.38 reinvites. A->(NAT)->Kami->PSTN When NATted caller does reinvite, and remote side answers with 200 OK+SDP, Kami (why?) doesn't rewrite SDP for its own IP (so NATted caller can receive RTP) and leaves remote's IP as media address, so of course there is no RTP/UDPTL. All other NAT related stuff works fine. Even if remote side sends me T.38 reinvite first, everything works properly. Config file is default from Kami sources. Any ideas?

2 2

RTPEngine recording
by Alex Balashov 14 Aug '17

14 Aug '17

I have gathered that RTPEngine has recently, or perhaps some time ago, evolved a recording feature set: https://kamailio.org/docs/modules/5.0.x/modules/rtpengine.html#rtpengine.f.… Does anyone have any experience using it with Kamailio? How does it work? Any gotchas or pitfalls? It looks like recording is done via a separate daemon specifically for that purpose. Does it emit directly in a playable format, or more or less just dump the raw, RTP-encapsulated frames? -- 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/

3 8

DNS Geo Location Kamailio servers, registration and routing
by Sadeesh Radhakrishnan 14 Aug '17

14 Aug '17

We have kamailio servers that do DNS-Geo-Location on different locations, using same domain name. So the users from a country will be preferred to his nearby server by DNS routing. Similarly users from different countries will be registering to their nearby kamailio sip servers. All the servers share a common user location table on database. Now the users are getting registered fine, with no issue. We want the users at one location should be able to call user@other-location using the user location contact information. When we tried, the INVITE is sent on common domain name and that reverse resolves dns to same server and gets a 404 not found. I need some suggestion / ideas in order to make it happen. Thanks

2 1

set digest mime
by wsotest.512 14 Aug '17

14 Aug '17

2 1

Inconsistency in sems v1.6 installation: sems.conf is missing
by leslie mekiyess 14 Aug '17

14 Aug '17

For some reason, the installation creates the etc directory under the /usr/local/etc/sems/ : make[2]: Entering directory `/usr/src/sems-1.6.0/apps/conf_auth' mkdir -p **/usr/local/etc/sems/**etc**/** installing conf_auth.conf make[2]: Leaving directory `/usr/src/sems-1.6.0/apps/conf_auth' Then at the make install ends the following is printed: *** install complete. Run SEMS with *** *** /usr/local/sbin/sems -f /**usr/local/etc/sems/sems.conf** At bottom libe, while following the spain document instructions, I figured out that the sems.conf wasn;t created at the /usr/local/etc/sems/ path. I can create a new one or even copy this one found : /usr/src/sems-1.6.0/apps/mobile_push/load_test/sems_cfg/core/etc/sems.conf but I am afraid of further files/directory inconsistencies. Any assistance is very appreciated.

2 1

acc billsec
by Aidar Kamalov 14 Aug '17

14 Aug '17

Hello, is there any way to log in cdr duration and billsec? -- Aydar A. Kamalov

3 3

UAC RPC Commands problem
by Volkan Oransoy 14 Aug '17

14 Aug '17

Hi, I use Kamailio 5.0. I try to run RPC commands for UAC module but some of them doesn’t work like documented. kamcmd uac.reg_dump kamcmd uac.reg_reload These commands work as expected. But commands with extra arguments give error. kamcmd> uac.reg_refresh 908508850000 error: 400 - Invalid Parameters kamcmd> uac.reg_info l_uuid 908508850000 error: 404 - Record not found Do I miss something? Thanks, /Volkan

2 3

Carrierroute, fallback, reinvites
by yu＠yu-boot.ru 12 Aug '17

12 Aug '17

Hello. Trying to implement fallback route for carrierroute and got sick to make reinvites working correctly. Until fallback route triggers, everything is OK. Anyone has working config for that purposes (NAT traversal, reinvites, carrierroute ), please? And abstract question - is it generally a good idea to implement a simple softswitch (not PBX like Asterisk) like Yate or Freeswitch with Kami? Now I just want to learn Kami and learn SIP deeper in general.

2 1

Route calls from Kamailio to Asterisk behind NAT
by vs＠cell.tel 12 Aug '17

12 Aug '17

3 5

Re: [SR-Users] [OpenSIPS-Users] ClueCon Slides: Industrial Grade FreeSWITCH, Scaling, Balancing, High Availability, SIP, WebRTC
by Abdul Basit 11 Aug '17

11 Aug '17

nice presentation!!! -- regards, abdul basit | p: +92 32 1416 4196 | o: +92 30 0841 1445 On 10 August 2017 at 23:12, Giovanni Maruzzelli <gmaruzz(a)gmail.com> wrote: > *Industrial Grade FreeSWITCH Scaling, Balancing and High Availability for > SIP and WebRTC* > > Scaling your FreeSWITCH platform to serve a growing user base is a > critical challenge. We'll go through the best techniques, practices, and > implementations for Voice and Video Calls, Conferencing, WebRTC, SIP, > Chatting, Presence and Instant Messaging > Slides can also be downloaded from http://opentelecom.it/cluecon2017.pdf > > Enjoy ClueCon! > > -giovanni > > -- > > Sincerely, > > Giovanni Maruzzelli > OpenTelecom.IT > cell: +39 347 266 56 18 > > _______________________________________________ > Users mailing list > Users(a)lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > >

1 0

Dispatcher call id hash for server side invite
by Ryan Wagoner 11 Aug '17

11 Aug '17

For a call invite from a phone to an load balanced asterisk server farm I can use ds_select_dst with hash over callid for the algorithm. What I don't understand is what happens for a server side invite. Say user A calls user B. The server will send and invite to user B's device. User B's device will reply and ds_select_dst won't have the call id hash so it will choose a random server, which might not be the server sending the invite. How do I mark these server side invites so the call hash is known by kamailio? Or am I thinking about this the wrong way? Thanks, Ryan

3 6

Issue in Dialplan Module
by Logeshwaran G 10 Aug '17

10 Aug '17

Hi, I am facing an Problem in Dialplan Module. Scenario: ======== I am Dialling 00091xxxxxxxxxx number, I need to remove only the prefix 000, then the call placement will be 91xxxxxxxxxx. My dialplan rule: ============== MariaDB [kamailio]> select * from dialplan; +----+------+----+----------+-----------+-----------+-----------+----------+-------+ | id | dpid | pr | match_op | match_exp | match_len | subst_exp | repl_exp | attrs | +----+------+----+----------+-----------+-----------+-----------+----------+-------+ | 1 | 1 | 1 | 1 | ^000$ | 0 | ^000$ | | | +----+------+----+----------+-----------+-----------+-----------+----------+-------+ But the prefix 000 is not replaced. Please Guide me to Resolve this issue. Do I need to change somet rules?? Additional info: ============= But, With additional to the above dialplan data rule I used the below rule. +----+------+----+----------+-----------+-----------+-----------+----------+-------+ | id | dpid | pr | match_op | match_exp | match_len | subst_exp | repl_exp | attrs | +----+------+----+----------+-----------+-----------+-----------+----------+-------+ | 1 | 1 | 1 | 1 | ^000$ | 0 | ^000$ | | | 111 +----+------+----+----------+-----------+-----------+-----------+----------+-------+ The above rule replacing only the 000, If we Dial 000xxxx its omitting, While Dialing 000 its replacing that with 111. Thanks & Regards, Logeshwaran

4 23

BUG: invalid branch
by Yasin CANER 10 Aug '17

10 Aug '17

2 1

Kamailio: how to route RTP media directly to endpoint
by wsotest.512 10 Aug '17

10 Aug '17

Hi all, We have usual config Kamailio + Asterisk where Kamailio play as sip and rtp proxy. Kamailio have public IP, asterisk - no. All calls between clients now going like that: UserA ---sip--> Kamailio --> Asterisk --> UserB \-rtp--> Kamailio (rtpproxy) --> Asterisk --> UserB All clients of course from Internet and behind Nat. Main problem is amount of traffic going through Kamailio and Asterisk. We need to pay for every additional GB behind limit in tariff plan to hosting provider. So we decided to try route all rtp traffic between users directly. UserA ---sip--> Kamailio --> Asterisk --> UserB \-rtp--> --> --> UserB Is it possible at all? Maybe someone already did it . -- BR, Alex

3 2

IMPORTANT: acc module - SQL define removed and diameter code relocated
by Daniel-Constantin Mierla 09 Aug '17

09 Aug '17

Hello, I just pushed a bit of clean up to acc module, therefore be aware of following changes: 1) the define conditions on SQL_ACC were removed -- this was enabled for more than 10 years and only made the code look complex and hard to follow up its logic. 2) the code related to DIAMETER accounting was relocated to acc_diameter (new) module. It was a consistent size of code that was not enabled for sooo... long. It is now a dedicated module, similar to acc_radius. The diameter accounting code, even a new module now, is in the same stage, compiling but not tested, in pair with auth_diameter module, it may work, but very likely not. In summary, what's important for those using the acc module -- it offers the same functionality as it was enabled by default in the past 10 years or more, only the unused code was relocated. It offers the functionality of writing accounting records to syslog and sql databases. The acc module is now slimmer, only with the code that it needs, therefore easier to maintain and enhance for the future. Hopefully, there was no side effect with this update -- anyhow, if you find any issue, just open a bug report on github project. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - www.kamailioworld.com

1 0

Kamailio: how to route RTP media directly to endpoint
by Aleksey Hodin 09 Aug '17

09 Aug '17

Hi all, We have usual config Kamailio + Asterisk where Kamailio play as sip and rtp proxy. Kamailio have public IP, asterisk - no. All calls between clients now going like that: UserA ---sip--> Kamailio --> Asterisk --> UserB \-rtp--> Kamailio (rtpproxy) --> Asterisk --> UserB All clients of course from Internet and behind Nat. Main problem is amount of traffic going through Kamailio and Asterisk. We need to pay for every additional GB behind limit in tariff plan to hosting provider. So we decided to try route all rtp traffic between users directly. UserA ---sip--> Kamailio --> Asterisk --> UserB \-rtp--> --> --> UserB Is it possible at all? Maybe someone already did it ... -- BR, Alex

1 0

Registrar limit
by Yu Boot 09 Aug '17

09 Aug '17

How to limit one user login with one AOR (IP) and always use IP from last registration attempt?

2 1

RFC: Moving misc/obsolete/ to its own git repository
by Daniel-Constantin Mierla 09 Aug '17

09 Aug '17

Hello, trying to make the git repository a bit slimmer and avoid unuseful indexing of unused modules, I am considering to move misc/obsolete to its own github repository (planned to be named kamailio-obsolete). The initial idea was to keep there modules that were duplicated or no longer maintained, just in case someone will come later and wants to revive it. However, iirc, it is more than 4 years and nobody touched them, so very likely it requires a bit of work to revive any of them, so a fresh re-import will be the same from an external repository. If anyone still wants to keep them in the main kamailio repository, respond on the mailing list and present the benefits. If not, I will do the operation in the next several days. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - www.kamailioworld.com

1 1

"kamctl restart" fails
by Yu Boot 09 Aug '17

09 Aug '17

# kamctl restart INFO: Stopping Kamailio : INFO: stopped INFO: Starting Kamailio : 2292 pts/0 S+ 0:00 /bin/egrep kamailio -rw-r--r--. 1 kamailio kamailio 5 Aug 8 11:15 /var/run/kamailio.pid INFO: PID file exists (/var/run/kamailio.pid)! Kamailio already running? "service kamailio restart" works fine. What's wrong? Centos 6.9 x64, Kamailio 4.3 installed from official repo.

3 3

how to remove a line in sdp
by Sunil More 09 Aug '17

09 Aug '17

Hello All, I have the following SDP coming in from a carrier and would like to remove the following line. Cant figure how to go about it. line to remove : *a=cdsc: 1 image udptl t38* sdp is as below Content-Type: application/sdp. Content-Length: 293. . v=0. o=hiQ9200 5859020170708143552 1175715965 IN IP4 X.Y.Z.A. s=Phone Call via hiQ9200 SIPCA. c=IN IP4 X.Y.Z.A. t=0 0. m=audio 16648 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sqn: 0. *a=cdsc: 1 image udptl t38.* a=sendrecv. a=ptime:20. This reply comes from a upstream carrier in session progress and 200ok causing one-way audio for me. Thanking You, Sunil More Ph : 9503338275

2 1

Billing plan
by Logeshwaran G 09 Aug '17

09 Aug '17

Hi, Is there any billing application or modules to use with kamailio , for example if i want to check the credits of the user on every passing minute , and when he's out of credits i stop the call. Thanks & Kind Regards, Logeshwaran G

2 1

sqlops error
by Logeshwaran G 09 Aug '17

09 Aug '17

modparam("sqlops","sqlcon",DBURL) Whether the above modparam implementation is wrong ? I am not sure Its saying some error as 0(2519) ERROR: sqlops [sql_api.c:669]: sql_parse_param(): invalid sqlops parameter

3 5

Core crash in del_lump() "offset exceeds message size" when using subst() inside tm:local-request route.
by Cody Herzog 08 Aug '17

08 Aug '17

Hello. Inside the tm:local-request route, I am trying to modify the body of outgoing NOTIFY messages using the subst_body() function. Things seem to work fine most of the time. The NOTIFY messages are correctly modified. However, I will very rarely hit a crash. I can force the crash to happen more quickly by stress testing with thousands of NOTIFY messages being modified. Is it safe to call subst() or subst_body() inside the tm:local-request route? If not, is there another way I can modify the outgoing NOTIFY messages safely? Note that I am also calling append_hf() inside the same tm:local-request route, but that has been working for a very long time without causing any problems. Could there be some kind of bad interaction between append_hf() and subst_body()? Strangely, it seems like using subst() makes the crash more likely to happen. For some reason, subst_body() seems to be more robust. Here are the logs I see: --------- Jul 28 10:57:40 SIPCOMM-VEGAS-TEST /usr/local/sbin/kamailio[2476]: CRITICAL: <core> [data_lump.c:292]: del_lump(): offset exceeds message size (1625 > 1194) aborting... Jul 28 10:57:43 SIPCOMM-VEGAS-TEST /usr/local/sbin/kamailio[2481]: CRITICAL: <core> [pass_fd.c:275]: receive_fd(): EOF on 53 Jul 28 10:57:44 SIPCOMM-VEGAS-TEST /usr/local/sbin/kamailio[2367]: ALERT: <core> [main.c:731]: handle_sigs(): child process 2476 exited by a signal 6 Jul 28 10:57:44 SIPCOMM-VEGAS-TEST /usr/local/sbin/kamailio[2367]: ALERT: <core> [main.c:734]: handle_sigs(): core was generated Jul 28 10:57:44 SIPCOMM-VEGAS-TEST /usr/local/sbin/kamailio[2367]: INFO: <core> [main.c:756]: handle_sigs(): terminating due to SIGCHLD --------- Here is a GDB back trace from the core dump: --------- #0 0x00007fd91a567c37 in __GI_raise (sig=sig@entry=6) at ../nptl/sysdeps/unix/sysv/linux/raise.c:56 #1 0x00007fd91a56b028 in __GI_abort () at abort.c:89 #2 0x000000000044e981 in del_lump (msg=0x7fd917c5ed60 <lreq>, offset=1625, len=20, type=HDR_OTHER_T) at data_lump.c:293 #3 0x00007fd91676297f in subst_f (msg=0x7fd917c5ed60 <lreq>, subst=0x7fd919e5ccc0 "\320\356\345\031\331\177", ignored=0x0) at textops.c:681 #4 0x000000000041e1dd in do_action (h=0x7ffd364b87c0, a=0x7fd919e9b438, msg=0x7fd917c5ed60 <lreq>) at action.c:1059 #5 0x000000000042a917 in run_actions (h=0x7ffd364b87c0, a=0x7fd919e9b438, msg=0x7fd917c5ed60 <lreq>) at action.c:1548 #6 0x000000000042af94 in run_actions_safe (h=0x7ffd364ba5b0, a=0x7fd919e9b438, msg=0x7fd917c5ed60 <lreq>) at action.c:1613 #7 0x0000000000549972 in rval_get_int (h=0x7ffd364ba5b0, msg=0x7fd917c5ed60 <lreq>, i=0x7ffd364b8b58, rv=0x7fd919e9ced8, cache=0x0) at rvalue.c:912 #8 0x000000000054dc7c in rval_expr_eval_int (h=0x7ffd364ba5b0, msg=0x7fd917c5ed60 <lreq>, res=0x7ffd364b8b58, rve=0x7fd919e9ced0) at rvalue.c:1910 #9 0x000000000041dc52 in do_action (h=0x7ffd364ba5b0, a=0x7fd919e9d990, msg=0x7fd917c5ed60 <lreq>) at action.c:1029 #10 0x000000000042a917 in run_actions (h=0x7ffd364ba5b0, a=0x7fd919e9d990, msg=0x7fd917c5ed60 <lreq>) at action.c:1548 #11 0x000000000041e0c4 in do_action (h=0x7ffd364ba5b0, a=0x7fd919e9dad8, msg=0x7fd917c5ed60 <lreq>) at action.c:1044 #12 0x000000000042a917 in run_actions (h=0x7ffd364ba5b0, a=0x7fd919e973b8, msg=0x7fd917c5ed60 <lreq>) at action.c:1548 #13 0x000000000041ab46 in do_action (h=0x7ffd364ba5b0, a=0x7fd919e95d70, msg=0x7fd917c5ed60 <lreq>) at action.c:677 #14 0x000000000042a917 in run_actions (h=0x7ffd364ba5b0, a=0x7fd919e8f330, msg=0x7fd917c5ed60 <lreq>) at action.c:1548 #15 0x000000000041e0c4 in do_action (h=0x7ffd364ba5b0, a=0x7fd919e95eb8, msg=0x7fd917c5ed60 <lreq>) at action.c:1044 #16 0x000000000042a917 in run_actions (h=0x7ffd364ba5b0, a=0x7fd919e8c8e0, msg=0x7fd917c5ed60 <lreq>) at action.c:1548 #17 0x000000000042b07d in run_top_route (a=0x7fd919e8c8e0, msg=0x7fd917c5ed60 <lreq>, c=0x0) at action.c:1634 #18 0x00007fd917a0539a in t_uac_prepare (uac_r=0x7ffd364baf20, dst_req=0x7ffd364ba8f0, dst_cell=0x7ffd364ba8f8) at uac.c:391 #19 0x00007fd917a07945 in t_uac_with_ids (uac_r=0x7ffd364baf20, ret_index=0x0, ret_label=0x0) at uac.c:599 #20 0x00007fd917a07918 in t_uac (uac_r=0x7ffd364baf20) at uac.c:584 #21 0x00007fd917a09b9e in req_within (uac_r=0x7ffd364baf20) at uac.c:802 #22 0x00007fd914674052 in send_notify_request (subs=0x7ffd364bb200, watcher_subs=0x0, n_body=0x0, force_null_body=0) at notify.c:1600 #23 0x00007fd9146754b3 in notify (subs=0x7ffd364bb200, watcher_subs=0x0, n_body=0x0, force_null_body=0) at notify.c:1690 #24 0x00007fd9146cec6a in update_subscription (msg=0x7fd919efeb70, subs=0x7ffd364bb200, to_tag_gen=1, sent_reply=0x7ffd364bb16c) at subscribe.c:697 #25 0x00007fd9146d45b7 in handle_subscribe (msg=0x7fd919efeb70, watcher_user=..., watcher_domain=...) at subscribe.c:1057 #26 0x00007fd9146d08b0 in handle_subscribe0 (msg=0x7fd919efeb70) at subscribe.c:816 #27 0x000000000041e155 in do_action (h=0x7ffd364bd680, a=0x7fd919e19508, msg=0x7fd919efeb70) at action.c:1053 #28 0x000000000042a917 in run_actions (h=0x7ffd364bd680, a=0x7fd919e19508, msg=0x7fd919efeb70) at action.c:1548 #29 0x000000000041e112 in do_action (h=0x7ffd364bd680, a=0x7fd919e19650, msg=0x7fd919efeb70) at action.c:1048 #30 0x000000000042a917 in run_actions (h=0x7ffd364bd680, a=0x7fd919e02638, msg=0x7fd919efeb70) at action.c:1548 #31 0x000000000041e0c4 in do_action (h=0x7ffd364bd680, a=0x7fd919e199c0, msg=0x7fd919efeb70) at action.c:1044 #32 0x000000000042a917 in run_actions (h=0x7ffd364bd680, a=0x7fd919e199c0, msg=0x7fd919efeb70) at action.c:1548 #33 0x000000000041e112 in do_action (h=0x7ffd364bd680, a=0x7fd919e19b08, msg=0x7fd919efeb70) at action.c:1048 #34 0x000000000042a917 in run_actions (h=0x7ffd364bd680, a=0x7fd919df99d8, msg=0x7fd919efeb70) at action.c:1548 #35 0x000000000041ab46 in do_action (h=0x7ffd364bd680, a=0x7fd919dbe960, msg=0x7fd919efeb70) at action.c:677 #36 0x000000000042a917 in run_actions (h=0x7ffd364bd680, a=0x7fd919daedf8, msg=0x7fd919efeb70) at action.c:1548 #37 0x000000000042b07d in run_top_route (a=0x7fd919daedf8, msg=0x7fd919efeb70, c=0x0) at action.c:1634 #38 0x000000000050f678 in receive_msg (buf=0x7fd8f2c6ce28 "SUBSCRIBE sip:endpoint-1720@intouchstaging.com SIP/2.0\r\nRecord-Route: <sip:64.64.203.109:443;transport=tls;lr=on>\r\nAccept: application/dialog-info+xml, application/pidf+xml, application/pidf-diff+xml,"..., len=1284, rcv_info=0x7fd8f2c6cb50) at receive.c:196 #39 0x00000000005fb99e in receive_tcp_msg (tcpbuf=0x7fd8f2c6ce28 "SUBSCRIBE sip:endpoint-1720@intouchstaging.com SIP/2.0\r\nRecord-Route: <sip:64.64.203.109:443;transport=tls;lr=on>\r\nAccept: application/dialog-info+xml, application/pidf+xml, application/pidf-diff+xml,"..., len=1284, rcv_info=0x7fd8f2c6cb50, con=0x7fd8f2c6cb38) at tcp_read.c:1207 #40 0x00000000005fd48c in tcp_read_req (con=0x7fd8f2c6cb38, bytes_read=0x7ffd364bdb7c, read_flags=0x7ffd364bdb84) at tcp_read.c:1411 #41 0x0000000000600aee in handle_io (fm=0x7fd919f07d80, events=1, idx=-1) at tcp_read.c:1643 #42 0x00000000005f3ab8 in io_wait_loop_epoll (h=0xa3d3e0 <io_w>, t=2, repeat=0) at io_wait.h:1061 #43 0x0000000000601cf7 in tcp_receive_loop (unix_sock=57) at tcp_read.c:1755 #44 0x00000000005ea235 in tcp_init_children () at tcp_main.c:4788 #45 0x00000000004ab048 in main_loop () at main.c:1675 #46 0x00000000004b10c2 in main (argc=13, argv=0x7ffd364be258) at main.c:2566 --------- Thanks very much. -Cody

2 8

kamailio + dispatcher asterisks + pstn gateways
by Aidar Kamalov 08 Aug '17

08 Aug '17

Hello, Whats the best way to provide subject? Now I have central kamailio server and (asterisk "media" servers + asterisk pstn gateway in different cities). Kamailio default config with some changes in route[LOCATION] if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: xwarn("Contact $rU not registered. Forwarding call to dispatcher"); append_hf("X-External: true\r\n"); $rU = "00"+$rU; break; case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } ...... # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } if(!ds_is_from_list()) { route(DISPATCH); } route(RELAY); So all calls going to asterisk server. Not local calls with 00 prefix. And asterisk context: context kamailio { _X. => { Noop(CALL to kamailio user ${EXTEN}); Dial(SIP/${EXTEN}@${SIPDOMAIN},60,RTt); Hangup; } _00X. => { Noop(CALL to external user ${EXTEN}); Dial(SIP/PSTN-SERVER/${EXTEN:2},60,RTt); Hangup; } } Is it right way? Maybe any better approach? How to keep route logic at kamailio, but use dispatcher asterisk to proccess the calls to pstn(all endpoints at one city to pstn asterisk at this city. In others city there are other pstn gateways)? -- Aydar A. Kamalov

1 0

How to simulate a group call with kamailio ?
by leslie mekiyess 08 Aug '17

08 Aug '17

Hi, I have installed the latest kamailio on linux integrated with MYSQL server, What are the procedures ( configuration / commands etc ...) to simulate a group call ? BR, Leslie.

1 0

carrierroute and failure route
by Yu Boot 08 Aug '17

08 Aug '17

Hello. I'm trying to make carrierroute work on default kamailio.cfg first route: FXS gw ->(NAT)->Kami->FS->PSTN second route FXS gw->(NAT)-> Kami->Mediant2000->E1->PSTN First one works OK, but if second route triggers, the following disaster occurs (tshark dump) https://pastebin.com/00Ayty36 carrierroute is used as following https://pastebin.com/5J1atq00 What's wrong?

5 12

How do I add a record to a group table ?
by leslie mekiyess 08 Aug '17

08 Aug '17

Hi, I am working with the last kamilio integrated with MySQL. Is there any configuration , db cmds instructions in order to simulate a group call ? BR, Leslie.

1 0

Kamailio Server not relaying any message
by Rahman Afzal 08 Aug '17

08 Aug '17

Hi, I have an ongoing issue with kamailio...my softclient registered with kamilio originates a call which gets routed to another switch from kamilio and that switch terminates the call to B party side. Now once calls is established, B party sends BYE messages or (any re-INVITE during the dialog) which comes to the kamailio but kamailio is not relaying it to the A party...Can someone please help in this regard. Regards, Abdul Rahman

1 0

issue in Dialplan
by Logeshwaran G 08 Aug '17

08 Aug '17

Please find the below rules I am using: +----+------+----+----------+--------------+-----------+---- -----------+----------+-------+ | id | dpid | pr | match_op | match_exp | match_len | subst_exp | repl_exp | attrs | +----+------+----+----------+--------------+-----------+---- -----------+----------+-------+ | 1 | 1 | 1 | 1 | ^000[0-9]+$ | 0 | ^000([0-9]+)$ | 1 | | Correct me If I am Wrong,

1 0

Module for Call placement between the two Different Switches
by Logeshwaran G 08 Aug '17

08 Aug '17

Kindly Someone please say the name of the kamailio module which is used for Call Placement(Sip Transmission) Between the Switches. Scenario: ======== I am Having my own kamailio Pbx setup, I Have my local subscriber list database, I need to allow the calls from the outside switch(For example Nexge or Itel etc...) What module I have to refer for this??

2 2

Some Fault messages I am trying to clear up
by Kevin Greene 07 Aug '17

07 Aug '17

Hi list people: When using Siremis with Kamailio, I see this error: First Fault [2017-08-07 22:55:17 (GMT)] An exception occurred while executing this script: Error message: #256, Unable to location template file system_internal_error.tpl. Script name and line number of error: /var/www/siremis-4.3.0/openbiz/bin/Resource.php:283 function: errorHandler ( 256, "Unable to location template file system_internal_error.tpl.", "/var/www/siremis-4.3.0/openbiz/bin/Resource.php", 283, Array(11) ) @ /var/www/siremis-4.3.0/openbiz/bin/sysheader.inc 117 function: userErrorHandler ( 256, "Unable to location template file system_internal_error.tpl.", "/var/www/siremis-4.3.0/openbiz/bin/Resource.php", 283, Array(11) ) @ function: trigger_error ( "Unable to location template file system_internal_error.tpl.", 256 ) @ /var/www/siremis-4.3.0/openbiz/bin/Resource.php 283 function: getTplFileWithPath ( "system_internal_error.tpl", "email" ) @ /var/www/siremis-4.3.0/openbiz/bin/BizSystem.php 544 function: getTplFileWithPath ( "system_internal_error.tpl", "email" ) @ /var/www/siremis-4.3.0/siremis/modules/service/userEmailService.php 108 function: SystemInternalErrorEmail ( Array(2), Null ) @ /var/www/siremis-4.3.0/siremis/modules/common/form/ErrorForm.php 34 function: Report ( ) @ /var/www/siremis-4.3.0/openbiz/bin/BizController.php 310 function: invoke ( ) @ /var/www/siremis-4.3.0/openbiz/bin/BizController.php 110 function: dispatchRequest ( ) @ /var/www/siremis-4.3.0/openbiz/bin/BizController.php 32 function: include_once ( "/var/www/siremis-4.3.0/openbiz/bin/BizController.php" ) @ /var/www/siremis-4.3.0/siremis/bin/controller.php 6 It looks like I am missing a template file to send a copy of the error message to the system administrator. Also: during the installation, I, using the install script, added a number of tables to the MySQL server. However, it seems I am missing a few columns in a couple of the tables. Is there some sort of system sanity script that will check the database for all the required tables and required columns? Example: 2017-08-07 23:12:14 (GMT)] An exception occurred while executing this script: Error message: #0, Error in query:: SELECT T0.id, T0.setid, T0.url, T0.flags, T0.weight, T0.description FROM rtpproxy T0 LIMIT 10. SQLSTATE[42S02]: Base table or view not found: 1146 Table 'kamailio.rtpproxy' doesn't exist Script name and line number of error: /var/www/siremis-4.3.0/openbiz/bin/data/BizDataObj_Lite.php:447 or: [2017-08-07 23:14:58 (GMT)] An exception occurred while executing this script: Error message: #0, Error in query:: SELECT T0.id, T0.name, T0.uri, T0.mohdir, T0.mohfile, T0.debug FROM mohqueues T0 LIMIT 10. SQLSTATE[42S02]: Base table or view not found: 1146 Table 'kamailio.mohqueues' doesn't exist or: [2017-08-07 23:16:57 (GMT)] An exception occurred while executing this script: Error message: #42S22, SQLSTATE[42S22]: Column not found: 1054 Unknown column 'src_user' in 'field list' Script name and line number of error: /var/www/siremis-4.3.0/openbiz/others/Zend/Db/Statement/Pdo.php:234 or: 2017-08-07 23:18:33 (GMT)] An exception occurred while executing this script: Error message: #0, Error in query:: SELECT T0.id, T0.method, T0.from_tag, T0.to_tag, T0.callid, T0.sip_code, T0.sip_reason, T0.time, T0.src_ip, T0.dst_user, T0.dst_domain, T0.dst_ouser, T0.src_user, T0.src_domain, T0.cdr_id FROM acc T0 WHERE method='INVITE' AND T0.src_user='admin' LIMIT 10. SQLSTATE[42S22]: Column not found: 1054 Unknown column 'T0.src_ip' in 'field list' In the case of the above table, it does not say which table. Anyway -- any help would be great. Thanks. Kevin

1 0

load credentials in cnxcc from dburl
by Logeshwaran G 07 Aug '17

07 Aug '17

Is there any possibility of loading credential from the db in the cnxcc module. For Example: modparam("auth_db", "load_credentials", "$avp(status)=status") Like the above one , I need it for CNXCC

3 7

Question regarding t_check_trans()
by Joe Baran 07 Aug '17

07 Aug '17

Probably an elementary question, but we're relatively new at digging into the details. We are trying to understand the following block of config code and have a question or two. Our assumption is the t_precheck_trans() function tells us whether another packet has arrived already as a retransmission. However, we're not clear on what the t_check_trans() is doing assuming the condition is true. Any direction on this? Thanks. route[CHECK_RETRANS] { # handle retransmissions if(t_precheck_trans()) { t_check_trans(); exit; } } Joe Baran

3 2

Endpoints distribution to dispatcher
by Aidar Kamalov 05 Aug '17

05 Aug '17

Hello, I want to distribute endpoints calls in different cities to asterisks in the same cities where the endpoint. Now I use dispatcher for distibute calls beetwen asterisks, but without reference to the city. Can you please suggest better module and way for this? Is it better to distribute based on ip? Or may be based on groups? -- Aydar A. Kamalov

2 2

Fwd: Re: Fwd: kemi lua routing block as route in standart cfg
by Yuriy Gorlichenko 04 Aug '17

04 Aug '17

Cool!!! Thx for this reply and help. Yep i will need. I using WS. On Aug 4, 2017 15:33, "Daniel-Constantin Mierla" <miconda(a)gmail.com> wrote: v5.0 doesn't have many of the event route available for kemi. If you use master branch (upcoming 5.1), modules having event routes export a parameter 'event_callback' which needs to be set to a name of a function inside the embedded language script, see the one for tm at: - https://kamailio.org/docs/modules/devel/modules/tm.html#tm.p .event_callback The dialog module should have same mod param. For websocket, do you need the one for xhttp module? If yes, it's available. The event_route[websocket:closed] is not available, as I can see in the code, but I will add it. Cheers, Daniel On 03.08.17 08:38, Yuriy Gorlichenko wrote: Thank you for your response! Probably it was at 5.0.x kemi guide. But anyway Domt uderstand how to call it. I need tm:local reqest, Websockets event_route and dialog. Thats it) On Aug 3, 2017 08:15, "Daniel-Constantin Mierla" <miconda(a)gmail.com> wrote: > Hello, > > where is that TBD note? It needs to be adjusted, probably it is a leftover > from first days of this feature. > > The failure_route, branch_route and onreply_route can be now natively in > the embedded language, see the Lua script example I provide with the link > in my previous email. > > If you are using event_route blocks, let me know which and I can say if > they are available -- many of them are, but some are missing, as I am not > using all the modules available and didn't have the time to dig in all of > them. > > Cheers, > Daniel > > On 02.08.17 13:42, Yuriy Gorlichenko wrote: > > > thx. I just thought that it's possibe to concatenate both engines together. > > So this is my last question regardng kemi: > > Main goal why i still not use kemi at the my projects is a: > > TBD: the options for specific event_route blocks. Meanwhile, should work > using hybrid configuration with request_route/reply_route/… in embedded > interpreter and the other routing blocks in native kamailio.cfg. > > As i uderstand it is not possible to use event_route in a kemi. only > native. If yes - do you expect t ointegrate this part of routes at the kemi > interface in future? > > > > 2017-08-02 14:36 GMT+03:00 Daniel-Constantin Mierla <miconda(a)gmail.com>: > >> Hello, >> >> if you want to do routing in lua via kemi, the kamailio.cfg has only the >> passive part (global parameters, loading modules and modparams) and the >> active part (routing blocks) is a separate file written in the desired >> scripting language (e.g., lua). See examples: >> >> - https://github.com/kamailio/kamailio/blob/master/misc/exampl >> es/kemi/kamailio-basic-kemi.cfg >> >> - https://github.com/kamailio/kamailio/blob/master/misc/exampl >> es/kemi/kamailio-basic-kemi-lua.lua >> >> If you still want to have the native kamailio.cfg with its routing >> blocks, but executed some Lua scripts inline, see app_lua exported >> functions: >> >> - http://www.kamailio.org/docs/modules/5.1.x/modules/app_lua.h >> tml#app_lua.f.lua_run >> >> Cheers, >> Daniel >> >> On 02.08.17 13:28, Yuriy Gorlichenko wrote: >> >> Hi, no >> I wanna include lua routes part at existing kamailio.cfg file >> >> Just wanna rewrite some route with lua code using kemi because lua more >> flexible for my needs, but want to call this route at the cfg file. >> >> like >> kamailio.cfg >> ... >> route { >> [MY_SUPER_ROUTE_IN_LUA] >> } >> ... >> >> Where MY_SUPER_ROUTE_IN_LUA is a route written with kemi on lua language. >> >> >> 2017-08-02 14:10 GMT+03:00 Daniel-Constantin Mierla <miconda(a)gmail.com>: >> >>> Hello, >>> >>> >>> On 02.08.17 12:02, Yuriy Gorlichenko wrote: >>> > Hello. I found how to implement lua routimg logic with >>> > cfgengine but didn't found how to include cimple route in lua at the >>> > cfg file >>> > >>> > Presume i have routing logic and want to include >>> > my_super_custom_route written with Kemo on lua. How i need to set up >>> > my config for it work with both engines? >>> > >>> I am not sure I understand what you try to do. The path to yiour Lua >>> script with routing logic has to be set via modparam of app_lua module. >>> >>> Are you trying to include a lua file in another lua file? >>> >>> Cheers, >>> Daniel >>> >>> -- >>> Daniel-Constantin Mierla >>> www.twitter.com/miconda -- www.linkedin.com/in/miconda >>> Kamailio Advanced Training - www.asipto.com >>> Kamailio World Conference - www.kamailioworld.com >>> >>> >> >> >> _______________________________________________ >> Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> >> >> -- >> Daniel-Constantin Mierlawww.twitter.com/miconda -- www.linkedin.com/in/miconda >> Kamailio Advanced Training - www.asipto.com >> Kamailio World Conference - www.kamailioworld.com >> >> > > > > _______________________________________________ > Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > > -- > Daniel-Constantin Mierlawww.twitter.com/miconda -- www.linkedin.com/in/miconda > Kamailio Advanced Training - www.asipto.com > Kamailio World Conference - www.kamailioworld.com > > -- Daniel-Constantin Mierlawww.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - www.kamailioworld.com

1 0

Pull Request or Issue? - coredump in case no contentLengthHdr present in REGISTER
by Valentin Christoph 04 Aug '17

04 Aug '17

Hi all, We have identified an IMS related issue, when the "Content-Length" header is missing in REGISTER request. It's a missing null pointer check in file modules/ims_auth/utils.c We are using Kamailio 4.4.5 for this purpose and we would need a fix in that release. I'm not sure, how to proceed now? Should I just create an "Issue" at GitHub, or is it possible for me to create a "Pull Request" at GitHub (would be less effort for you, I guess). Anyway, the change we did in our local GIT repository, is shown in following picture: [cid:image001.jpg@01D30C76.DE88AE10] Thanks, Christoph The information contained in this e-mail message is privileged and confidential and is for the exclusive use of the addressee. The person who receives this message and who is not the addressee, one of his employees or an agent entitled to hand it over to the addressee, is informed that he may not use, disclose or reproduce the contents thereof, and is kindly asked to notify the sender and delete the e-mail immediately.

2 1

Fwd: kemi lua routing block as route in standart cfg
by Yuriy Gorlichenko 04 Aug '17

04 Aug '17

thx. I just thought that it's possibe to concatenate both engines together. So this is my last question regardng kemi: Main goal why i still not use kemi at the my projects is a: TBD: the options for specific event_route blocks. Meanwhile, should work using hybrid configuration with request_route/reply_route/… in embedded interpreter and the other routing blocks in native kamailio.cfg. As i uderstand it is not possible to use event_route in a kemi. only native. If yes - do you expect t ointegrate this part of routes at the kemi interface in future? 2017-08-02 14:36 GMT+03:00 Daniel-Constantin Mierla <miconda(a)gmail.com>: > Hello, > > if you want to do routing in lua via kemi, the kamailio.cfg has only the > passive part (global parameters, loading modules and modparams) and the > active part (routing blocks) is a separate file written in the desired > scripting language (e.g., lua). See examples: > > - https://github.com/kamailio/kamailio/blob/master/misc/exampl > es/kemi/kamailio-basic-kemi.cfg > > - https://github.com/kamailio/kamailio/blob/master/misc/exampl > es/kemi/kamailio-basic-kemi-lua.lua > > If you still want to have the native kamailio.cfg with its routing blocks, > but executed some Lua scripts inline, see app_lua exported functions: > > - http://www.kamailio.org/docs/modules/5.1.x/modules/app_lua.h > tml#app_lua.f.lua_run > > Cheers, > Daniel > > On 02.08.17 13:28, Yuriy Gorlichenko wrote: > > Hi, no > I wanna include lua routes part at existing kamailio.cfg file > > Just wanna rewrite some route with lua code using kemi because lua more > flexible for my needs, but want to call this route at the cfg file. > > like > kamailio.cfg > ... > route { > [MY_SUPER_ROUTE_IN_LUA] > } > ... > > Where MY_SUPER_ROUTE_IN_LUA is a route written with kemi on lua language. > > > 2017-08-02 14:10 GMT+03:00 Daniel-Constantin Mierla <miconda(a)gmail.com>: > >> Hello, >> >> >> On 02.08.17 12:02, Yuriy Gorlichenko wrote: >> > Hello. I found how to implement lua routimg logic with >> > cfgengine but didn't found how to include cimple route in lua at the >> > cfg file >> > >> > Presume i have routing logic and want to include >> > my_super_custom_route written with Kemo on lua. How i need to set up >> > my config for it work with both engines? >> > >> I am not sure I understand what you try to do. The path to yiour Lua >> script with routing logic has to be set via modparam of app_lua module. >> >> Are you trying to include a lua file in another lua file? >> >> Cheers, >> Daniel >> >> -- >> Daniel-Constantin Mierla >> www.twitter.com/miconda -- www.linkedin.com/in/miconda >> Kamailio Advanced Training - www.asipto.com >> Kamailio World Conference - www.kamailioworld.com >> >> > > > _______________________________________________ > Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > > -- > Daniel-Constantin Mierlawww.twitter.com/miconda -- www.linkedin.com/in/miconda > Kamailio Advanced Training - www.asipto.com > Kamailio World Conference - www.kamailioworld.com > >

2 3

Sip-clients not connecting to kamailio, What am I missing?
by Viktor Silfver 04 Aug '17

04 Aug '17

Hi, I’m very new to Kamailio, in fact I have never got Kamailio to work properly. I’m following this tutorial http://kamailio.org/docs/tutorials/5.1.x/kamailio-install-guide-deb/ <http://kamailio.org/docs/tutorials/5.1.x/kamailio-install-guide-deb/> step by step, but my subscribers are not able to connect to Kamailio. (X-lite and Zoiper) I can install Kamialio, create the database, start Kamailio, add subscribers, without any failures or error messages. I’m running debian jessie. When i’m running wireshark I get the message Port unreachable. What am I missing? Can someone give me a hint on what I should look for?

2 1

route[LOCATION] is it bug?
by Aidar Kamalov 04 Aug '17

04 Aug '17

Hello, I 'm new in kamailio. I have kamailio with dispatcher and two asterisks. Endpoints register to kamailio. Calls beetwen endpoints going through dispatcher to asterisks. Calls to other destination rejected by kamailio with 404. But If i put xlog("AAAA") or xwarn in route[LOCATION] in if (!lookup("location")) block (default kamailio 5.0.2 config) calls to other destination goes through dispatcher to asterisks. I cant't understand WHY? :( (sorry for my bad English) route[LOCATION] { #!ifdef WITH_SPEEDDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") { if(sd_lookup("speed_dial")) { route(SIPOUT); } } #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) { route(SIPOUT); } #!endif $avp(oexten) = $rU; if (!lookup("location")) { xwarn("method ($rm) r-uri ($ru)\n"); ################## HERE ########################## $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } if(!ds_is_from_list()) { route(DISPATCH); } route(RELAY); exit; } -- Aydar A. Kamalov

2 2

Kamailio inserting Quote " mark in SIP header
by Arslan Saeed 03 Aug '17

03 Aug '17

Hi All, We are seeing that Kamailio's nat_helper_module function of add_rcv_param(), inserts " (quote) mark to some of the SIP headers (as below in the contact header in sip 200 OK response) Contact: <sip:172.70.1.2:5060;bnc;received="sip:118.107.184.242:25090 ">;expires=172 We got few SIP UAs that are complaining about the quote mark inclusion in contact header and responding back with SIP 400 bad syntax status code. Requesting views / suggestions from community about this behavior of kamailio and any workarounds available. Cheers Arslan

2 1

Using a number as htable key name
by Jan Hazenberg 03 Aug '17

03 Aug '17

Hello, I'm trying to use a htable to reroute some numbers but when i'm trying to use kamcmd to add a entry where the element name is a number instead of a string i get a 500 error kamcmd htable.seti force_number 1234 1 error: 500 - Not enough parameters (htable name, key name and value) Also with quotes i get the 500 error: kamcmd htable.seti force_number "1234" 1 error: 500 - Not enough parameters (htable name, key name and value) But this works: kamcmd htable.seti force_number a1234 1 Is there a way to escape this so i can use a number as element name? Regards, Jan Hazenberg

1 0

Kamailio example configuration: auth section incomplete?
by George Diamantopoulos 02 Aug '17

02 Aug '17

Hello again, Still getting familiar with kamailio, and I'm wondering about the AUTH route in the example configuration file. Here's a reducted-simplified version of it for reference (from git master, without IP AUTH and comments): ______________________________________________ route[AUTH] { if (is_method("REGISTER") || from_uri==myself) { if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } return; } ______________________________________________ So the way I see it, what happens is the following: * All REGISTERs will be challenged * All SIP messages with kamailio's aliases in the "From" header URI will be challenged * All SIP messages with no reference to kamailio's aliases in both R-URI and "From" header URI will be dropped The question is, what about messages that do not enter either of the two conditionals? For example, I expect the following to be very common: * Method: INVITE * R-URI: myself * From: username@"UAC's local IP address" (not myself) * To: myself So in the example above, the auth route will return without either having challenged or dropped the request, am I correct? This is because: * For the challenge: Method is not REGISTER and "From URI" is not one of kamailio host's aliases (cumulatively) * For dropping after sending 403: "From URI" is not one of kamailio's host's aliases (which calculates to true) but R-URI is "myself" So I'm guessing we're expecting the challenge to come from elsewhere in cases like the example above? Or is there something else I'm missing here? Thanks! BR, George

2 5

Issue with ims_isc module? (distinct PSI)
by Valentin Christoph 02 Aug '17

02 Aug '17

Hi, Can anyone comment on issue #1168? https://github.com/kamailio/kamailio/issues/1168 Thanks, Christoph The information contained in this e-mail message is privileged and confidential and is for the exclusive use of the addressee. The person who receives this message and who is not the addressee, one of his employees or an agent entitled to hand it over to the addressee, is informed that he may not use, disclose or reproduce the contents thereof, and is kindly asked to notify the sender and delete the e-mail immediately.

3 3

is_first_hop
by Nicolas Breuer 02 Aug '17

02 Aug '17

Hello, I tried to understand the concept of first hop in "replies" In the documentation ; For incoming SIP replies, it means that top Record-Route URI is 'myself' and source address is not matching it (to avoid detecting in case of local loops). Note that it does not detect spirals, which can have the condition for replies true also in the case of additional SIP reply receival. Here's an incoming reply received on 217.112.180.10. Via: SIP/2.0/UDP 217.112.180.10;branch=z9hG4bK723f.db3510ce15ac9bb6467e115ecc2fe2a7.0. Via: SIP/2.0/UDP 217.112.180.9;rport=5060;branch=z9hG4bK723f.741041093fec570744afcf44f2194581.0. Via: SIP/2.0/UDP 217.110.38.229:5060;rport=5060;branch=z9hG4bK0eB7070489d338025ed. Record-Route: <sip:217.112.180.9:5060;lr=on;ftag=7B84E4CBh3;did=4cf.c94;nat=yes>. Record-Route: <sip:217.112.180.10:5060;lr=on;ftag=7B84E4CBh3;did=4cf.448>. The Record-Route is 217.112.180.9 but the first proxy is 217.112.180.10.

1 0

SIP Proxy/Redirect
by Aaron Tecey 02 Aug '17

02 Aug '17

Hi All, Hoping for some help as a beginner. I need to configure a sip proxy server so that it redirects all processes to a freeswitch server on another network. SIP UACs are on an isolated network and have only a connection to the sip proxy. All I need the proxy to do is pass on all requests to the the freeswitch UAS on a different network, as it will handle registering etc. Is there a basic setup for this type of behaviour? Many thanks, Aaron

2 1

error 401 and 500 (7/SL)
by jurek 02 Aug '17

02 Aug '17

Hello i have problem with my kamailio sip server on rpi 3b i configured tls with lets encrypt when i start kamailio with #!define WITH_AUTH its always return 401 error but without tls working . if i start kamailio without #!define WITH_AUTH its return [500] reason [I'm terribly sorry, server error occurred (7/SL)] for proxy [<sip:jurek.tk:5061;transport=tls>] How to repair it ?

1 0

kemi lua routing block as route in standart cfg
by Yuriy Gorlichenko 02 Aug '17

02 Aug '17

Hello. I found how to implement lua routimg logic with cfgengine but didn't found how to include cimple route in lua at the cfg file Presume i have routing logic and want to include my_super_custom_route written with Kemo on lua. How i need to set up my config for it work with both engines?

2 3

Reverse DNS lookups inside Kamailio
by Sebastian Damm 02 Aug '17

02 Aug '17

Hi, for a project I need the option to do a reverse lookup of an IP address and use the value from it in a variable. As far as I could see the ipops module only allows SRV and A record lookups. Is there anything I missed? And if there is nothing inside Kamailio, how could a good approach look like? Thanks in advance, Sebastian

2 1

T.38 fax, rtpproxy, re-invite
by Yu Boot 02 Aug '17

02 Aug '17

Hello. I'm using almost default kamailio.cfg with nathelper, rtpproxy, PSTN dialing and other default features. I have following scheme: FXS gateway -> (NAT) -> Kamailio -> FreeSwitch Voice calls work perfectly, no one-direction sound and so on. But with fax machine there was a problem. In wireshark I saw UDPTL from FS to Kamailio IP, and no UDPTL at all to FXS gw. I've guessed that something's wrong with a re-INVITE request processing in Kamailio. I've added this to WITHINDLG route: else if ( is_method("INVITE") ) { record_route(); } And problem was magically solved, T.38 now works OK. Is this OK solution or this may break something elsewhere? Thanks.

2 1

Kamailio CDR "best practices"
by Yu Boot 02 Aug '17

02 Aug '17

How to make Kamailio write "normal" CDRs? acc cdr module writes start time and end time in unixtime format, that is very uncomfortable. Fnd stcond question - how to make Kamailio to write CDR in ordinary text file?

2 3

Usleep on Reply
by Nicolas Breuer 01 Aug '17

01 Aug '17

Hello All, I'm trying to code something on the on_reply part to do something if I receive a 183 ( Session in progress ) , followed immediately by a "180" I tried to sleep 1 second the 183 and recheck if I received before a 180 but seems the "sleep" function sleeps everything ( also the received 180 ) Is a developper can confirm that ? The goal is to detect if a 180 follow the 183.

3 4

Getting Error - error reading: Connection reset by peer (104)
by Prem Kumar N 01 Aug '17

01 Aug '17

Hi All, I have a setup with two kamailio v5.0 and a load balancer(keepalived) It seems to check the heath of kamailio server load balancer establishing the tcp conneciton and resetting the connection once its got established. and due to resetting the connection I keep on getting following log message. Jul 31 18:14:18 localhost /usr/local/kamailio/sbin/kamailio[9388]: ERROR: <core> [core/tcp_read.c:291]: tcp_read_data(): error reading: Connection reset by peer (104) ([192.168.xxx.xxx]:46574 -> Jul 31 18:14:18 localhost /usr/local/kamailio/sbin/kamailio[9388]: ERROR: <core> [core/tcp_read.c:292]: tcp_read_data(): -> [192.168.yyy.yyy]:443) Jul 31 18:14:18 localhost /usr/local/kamailio/sbin/kamailio[9388]: ERROR: <core> [core/tcp_read.c:1319]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f6547989e78 r: 0x7f6547989ef8 Jul 31 18:14:18 localhost /usr/local/kamailio/sbin/kamailio[9389]: ERROR: <core> [core/tcp_read.c:291]: tcp_read_data(): error reading: Connection reset by peer (104) ([192.168.xxx.xxx]:44279 -> Jul 31 18:14:18 localhost /usr/local/kamailio/sbin/kamailio[9389]: ERROR: <core> [core/tcp_read.c:292]: tcp_read_data(): -> [192.168.yyy.yyy]:5060) Jul 31 18:14:18 localhost /usr/local/kamailio/sbin/kamailio[9389]: ERROR: <core> [core/tcp_read.c:1319]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f6547989e78 r: 0x7f6547989ef If any one can suggest, how can i get rid of these logs. Thanks Prem

2 2

uac_auth credentials
by Volkan Oransoy 01 Aug '17

01 Aug '17

Hi all, I use UAC module for remote registrations on my proxy box. There is no problem with incoming calls and I successfully register and receive calls. I have a Freeswitch box behind this proxy box and I want my proxy to add authorisation headers for outgoing calls transparently. I already have credentials at uacreg table for remote registrations. Is it possible use same credentials at uac_auth() function? Or should I run SQL to retrieve necessary fields to feed uac_auth? Thanks! /Volkan

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