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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 >>>> >>>> >>> >

2 2

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

1 0

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?

1 0

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

3 6

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/

2 1

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/

2 2

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

4 21

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

3 2

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

3 2

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

1 0

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