sr-users April 2020

sr-users@lists.kamailio.org

70 participants
96 discussions

How to install the PHONENUM module?
by Edward Romanenco 20 Apr '20

20 Apr '20

Hi, I'm trying to add the PHONENUM module into my Kamailio installation. For this, I've cloned the main branch of Libphonenumber<https://github.com/google/libphonenumber/tree/master/cpp> and followed the installation rules as they appear in the relevant README file: Building and testing the library -------------------------------- $ cd libphonenumber/cpp $ mkdir build $ cd build $ cmake .. $ make $ ./libphonenumber_test It all went through and the library was created, but when I try to make and install the module itself, I'm getting the following error. Looks like the library wasn't included in the building context, can you please lend me a hand and tell me how do I include it? root@kamaillioegress:/usr/local/src/kamailio-5.1/kamailio/src/modules/phonenum# make Compiling cphonenumber.cpp g++ -fPIC -DPIC -g -funroll-loops -Wcast-align -m64 -minline-all-stringops -falign-loops -ftree-vectorize -fno-strict-overflow -Wall -Wno-write-strings -Wno-deprecated -Wno-unused-function -Wno-sign-compare -Wno-strict-aliasing -fPIC -DPIC -g -funroll-loops -Wcast-align -m64 -minline-all-stringops -falign-loops -ftree-vectorize -fno-strict-overflow -Wall -DNAME='"kamailio"' -DVERSION='"5.2.0-dev6"' -DARCH='"x86_64"' -DOS='linux_' -DOS_QUOTED='"linux"' -DCOMPILER='"gcc 5.4.0"' -D__CPU_x86_64 -D__OS_linux -DVERSIONVAL=5002000 -DCFG_DIR='"/usr/local/kamailio-5.1/etc/kamailio/"' -DRUN_DIR='"/var/run/kamailio/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST -DUSE_NAPTR -DWITH_XAVP -DMEM_JOIN_FREE -DF_MALLOC -DQ_MALLOC -DTLSF_MALLOC -DDBG_SR_MEMORY -DUSE_TLS -DTLS_HOOKS -DUSE_CORE_STATS -DSTATISTICS -DMALLOC_STATS -DWITH_AS_SUPPORT -DUSE_SCTP -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_SCHED_SETSCHEDULER -DHAVE_IP_MREQN -DUSE_RAW_SOCKS -DHAVE_EPOLL -DHAVE_SIGIO_RT -DSIGINFO64_WORKARROUND -DUSE_FUTEX -DHAVE_SELECT -I/opt/local/include -DKAMAILIO_MOD_INTERFACE -DMOD_NAME='"phonenum"' -c cphonenumber.cpp -o cphonenumber.o cphonenumber.cpp:24:65: fatal error: phonenumbers/geocoding/phonenumber_offline_geocoder.h: No such file or directory compilation terminated. Makefile:22: recipe for target 'cphonenumber.o' failed make: *** [cphonenumber.o] Error 1 Edward

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Free Persian Asterisk Book and COVID-19 Pandemic
by Mojtaba 19 Apr '20

19 Apr '20

In the context of the COVID-19 pandemic, Nasim Telecom decided to publish the first six-captures of Persian Asterisk book that was written by me and Mr Najafi, published in 2017. Also it was introduced by Mr. David Duffett in AstriCon 2017 on Florida. It would be useful for all people which understand Persian language, Also all other people could use it by going code commands on it and examples. For downloading it go the following link: https://www.nasimtelecom.com/en/persian-asterisk-book/ So, be careful and safety at first, Stay at home and read your favourite books. With Best Regards. --Mojtaba Esfandiari.S

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rtpenigne & STUN
by Ahed Saed 17 Apr '20

17 Apr '20

Hi, I'm using kamailio 5.3 & rtpengine ( both kamailio and rtpengine resides on public IP on same machine) am using below logic to handle clients behind NAT if(nat_uac_test("8")){ rtpengine_manage("SIP-source-address replace-origin replace-session-connection "); }else{ rtpengine_manage("trust-address replace-origin replace-session-connection "); } I don't want all the rtp packet to go over the server, i just want to make it use the rtpengine just in case it is needed. but am not sure how this can be done using kamailio, from the client side i'll try to use STUN, but i understand STUN doesn't work all the time ( ~90% of the cases it will work) can anyone help me with this ? another question - if one client works with STUN and the other is behind NAT how this should be handled? Cheers

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TCP/TLS connection (id: 0) for WebSocket could not be found
by Mack Hendricks 17 Apr '20

17 Apr '20

Hey All, I get this message when trying to route request to a WebSocket client: Apr 15 04:59:37 dSIP060entNightly-0 /usr/sbin/kamailio[22071]: DEBUG: <core> [core/msg_translator.c:1762]: check_boundaries(): no multi-part body Apr 15 04:59:37 dSIP060entNightly-0 /usr/sbin/kamailio[22071]: DEBUG: <core> [core/msg_translator.c:429]: clen_builder(): content-length: 651 (651) Apr 15 04:59:37 dSIP060entNightly-0 /usr/sbin/kamailio[22071]: WARNING: <core> [core/msg_translator.c:2786]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found Apr 15 04:59:37 dSIP060entNightly-0 /usr/sbin/kamailio[22071]: ERROR: <core> [core/msg_translator.c:2002]: build_req_buf_from_sip_req(): could not create Via header Apr 15 04:59:37 dSIP060entNightly-0 /usr/sbin/kamailio[22071]: ERROR: tm [t_fwd.c:476]: prepare_new_uac(): could not build request Apr 15 04:59:37 dSIP060entNightly-0 /usr/sbin/kamailio[22071]: ERROR: tm [t_fwd.c:1738]: t_forward_nonack(): failure to add branches Apr 15 04:59:37 dSIP060entNightly-0 /usr/sbin/kamailio[22071]: DEBUG: tm [t_funcs.c:337]: t_relay_to(): t_forward_nonack returned error -2 (-2) Apr 15 04:59:37 dSIP060entNightly-0 /usr/sbin/kamailio[22071]: DEBUG: tm [t_funcs.c:355]: t_relay_to(): -2 error reply generation delayed Apr 15 04:59:37 dSIP060entNightly-0 /usr/sbin/kamailio[22071]: exec: *** cfgtrace:request_route=[RELAY] c=[/etc/kamailio/kamailio.cfg] l=1176 a=24 n=sl_reply_error Apr 15 04:59:37 dSIP060entNightly-0 /usr/sbin/kamailio[22071]: ERROR: sl [sl_funcs.c:362]: sl_reply_error(): stateless error reply used: No error (2/SL) Apr 15 04:59:37 dSIP060entNightly-0 /usr/sbin/kamailio[22071]: exec: *** cfgtrace:request_route=[RELAY] c=[/etc/kamailio/kamailio.cfg] l=1178 a=2 n=exit Here is what my location table looks like. It should try to send the call to sip:98.209.240.245:56291;transport=ws *************************** 2. row *************************** id: 124 ruid: uloc-5e9685e8-5650-92 username: 2000 domain: ws-test.com contact: sips:2000@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=wss received: sip:98.209.240.245:56291;transport=ws path: NULL expires: 2020-04-15 05:03:17 q: -1.00 callid: 7ef0a550-b2b9-24c6-5c3b-8e0c667f5533 cseq: 50096 last_modified: 2020-04-15 04:59:57 flags: 0 cflags: 524352 user_agent: IM-client/OMA1.0 sipML5-v1.2016.03.04 socket: NULL methods: NULL instance: NULL reg_id: 0 server_id: 0 connection_id: -1 keepalive: 1 partition: 18 The Websocket list looks like this: Server01:~# kamcmd ws.dump { connections: { 8: wss:98.209.240.245:56291 -> wss:134.122.27.49:4443 (state: OPEN, last used 47s ago, sub-protocol: sip) } info: { wscounter: 1 truncated: no } } I’m not sure what I’m doing wrong. Any ideas? Thanks in advance -Mack

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Kamailio in osm
by Pavithra M 17 Apr '20

17 Apr '20

Hi, Do anybody have worked orchestrating kamailio as an IMS server through osm by launching vnfs in openstack . If so kindly reply me as I need some inputs on that.

1 0

Dealing with mobile client state while using Push Notification and TSILO
by Hubert Odziemczyk 17 Apr '20

17 Apr '20

Hello! We are trying to setup a call-flow with TSILO and Push Notifications for solution with a mobile SIP client (Linphone), based on presentation by Federico Cabiddu (http://www.kamailio.org/events/2015-KamailioWorld/Day2/20-Federico.Cabiddu-…). We have an issue with "managing" state of the client (active or not) when there is a call coming. To be more specific, lookup in location table doesn't always give a proper answer, as contact expiration is different from actual lifetime of the app (which also differs between iOS and Android). One of proposed solutions is setting Expires=1 for REGISTER and then always rely on Push Notifications, but this scenario seems to be unreliable with some clients that are in foreground, as Push Notification doesn't trigger REGISTER again. Question is: did you perceive similar problems? What are your solutions to deal with them? Thank you in advance! Best regards, Hubert

2 1

Announcing SaraPhone, SIP WebRTC Open Source business phone
by Giovanni Maruzzelli 16 Apr '20

16 Apr '20

My fellow VoIPers, I am pleased to announce the early availability of: SaraPhone ------------------ SaraPhone is a bare bone SIP WebRTC voice phone, complete with most features real companies want to use in real world: HotDesking, Redial, BLFs, MWI, DND, PhoneBook, Hold, Transfer, Mute, Attended Transfer, Notifications, running on all Browsers both on Desktop and SmartPhone. SaraPhone is fully integrated with FusionPBX, the full-featured domain based multi-tenant PBX and voice switch for FreeSwitch. Based on SIP.js, SaraPhone works with all WebRTC compliant SIP proxies, gateways, and servers (Asterisk, Kamailip, OpenSIPS, etc). Initial author is Giovanni Maruzzelli, and SaraPhone gets its name from Giovanni's wife, Sara Hosseini. In addition to providing all of the usual DeskPhone functionality, SaraPhone got: - Desktop Notification for Incoming Calls - Live MWI update - Real Time BLFs status update - BLF click to call - Caller Name and Number Display - Call Error Cause Display - AutoAnswer - Network Disconnect Reload - Show and Set Caller-ID (incoming-outbound) You an find it in GitHub ( https://github.com/gmaruzz/saraphone ). Anyone interested can play with it :). Have fun, giovanni -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18

2 2

kamailio integration with MS teams.
by Nasida Yuriy 16 Apr '20

16 Apr '20

Hi guys, Do you have some expirience with kamailio integration with MS teams ? I follow the instructon from here https://skalatan.de/en/blog/kamailio-sbc-teams TLS part is configured correctly. I also got OPTIONS pings working between MS teams and kamailio by following of this instruction. That is very good. But... There is issues with outbound calls from MS teams to kamailo. MS side send INVITE, kamailio responds with 180 and 200 OK, but looks like MS ignores us. I know that I probably should use record_route_preset function here to modify Record-Route headers to satisfy MS wishes. But nothing helps. I also tried to modify contact header in 200 OK like it was done for OPTIONS - no luck. Anyone has a working example of kamailio files for MS team ? Or at least sucessful pcap with them. I can be wrong but looks like MS doesn't respect RFC at all. Thanks. Regards, Team lead<https://www.odesk.com/companies/Nasida_~0168bc9fd46660bcca> [https://odesk-prod-portraits.s3.amazonaws.com/Companies:1712760:CompanyLogo…]

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Weird dialplan result
by David Villasmil 15 Apr '20

15 Apr '20

Hello guys, So i have the following dialplan (almost literally, 2 records) +----+------+-----+----------+-----------+-----------+-----------+----------+---------------------------------------+ | id | dpid | pr | match_op | match_exp | match_len | subst_exp | repl_exp | attrs | +----+------+-----+----------+-----------+-----------+-----------+----------+---------------------------------------+ | 10 | 1 | 100 | 0 | 910000000 | 0 | | | 1.2.3.4:5060 | | 11 | 1 | 100 | 0 | 910000001 | 0 | | | sip.somedomain.net:5060 | And with kamcmd: kamcmd> dialplan.translate 1 s:910000000 { Output: Attributes: 1.2.3.4:5060 } and kamcmd> dialplan.translate 1 s:910600000 error: 500 - No translation So that works... now, when i get an INVITE I do: dp_match("1", "$rU"); xlog("[DIALPLAN] '$rU' was translated to '$var(destination)'\n"); if ( $var(destination)!=0 ) { $ru = "sip:" + $fU + "@" + $var(destination); route(RELAY); } And this prints: <script>: [DIALPLAN] '910600000' was translated to '195.114.216.188:5060' What am I doing wrong? Kind of confused here... Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 1

Kamailio propagates 180 and 200 OK OUT OF ORDER
by Luis Rojas G. 15 Apr '20

15 Apr '20

Good day, I am testing the dispatcher module, using Kamailio as stateless proxy. I have a pool of UAC (scripts in SIPP) and a pool of UAS (also scripts in SIPP) for the destinations. Kamailio version is kamailio-5.3.3-4.1.x86_64. Problem I have is, if UAS responds 180 and 200 OK to Invite immediately, sometimes they are propagated out of order. 200 OK before 180, like this : UAS is 172.30.4.195:5061. UAC is 172.30.4.195:5080. Kamailio is 192.168.253.4:5070 Difference between 180 and 200 is just about 50 microseconds. My guess is that both messages are received by different instances of Kamailio, and then because of context switches, even though the 180 is received before, that process ends after the processing of 200. However, I had the idea that in order to avoid these problems the kamailio processes synchronized with each other using a shared memory. I tried using stateful proxy and I obtained the same result. By the way, anyone has any idea about how Kamailio's share memory is implemented? It clearly does not use the typical system calls shmget(), shmat(), because they are not shown by ipcs command. Before posting here I googled, but I couldn't find anything related to this. I can't believe I am the only one who ever had this problem, so I guess I am doing something wrong... Please, any help. I'm really stuck on this. Thanks. -- Luis Rojas Software Architect Sixbell Los Leones 1200 Providencia Santiago, Chile Phone: (+56-2) 22001288 mailto:luis.rojas@sixbell.com http://www.sixbell.com -- Luis Rojas Software Architect Sixbell Los Leones 1200 Providencia Santiago, Chile Phone: (+56-2) 22001288 mailto:luis.rojas@sixbell.com http://www.sixbell.com

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sr-users April 2020