- sr-users - kamailio.org

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

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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

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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

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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!

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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

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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) *

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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

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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

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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

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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*

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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

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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*

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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

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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 }

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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

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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

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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

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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

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