- sr-users - kamailio.org

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

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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*

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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

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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.

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