- sr-users - kamailio.org

How does dlg_bridge work?
by Pranathi Venkatayogi 04 May '17

04 May '17

I have a TEXT call from cust1@somedomain to Kamailio. I have an agent who is logged in, but is not in any call yet. I want now refer cust1@somedomain to talk to agent directly. How do I do that? I tried - dlg_bridge("$dlg(from_contact)", "$var(agentSipAddress)",""); I get an "audio call" from "Kamailio" to customer, but nothing happens on the agent side. I tried "dlg_refer" as well but nothing happens at all. Should I handcraft "Refer" message using uac module and send to "customer" instead? Please suggest. Thanks Pranathi Venkatayogi System Developer II (520) 745-9447 x4466 www.cyracom.com<http://www.cyracom.com/> Join us: [cid:image001.gif@01D1C9C9.C4E8B6D0] <https://www.facebook.com/pages/CyraCom-LLC/134704783312720> [cid:image002.gif@01D1C9C9.C4E8B6D0] <https://twitter.com/cyracom> [cid:image003.gif@01D1C9C9.C4E8B6D0] <http://www.linkedin.com/company/cyracom> [cid:image004.gif@01D1C9C9.C4E8B6D0] <http://www.cyracom.com/blog/> [cid:image005.png@01D1C9C9.C4E8B6D0] DISCLAIMER: This e-mail and any attached content may contain confidential or privileged material delivered for the sole use of the intended recipient(s). Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender immediately by reply e-mail and delete all copies of this message. It is the recipient's responsibility to scan this e-mail and any attachments for viruses. The content of this e-mail message, including any attachments, does not comprise a contract or a portion of a contract, and so does not bind CyraCom International, Inc. or any of its agents or subsidiaries. CyraCom, LLC and Voiance Language Services, LLC are wholly owned subsidiaries of CyraCom International, Inc.

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Kamailio KEMI / Python and the usage of KSR.pv.get("$rm") vs msg.Method
by Samuel F. 04 May '17

04 May '17

Hi, I see in the examples that KSR.pv.get("$rm") is frequently used rather than the more simple msg.Method. Why is this preferred? Kind regards, Samuel F

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Freeswitch 403 Forbidden on invite
by Ravi sanyal 04 May '17

04 May '17

Hello, I'm new to using freeswitch. I'm using a client called SIPJS to communicate with freeswitch. The SIP message and it's response: INVITE sip:1001@104.130.11.232 SIP/2.0 Via: SIP/2.0/WS CODE;branch=BRANCH Max-Forwards: 70 To: <sip:1001@104.130.11.232> From: "1000" <sip:1000@104.130.11.232>;tag=7b5phc9nhb Call-ID: sun3fqjm7nb7rs06dbuu CSeq: 3769 INVITE Contact: <sip:k86b2npa@v9figramd1qs.invalid;transport=ws;ob> Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER Supported: outbound User-Agent: SIP.js/0.7.7 Content-Type: application/sdp Content-Length: 759 v=0 o=- 84832118975142245 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE data a=msid-semantic: WMS m=application 49422 DTLS/SCTP 5000 c=IN IP4 111.69.178.179 a=candidate:1520298561 1 udp 2113937151 192.168.178.32 49422 typ host generation 0 network-cost 50 a=candidate:900384389 1 udp 2113939711 2406:e007:2f57:1:e951:8d6e:71bd:cff8 49423 typ host generation 0 network-cost 50 a=candidate:842163049 1 udp 1677729535 111.69.178.179 49422 typ srflx raddr 192.168.178.32 rport 49422 generation 0 network-cost 50 a=ice-ufrag:6Jk+ a=ice-pwd:XDsf8/lOfUpw8f3CQas5srLl a=fingerprint:sha-256 23:DF:44:BC:1B:A5:B3:77:E0:ED:54:F9:A1:EE:CB:A1:99:FC:3A:0F:A2:2B:60:07:9A:F9:2D:12:BB:5B:60:49 a=setup:actpass a=mid:data a=sctpmap:5000 webrtc-datachannel 1024 SIP/2.0 403 Forbidden Via: SIP/2.0/WS CODE;branch=BRANCH;received=111.69.178.179;rport=64431 From: "1000" <sip:1000@104.130.11.232>;tag=7b5phc9nhb To: <sip:1001@104.130.11.232>;tag=gt0m2j3aa4m1K Call-ID: sun3fqjm7nb7rs06dbuu CSeq: 3769 INVITE User-Agent: FreeSWITCH-mod_sofia/1.9.0-380-c66a012~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 I thought to try get the invite to happen after subscribe events had taken place however the same messages are received back. This leads me to think that it's something to do with my freeswitch installation. Register messages are going through fine. that tells me that the users are being authenticated when talking to freeswitch. Nathan -- *Virtual Guard Ltd* *info(a)virtualguard.co.nz <info(a)virtualguard.co.nz>*

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How does dlg_bridge work?
by Pranathi Venkatayogi 03 May '17

03 May '17

I have TEXT call from cust1@somedomain to Kamailio. I have an agent who is logged in, but is not in any call yet. I want now refer cust1@somedomain to talk to agent directly. How do I do that? I tried - dlg_bridge("$dlg(from_contact)", "$var(agentSipAddress)",""); I get an "audio call" from "Kamailio" to customer, but nothing happens on the agent side. I tried "dlg_refer" as well but nothing happens at all. I am attaching Kamailio log / my config files for reference. Should I handcraft "Refer" message using uac module and send to "customer" instead? Please suggest. Thanks Pranathi Venkatayogi System Developer II (520) 745-9447 x4466 www.cyracom.com<http://www.cyracom.com/> Join us: [cid:image001.gif@01D1C9C9.C4E8B6D0] <https://www.facebook.com/pages/CyraCom-LLC/134704783312720> [cid:image002.gif@01D1C9C9.C4E8B6D0] <https://twitter.com/cyracom> [cid:image003.gif@01D1C9C9.C4E8B6D0] <http://www.linkedin.com/company/cyracom> [cid:image004.gif@01D1C9C9.C4E8B6D0] <http://www.cyracom.com/blog/> [cid:image005.png@01D1C9C9.C4E8B6D0] DISCLAIMER: This e-mail and any attached content may contain confidential or privileged material delivered for the sole use of the intended recipient(s). Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender immediately by reply e-mail and delete all copies of this message. It is the recipient's responsibility to scan this e-mail and any attachments for viruses. The content of this e-mail message, including any attachments, does not comprise a contract or a portion of a contract, and so does not bind CyraCom International, Inc. or any of its agents or subsidiaries. CyraCom, LLC and Voiance Language Services, LLC are wholly owned subsidiaries of CyraCom International, Inc.

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Unknown reason for 500 I'm terribly sorry, server error occurred
by Pranathi Venkatayogi 03 May '17

03 May '17

I am encountering "500 I'm terribly sorry, server error occurred" . However I don't see this message in the Kamailio.log or anything related to find the root cause. I am attaching herewith the packet capture and the entire Kamailio log and the config files. Can someone please shed some light how do I go about debugging this? Pranathi Venkatayogi System Developer II (520) 745-9447 x4466 www.cyracom.com<http://www.cyracom.com/> Join us: [cid:image001.gif@01D1C9C9.C4E8B6D0] <https://www.facebook.com/pages/CyraCom-LLC/134704783312720> [cid:image002.gif@01D1C9C9.C4E8B6D0] <https://twitter.com/cyracom> [cid:image003.gif@01D1C9C9.C4E8B6D0] <http://www.linkedin.com/company/cyracom> [cid:image004.gif@01D1C9C9.C4E8B6D0] <http://www.cyracom.com/blog/> [cid:image005.png@01D1C9C9.C4E8B6D0] DISCLAIMER: This e-mail and any attached content may contain confidential or privileged material delivered for the sole use of the intended recipient(s). Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender immediately by reply e-mail and delete all copies of this message. It is the recipient's responsibility to scan this e-mail and any attachments for viruses. The content of this e-mail message, including any attachments, does not comprise a contract or a portion of a contract, and so does not bind CyraCom International, Inc. or any of its agents or subsidiaries. CyraCom, LLC and Voiance Language Services, LLC are wholly owned subsidiaries of CyraCom International, Inc.

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rtpengine cpu usage
by Diego Nadares 03 May '17

03 May '17

Hi guys, I'm using rtpengine (compiled from source) and I see a lot of difference in cpu usage comparing with rtpproxy. Is this normal? With 600 calls: cpu *41.7%* rtpengine #rtpengine threads 4 # lscpu Architecture: x86_64 CPU op-mode(s): 32-bit, 64-bit Byte Order: Little Endian *CPU(s): 2* On-line CPU(s) list: 0,1 Thread(s) per core: 1 Core(s) per socket: 1 Socket(s): 2 NUMA node(s): 1 Vendor ID: GenuineIntel CPU family: 6 Model: 26 Model name: Intel(R) Xeon(R) CPU X7560 @ 2.27GHz Stepping: 4 CPU MHz: 2259.955 BogoMIPS: 4522.00 Hypervisor vendor: VMware Virtualization type: full L1d cache: 32K L1i cache: 32K L2 cache: 256K L3 cache: 24576K NUMA node0 CPU(s): 0,1 Calls established with a=rtpmap:0 PCMU/8000 Thanks in advance! Diego.

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if condition causes the kamailio 4.4.1 to not start
by David Escartín Almudévar 03 May '17

03 May '17

hello all we have seen today something quite weird having this line in the kamailio.cfg (1) else if (status =~ "4" || status =~ "5" || status =~ "6") { causes the process not to start hacing this error May 3 10:28:14 xxxx /usr/local/kamailio/sbin/kamailio[2896]: ERROR: <core> [io_wait.h:459]: io_watch_add(): epoll_ctl failed: Invalid argument [22] May 3 10:28:14 xxxx /usr/local/kamailio/sbin/kamailio[2896]: CRITICAL: <core> [tcp_main.c:4432]: tcp_main_loop(): failed to add process 37 unix socket to the fd list May 3 10:28:14 xxxx /usr/local/kamailio/sbin/kamailio[2896]: CRITICAL: <core> [tcp_main.c:4523]: tcp_main_loop(): exiting... but placing (2) } else if (status =~ "^[456]") { instead, makes it work this happens with kamailio 4.4.1 and the platform is running on centos6.5 on the other hand we have the same line (1) on an ubuntu 10.0.4 makes the kamailio also start properly could you know why this is happening? best regards david

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Re: [SR-Users] topos and PRACK problem
by Sergey Basov 03 May '17

03 May '17

Hi, Daniel. Seems I found how to fix PRACK handling. It works for me. please lock at https://github.com/kamailio/kamailio/pull/1097 Thank you. -- Best regards, Sergey Basov e-mail: sergey.v.basov(a)gmail.com 2017-04-27 13:04 GMT+03:00 Sergey Basov <sergey.v.basov(a)gmail.com>: > Yes, you are right. > > But now before 200 OK there empty field b_contact. > > May be you does not populate it from contact in 183 Progress? > > I see that this field is not empty only after 200 OK with a Contact field. > -- > Best regards, > Sergey Basov e-mail: sergey.v.basov(a)gmail.com > > > 2017-04-27 12:37 GMT+03:00 Daniel-Constantin Mierla <miconda(a)gmail.com>: >> Hello, >> >> thanks for troubleshooting further. I haven't got the time to look at >> the source code, but I expect that the b-leg attributes (contact, record >> routes) to be set on 200ok for dialog (topos_d). I think for PRACK, the >> routing information should be stored and taken from transaction (topos_t). >> >> Cheers, >> Daniel >> >> On 27.04.17 11:29, Sergey Basov wrote: >>> Hi, Daniel. >>> >>> I just done one more test topos with re-invite which comes from caller >>> to callee, same direction as PRACK. >>> >>> And I found that, in case of prack, b_contact field is empty... >>> >>> Please find attached debug=3 part of re-invite message. >>> >>> Hope it helps. >>> -- >>> Best regards, >>> Sergey Basov e-mail: sergey.v.basov(a)gmail.com >>> >>> >>> 2017-04-27 10:30 GMT+03:00 Sergey Basov <sergey.v.basov(a)gmail.com>: >>>> Hi, Daniel >>>> >>>> Please look at attached part of debug=3 while receiving and parsing PRACK. >>>> At line 208 seems rr module does not find correct part of record_route >>>> which in DB consists from 2 parts. >>>> >>>> So at line 219 and later uac module cannot restore uris. >>>> >>>> Thank you. >>>> -- >>>> Best regards, >>>> Sergey Basov e-mail: sergey.v.basov(a)gmail.com >>>> >>>> >>>> 2017-04-26 17:08 GMT+03:00 Sergey Basov <sergey.v.basov(a)gmail.com>: >>>>> Thanks for workaround. >>>>> >>>>> But I will wait for you solution ) >>>>> >>>>> I ready for testing ) >>>>> >>>>> Thank you Daniel for your work! >>>>> >>>>> -- >>>>> Best regards, >>>>> Sergey Basov e-mail: sergey.v.basov(a)gmail.com >>>>> >>>>> >>>>> 2017-04-26 16:57 GMT+03:00 Daniel-Constantin Mierla <miconda(a)gmail.com>: >>>>>> Hello, >>>>>> >>>>>> >>>>>> On 26.04.17 14:53, Sergey Basov wrote: >>>>>>> Hi All. >>>>>>> >>>>>>> I have just try to test topos with GW which requires PRACK. >>>>>>> >>>>>>> As you can see UA at packet 21 send PRACK to topos contact, but after >>>>>>> topos, on other kamailio side in PRACK request line present not >>>>>>> kontact but record-route header. >>>>>>> >>>>>>> Can you fix it? >>>>>>> >>>>>>> >>>>>> probably needs to look into the code. If you need a quick workaround, >>>>>> try to remove Supported header from INVITE so the callee should no >>>>>> longer Require 100rel. >>>>>> >>>>>> Cheers, >>>>>> Daniel >>>>>> >>>>>> -- >>>>>> Daniel-Constantin Mierla >>>>>> www.twitter.com/miconda -- www.linkedin.com/in/miconda >>>>>> Kamailio Advanced Training - May 22-24 (USA) - www.asipto.com >>>>>> Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> Kamailio (SER) - Users Mailing List >>>>>> sr-users(a)lists.kamailio.org >>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> >> -- >> Daniel-Constantin Mierla >> www.twitter.com/miconda -- www.linkedin.com/in/miconda >> Kamailio Advanced Training - May 22-24 (USA) - www.asipto.com >> Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com >>

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acces $dlg_var in failure_route
by Diego Nadares 03 May '17

03 May '17

Hi guys, I'm trying to access dialog vars in failure_route with no success. Dialog exists but vars are null and I'can modify them either. Is there any restriction? if ($dlg_var(v1) == 0){ $dlg_var(v1) = "1"; if (is_known_dlg()){ xlog("L_NOTICE", "negative_reply $ci $T_rpl($si) $dlg_var(v2)"); $dlg_var(v2) = $T_rpl($si); } } Log Apr 26 12:36:26 drwsip02 /usr/local/sbin/kamailio[27223]: NOTICE: <script>: negative_reply 6daf5e8d-f47972c1(a)172.31.4.21 * <null>*

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Oracle and package function
by Diego Nadares 03 May '17

03 May '17

Hi list, I compiled db_unixodbc and sqlops to connect with oracle db. I have a function that has dml inside. Because of this I can't just execute 'select pkg.function(a, b) from dual'. I'm trying with BEGIN / END. The row is inserted but always are log errors: Apr 29 02:41:57 dwsipm03 /usr/local/sbin/kamailio[7868]: ERROR: db_unixodbc [res.c:60]: db_unixodbc_get_columns(): no columns returned from the query Apr 29 02:41:57 dwsipm03 /usr/local/sbin/kamailio[7868]: ERROR: db_unixodbc [res.c:261]: db_unixodbc_convert_result(): getting column names failed Apr 29 02:41:57 dwsipm03 /usr/local/sbin/kamailio[7868]: ERROR: db_unixodbc [dbase.c:211]: db_unixodbc_store_result(): failed to convert result Apr 29 02:41:57 dwsipm03 /usr/local/sbin/kamailio[7868]: ERROR: <core> [db_query.c:188]: db_do_raw_query(): error while storing result Apr 29 02:41:57 dwsipm03 /usr/local/sbin/kamailio[7868]: ERROR: sqlops [sql_api.c:265]: sql_do_query(): cannot do the query [DECLARE ret CURSOR] My query: sql_query("cb","DECLARE ret CURSOR; BEGIN ret := pkg_partitioning.insert(True,'1'); END;"); I tried with other functions and no matter what type returns the oracle function; always logs errors. It's like unixodbc is waiting for a result with rows. Any hint? Thanks in advance, Diego

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Kamailio 4.4 NAT issue, nathelper I can not get to work
by Jason Davies 03 May '17

03 May '17

Hello I have my servers behind a NAT, with servers that make a call with bypass media / direct media all works fine. But when my media servers need to process audio I'm having issues. Looking at the SDP my local IP of the server is in there. I changed my config to add the nethelper.so and add the below: # ----- NAT Helper ----- #!ifdef WITH_NAT #modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1:7722") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) #modparam("nathelper", "sipping_bflag", 7) #modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") #modparam("registrar|nathelper", "received_avp", "$avp(i:80)") #modparam("usrloc", "nat_bflag", 6) #!endif In my route I added: if (search("User-Agent: FS Box") {fix_nated_sdp("3");};

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TLS SIP packet tracing and visualization
by Giovanni Maruzzelli 03 May '17

03 May '17

Hello fellows, after some experimentation with various tools, I come out with a little shell tool that maybe can be useful to you too. It can only work with non-forward secrecy ciphers, obviously, and only if is started before the client do the initial TLS handshake (eg, just restart the client). Forward secrecy cannot be decrypted after fact, so don't waste effort. An example of ciphers that can be decrypted are the "AES256-SHA" openssl cipher group. You can use ssldump to check what cipher is used by serverhello. Enjoy, make it better, and share it :) #!/bin/bash # brought to you by Giovanni Maruzzelli # SERVERIP="192.168.1.150" SERVERPORT="5061" PRIVKEY="/etc/certs/privkey.pem" STDERR2DEVNULL=" 2>/dev/null " REGEX="notyet" if [ -z "$1" ]; then REGEX="\\\.*" else REGEX="$1" fi FILTER="ssl.app_data and sip matches" FILTER2="$FILTER \"$REGEX\"" FILTER3="'$FILTER2'" ARGUMENT="-i 1 -Y $FILTER3 -E header=y -T fields -e frame.number -e frame.time -e frame.time_delta_displayed -e ip.src -e ip.dst -e sip.Status-Line -e sip.Request-Line -e sip.msg_hdr -l -d tcp.port\=\=5061,sip -o \"ssl.keys_list: $SERVERIP,$SERVERPORT,sip,$PRIVKEY\" $STDERR2DEVNULL | sed -u 's/\t/\n/g' | sed -u '/^$/d' | sed -u 's/^[0-9]*$/\n==&==============================/g'" echo "" echo "NB: if it do not works, edit script so that STDERR2DEVNULL=\" \" and try again" echo "" echo "NB: remember to quote and escape match patterns, using triple slash" echo " eg, for matching 1010(a)pbx.example.com, use \"1010(a)pbx.example.com \"" echo " eg, for matching anything, use \"\\\\\\.*\"" echo " eg, for matching *98, use \"\\\\\\*98\"" echo "USAGE: $0 \"\\\\\\*98(a)pbx.example.com\"" echo "" case "$1" in -help|--help|?) exit 0 ;; *) echo "THIS TIME WE'RE DOING:" echo "tshark $ARGUMENT" echo "" bash -c "tshark $ARGUMENT" ;; esac -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18

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Memory leak when reloading htable using db_cluster
by Kristian F. Høgh 03 May '17

03 May '17

Hi list, I use current git master. When I reload a htable using "kamcmd htable.reload htable1", the "ctl handler" process leaks 384 bytes of pkg memory If I use a direct mysql connection without db_cluster, the is no memory leak modparam("db_cluster", "connection", "con1=>mysql://user:pass@ip.addr/database") modparam("db_cluster", "connection", "con2=>mysql://user:pass@ip.addr2/database") modparam("db_cluster", "cluster", "cls1=>con1=9s9s;con2=8s8s") modparam("htable", "db_url", "cluster://cls1") # modparam("htable", "db_url", "mysql://user:pass@ip.addr/database" modparam("htable", "htable", "htable1=>size=8;autoexpire=0;dbtable=htable1;") 16(10429) ALERT: qm_status: 6513. N address=0x7f4e65e01720 frag=0x7f4e65e016e8 size=128 used=1 16(10429) ALERT: qm_status: alloc'd from db_cluster: dbcl_api.c: db_cluster_init(294) 16(10429) ALERT: qm_status: start check=f0f0f0f0, end check= c0c0c0c0, abcdefed 16(10429) ALERT: qm_status: 6514. N address=0x7f4e65e01808 frag=0x7f4e65e017d0 size=128 used=1 16(10429) ALERT: qm_status: alloc'd from core: db.c: db_do_init2(298) 16(10429) ALERT: qm_status: start check=f0f0f0f0, end check= c0c0c0c0, abcdefed 16(10429) ALERT: qm_status: 6515. N address=0x7f4e65e018f0 frag=0x7f4e65e018b8 size=128 used=1 16(10429) ALERT: qm_status: alloc'd from core: db.c: db_do_init2(298) 16(10429) ALERT: qm_status: start check=f0f0f0f0, end check= c0c0c0c0, abcdefed Regards, Kristian Høgh Uni-tel A/S

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SIP over WebSockets
by Nicolas Pace 02 May '17

02 May '17

Hi everyone, As I shared with you before, We are working on a mobile app to connect the Community Cellular Networks (that uses 2G Cellular Networks) with the rest of the world via SIP, and we are considering using Kamailio for this. We choosed to build a Progresive Web App (PWA) through WebSockets to reduce the friction involved in using a new tool. My question is, is there a reference config to use a web client through WebSockets? I don't have a clue on how to get from the stock config: https://gist.github.com/nicopace/9d44bf813fe309df69ff4cd0affe8324#file- _etc_kamailio_kamailio-conf to a fully functional WebSocket-enabled config. I've already read the WebSocket module documentation, but it is hard to understand as it assumes you already know how Kamailio works... too many moving parts :( I tried with https://github.com/caruizdiaz/kamailio-ws but it is giving me some errors: May 02 13:37:13 ubuntu kamailio[2329]: : <core> [cfg.y:3432]: yyerror_at(): parse error in config file /etc/kamailio/modules.cfg, line 48, column 12-25: failed to load module May 02 13:37:13 ubuntu kamailio[2329]: ERROR: <core> [cfg.y:3295]: yyparse(): cfg. parser: failed to find command ws_handle_handshake (params 0) May 02 13:37:13 ubuntu kamailio[2329]: : <core> [cfg.y:3435]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 143, column 41: unknown command, missing loadmodule? Which relates with: https://github.com/caruizdiaz/kamailio-ws/blob/master/modules.cfg#L48 and https://github.com/caruizdiaz/kamailio-ws/blob/master/kamailio-ws.cfg#L 143 What I'm trying to accomplish is to have a PWA that sends SIMPLE messages and CALLS to and from SIP Users (the 2G users are behind a FreeSwitch, so it could be done). Just to give you some context, we will also add Push Notifications as 'ringing' or 'meesage recieved' alert, and that would be all (we also don't know how to do it). Well... hope to hear from you guys! Thanks!

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TCP read/write buffer size on per connection or per module basis
by M S 02 May '17

02 May '17

Hi, I have a kamailio v4.4.5 deployment that serves TCP clients for SIP and XCAP requests (without SIP presence). The kamailio's core TCP parameters allow us to set TCP read and write buffer sizes which are set globally for all TCP connections. However, i noticed that in general SIP over TCP request is much smaller (<8KB for audio + video SIP INVITE request) then XCAP requests (say to support up to 500 contacts we need TCP buffers =~ 64KB). So, if I set global TCP buffer size according to XCAP service needs (e.g. 64KB), they waste too much RAM for SIP connections (assuming max SIP packet is 8KB, there is 56KB buffer space wasted per connection). On the other hand, if i set global TCP buffer sizes to SIP service needs (e.g. 8KB) then it is insufficient for XCAP (only 70-80 contacts would fix in XCAP list). Therefore, i am looking for a way to allow TCP buffer size settings per connection OR per module basis. Is it possible? If so then how? Thank you.

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Retrieving the custom header parameter value from contact header
by vivek＠advaitamtech.com 02 May '17

02 May '17

Hi, Am trying to extract a custom parameter from the contact header. Below it the example of contact header, Eg: Contact: <sip:fayazat@XXX.XXX.XXX.XXX:5060;tgrp=tg_0615_prsw1> I want to extract the parameter value of "tgrp" Tried using @hf_value.contact.p.tgrp . But it is giving me complete contact header. I want only "tgrp" paramter value "tg_0615_prsw1". Anyone , any sugeestions? Thanks, Vivek.

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TCP port re-using when doing HTTP transfer
by Ramachandran, Agalya (Contractor) 02 May '17

02 May '17

Hi Giacomo, I am using http_async_client module to do http transfer. Currently am testing the HTTP service with a mock server. If I send the HTTP response immediately for the POST request, then the TCP connections are reused from kamailio. In the mock server if I send the response with a delay of 1 sec then, TCP port are opened and closed for every HTTP transfer. Is there a way or any configuration to limit the number of TCP port opening for the HTTP transfers? Please clarify me. Regards, Agalya

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Kamailio IMS - Handling 2 SIP transports in an IMS client
by Haggerty, Paul (phaggert) 01 May '17

01 May '17

Hi, Our client registers with the IMS core using UDP but uses TCP when the packet size is over 1300 bytes. When the client sends the TCP reques, the PCSCF replies with 403 Forbidden "you need to reg with the SCSCF" We've traced this back to the code in the PCSCF and it looks like it is checking the via headers and the received ip/port combinations for a match. The actual contact header looks like it matches the registered client. Question is what do we need to do in the client to ensure that the PCSCF recognizes the request as one from a registered client? We've tried the config options is_registered_fallback2ip , ignore_contact_rxport_check and they don't seem to be used. I'm attaching a network trace and included here are some debug messages from the PCSCF. In the network trace the client is 10.8.0.6. PCSCF Trace messages ------------------------------------------- 34207) DEBUG: ims_registrar_pcscf [service_routes.c:208]: getContactP(): This is a request - using first via to find contact 10(34207) DEBUG: ims_registrar_pcscf [service_routes.c:216]: getContactP(): searching for contact with host:port:proto contact [2://10.8.0.6:49946] 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:457]: get_pcontact(): Searching for contact with AOR [sip:mcpttuser1@10.8.0.6:44580;ob;alias=10.8.0.6~44580~1;alias=10.8.0.6~49946~2] in P-CSCF usrloc based on VIA [2://10.8.0.6:49946] Received [2://10.8.0.6:49946], Search flag is 0 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:463]: get_pcontact(): Have an AOR to search for 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:468]: get_pcontact(): checking for rinstance10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:480]: get_pcontact(): no rinstance param 10(34207) DEBUG: ims_usrloc_pcscf [usrloc.c:148]: get_aor_hash(): Returning hash: [1662767777] 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:503]: get_pcontact(): get_pcontact slot is [161] 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:507]: get_pcontact(): comparing contact with aorhash [1662767777], aor [sip:mcpttuser1@10.8.0.6:44580;ob;alias=10.8.0.6~44580~1] 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:508]: get_pcontact(): contact host [10.8.0.6:44580] 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:509]: get_pcontact(): contact received [1:10.8.0.6:44580] 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:543]: get_pcontact(): contact not found in memory 10(34207) INFO: ims_registrar_pcscf [service_routes.c:258]: getContactP(): Contact not found based on Contact-header, trying IP/Port/Proto 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:457]: get_pcontact(): Searching for contact with AOR [sip:mcpttuser1@10.8.0.6:44580;ob;alias=10.8.0.6~44580~1;alias=10.8.0.6~49946~2] in P-CSCF usrloc based on VIA [2://10.8.0.6:49946] Received [2://10.8.0.6:49946], Search flag is 1 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:463]: get_pcontact(): Have an AOR to search for 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:468]: get_pcontact(): checking for rinstance10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:480]: get_pcontact(): no rinstance param 10(34207) DEBUG: ims_usrloc_pcscf [usrloc.c:148]: get_aor_hash(): Returning hash: [1662767777] 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:503]: get_pcontact(): get_pcontact slot is [161] 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:507]: get_pcontact(): comparing contact with aorhash [1662767777], aor [sip:mcpttuser1@10.8.0.6:44580;ob;alias=10.8.0.6~44580~1] 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:508]: get_pcontact(): contact host [10.8.0.6:44580] 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:509]: get_pcontact(): contact received [1:10.8.0.6:44580] 10(34207) DEBUG: ims_usrloc_pcscf [udomain.c:543]: get_pcontact(): contact not found in memory 10(34207) DEBUG: ims_registrar_pcscf [service_routes.c:263]: getContactP(): No entry in usrloc for 10.8.0.6:49946 (Proto 2) found! 10(34207) DEBUG: ims_registrar_pcscf [service_routes.c:300]: getContactP(): Asserted identity not set10(34207) ERROR: *** cfgtrace:request_route=[Orig_Initial] c=[/opt/OpenEPC/etc/pcscf2.cfg] l=975 a=26 n=xlog 10(34207) WARNING: <script>: 2.1 10(34207) ERROR: *** cfgtrace:request_route=[Orig_Initial] c=[/opt/OpenEPC/etc/pcscf2.cfg] l=976 a=26 n=send_reply Appreciate any help we can get, - Paul

1 0

handle_uri_alias persists through failure_route
by David Wilson 01 May '17

01 May '17

Hello All, We use registrar servers (call them A,B) for TLS connections from clients and a central routing proxy (P) that authenticates and manages all call routing. On INVITE, P uses lookup(“location”) and handle_uri_alias() to route the INVITE to A or B depending on the AOR from REGISTER. In failure_route I’m trying to send INVITE to a voicemail URI at host (V), using alias_db_lookup to retrieve it and sending with t_relay(). However, despite showing V as the host in the URI, the message is sent to A (or B) as if the handle_uri_alias is still in effect. I’ve tried calling revert_uri() before the alias_db_lookup(), but this doesn’t seem to help. Is there a way to reset the uri_alias? The failure_route (simplified) looks like this: failure_route[1] { if (method == "INVITE") { revert_uri(); if (alias_db_lookup("dbaliases")) { t_relay(); } } } I have also tried using t_relay(“$rd”, “$rp”) after confirming (by logging) that the uri contains the desired values, but kamailio failed to start, with this error logged: ERROR: tm [tm.c:672]: fixup_hostport2proxy(): TM module:fixup_hostport2proxy: bad port number <$rp> Any suggestions? Best regards, Dave.

3 3

Cannot get service to start
by Stephen Tyers 28 Apr '17

28 Apr '17

Hi Apologies, I was a bit hasty in my previous email, I forgot to run the Chown command I still can’t get the service to start and I’m getting the below error. I’ve gone through all the steps multiple times, please help. /etc/init.d/kamailio start [....] Starting kamailio (via systemctl): kamailio.serviceFailed to start kamailio.service: Unit kamailio.service failed to load: No such file or directory. failed! Kind regards Stephen tyers

2 1

Date Time with microsec
by Diego Nadares 28 Apr '17

28 Apr '17

Hi Guys, Is to convert or obtain $TV(s) - seconds (cached at first call per sip message) $TV(u) - microseconds (cached at first call per sip message) to datetime.microsecs? I'm trying to save 180/183 datetime (cached) like YYYY-MM-DD HH:mm:SS.microsecs $dlg_var(ring_time) = $timef(%Y-%m-%d %H:%M:%S); <--this is current time, not cached $dlg_var(ring_time_micro) = $TV(u); <-- this is cached us Thanks in advance! Diego.

2 3

htable Arrays and dbmode 1
by Sebastian Damm 28 Apr '17

28 Apr '17

Hi, I'm trying to use a htable array. For that, I have inserted some values in the database (in pseudo CSV): key_name; key_type; value_type; key_value '1234567'; 1; 0; 'foo' '1234567', 1, 0, 'bar' '1234567', 1, 0, 'baz' insert into htable (key_name , key_type , value_type , key_value ) values ('1234567', 1, 0, 'foo'),('1234567', 1, 0, 'bar'),('1234567', 1, 0, 'baz'); Now after just starting Kamailio, reading those values, from db and then stopping Kamailio again, the table content looks like this: key_name; key_type; value_type; key_value '1234567[2]'; 0; 0; 'baz' '1234567[0]', 0, 0, 'foo' '1234567[1]', 0, 0, 'bar' '1234567::size', 0, 1, 3 Is this how it is supposed to work? When re-reading those entries, the htable contents look the same as during the first start, but in the database it is a bit harder to read. And while I'm at it: Is there a way to regularly write back the current htable contents to the database? Or even to sync every write directly into the database? Thanks in advance Sebastian

1 0

Not able to replace route_uri in contact header uri
by Narayan P 28 Apr '17

28 Apr '17

Hi All, I am using pjsip client and kamailiio,both running on different servers.I want to replace the uri on which pjsip is running with the uri on which kamailio is running in contact header. I put some logs on route[REGISTRAR] and able to see the $route_uri as kamailio server uri($avp(s:fs) and $hdr(Contact) as pjsip server uri ($avp(s:contact).But I am not able to replace the uri with avp_subst() function. Can anybody help me how to replace the uri or with any any other function. Below I have put the snippet of config file. # Handle SIP registrations route[REGISTRAR] { if (!is_method("REGISTER")) return; if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); #!ifdef WITH_NATSIPPING # do SIP NAT pinging setbflag(FLB_NATSIPPING); #!endif } xlog("Narayan: before sending contact\n"); $avp(s:contact) = $hdr(Contact); $avp(s:fs) = $route_uri; xlog("Narayan: Forced socket is $avp(s:fs)\n"); xlog("Narayan: contact header is $avp(s:contact)\n"); avp_subst("$avp(s:contact)/avp(s:contac)/g", "/(.*)(a)(.*)/$route_uri/"); #avp_pushto("$ru/domain","$fd"); #rewritehostport("185.122.206.62:5060"); t_on_reply("MANAGE_REPLY"); t_relay(); #if (!save("location")) { # sl_reply_error(); #} exit; } Any suggestion will be highly appreciated. Thanks, Narayan

3 11

Problem with installing Kamailio 5
by Stephen Tyers 28 Apr '17

28 Apr '17

Hi, I’ve hit a road block when following URL http://kamailio.org/docs/tutorials/5.0.x/kamailio-install-guide-git/ I’m very new to open source and linux so please bare with me. I’ve reached step Kamailio start and I get the below error. Please help, I’ve think I’ve followed all the proceeding commands and created the databases. root@kamailio:/var/run# /etc/init.d/kamailio start -bash: /etc/init.d/kamailio: Permission denied Many thanks Steve

1 0

An Potential Issue in Kamailio Installation Guide
by Edward Lu 28 Apr '17

28 Apr '17

Dear Sir, This is Edward, nice to meet you. When I follow the installation guide of Kamailio 5.0.x https://kamailio.org/docs/tutorials/5.0.x/kamailio-install-guide-git/ to install Kamailio, I met an issue, it report "unit kamailio.service not found" when I issue command /etc/init.d/kamailio start. After google, I found one step was missed: to copy kamailio.service to /etc/system.d/system. Except above issue, all others are all okay for my installation on Ubuntu 16.04. Thanks & Regards, Edward

1 0

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