- sr-users - kamailio.org

Wrong Publish status
by Aleš Šturm 06 Jun '16

06 Jun '16

Hello, Kamailio 4.3: I am testing Presence module with more complex scenario. I have 2 SIP clients (presentity) with equal identities (3918) and one watcher (2025). All my clients are Monster 0.9.25. My Kamailio Presence module is configured with "retrieve_order=1" (presentity records are retrieve by priority order). I have situation when wrong status is send to the watcher (2025). 1. First all three clients registers and publish its own status. Below is print from my MySQL database (presentity table) mysql> select id,username,priority,expires,received_time,body from presentity ORDER BY received_time DESC; | id | username | priority | expires | received_time | body | 84481 | 3918 | 40 | 1464340695 | 1464337095 | <?xml version='1.0' encoding='utf-8' ?><presence entity="sip:3918@192.168.77.43" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpidf="urn:ietf:params:xml:ns:pidf:rpid" xmlns:op="urn:oma:xml:prs:pidf:oma-pres" xmlns:opd="urn:oma:xml:pde:pidf:ext"><tuple id="com1"><status><basic>open</basic></status><op:willingness><opd:basic>ope n</opd:basic></op:willingness></tuple><dm:person id="per123"><op:overriding-willingness><opd:basic>open</opd:basic></op:overr iding-willingness></dm:person></presence> | | 84480 | 3918 | 40 | 1464340687 | 1464337087 | <?xml version='1.0' encoding='utf-8' ?><presence entity="sip:3918@192.168.77.43" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpidf="urn:ietf:params:xml:ns:pidf:rpid" xmlns:op="urn:oma:xml:prs:pidf:oma-pres" xmlns:opd="urn:oma:xml:pde:pidf:ext"><tuple id="com1"><status><basic>open</basic></status><op:willingness><opd:basic>ope n</opd:basic></op:willingness></tuple><dm:person id="per123"><op:overriding-willingness><opd:basic>open</opd:basic></op:overr iding-willingness></dm:person></presence> | | 84479 | 2025 | 40 | 1464340682 | 1464337082 | <?xml version='1.0' encoding='utf-8' ?><presence entity="sip:2025@192.168.77.43" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpidf="urn:ietf:params:xml:ns:pidf:rpid" xmlns:op="urn:oma:xml:prs:pidf:oma-pres" xmlns:opd="urn:oma:xml:pde:pidf:ext"><tuple id="com1"><status><basic>open</basic></status><op:willingness><opd:basic>ope n</opd:basic></op:willingness></tuple><dm:person id="per123"><op:overriding-willingness><opd:basic>open</opd:basic></op:overr iding-willingness></dm:person></presence> | 2. Then first 3918 client publishes his mood >11111<. The status in >Presentity< table is refreshed under the id=84480. NOTIFY is sent to 2025 with status with mood >11111<. mysql> select id,username,priority,expires,received_time,body from presentity ORDER BY received_time DESC; | id | username | priority | expires | received_time | body | 84480 | 3918 | 40 | 1464340703 | 1464337103 | <?xml version='1.0' encoding='utf-8' ?><presence entity="sip:3918@192.168.77.43" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpidf="urn:ietf:params:xml:ns:pidf:rpid" xmlns:op="urn:oma:xml:prs:pidf:oma-pres" xmlns:opd="urn:oma:xml:pde:pidf:ext"><tuple id="com1"><status><basic>open</basic></status><op:willingness><opd:basic>ope n</opd:basic></op:willingness></tuple><dm:person id="per123"><dm:note>11111</dm:note><op:overriding-willingness><opd:basic>op en</opd:basic></op:overriding-willingness></dm:person></presence> | | 84481 | 3918 | 40 | 1464340695 | 1464337095 | <?xml version='1.0' encoding='utf-8' ?><presence entity="sip:3918@192.168.77.43" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpidf="urn:ietf:params:xml:ns:pidf:rpid" xmlns:op="urn:oma:xml:prs:pidf:oma-pres" xmlns:opd="urn:oma:xml:pde:pidf:ext"><tuple id="com1"><status><basic>open</basic></status><op:willingness><opd:basic>ope n</opd:basic></op:willingness></tuple><dm:person id="per123"><op:overriding-willingness><opd:basic>open</opd:basic></op:overr iding-willingness></dm:person></presence> | | 84479 | 2025 | 40 | 1464340682 | 1464337082 | <?xml version='1.0' encoding='utf-8' ?><presence entity="sip:2025@192.168.77.43" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpidf="urn:ietf:params:xml:ns:pidf:rpid" xmlns:op="urn:oma:xml:prs:pidf:oma-pres" xmlns:opd="urn:oma:xml:pde:pidf:ext"><tuple id="com1"><status><basic>open</basic></status><op:willingness><opd:basic>ope n</opd:basic></op:willingness></tuple><dm:person id="per123"><op:overriding-willingness><opd:basic>open</opd:basic></op:overr iding-willingness></dm:person></presence> | 3. After that the second 3918 client publishes his mood >22222<. The status in >Presentity< table is refreshed under the right id=84481. NOTIFY is sent to 2025 with status with mood >22222<. mysql> select id,username,priority,expires,received_time,body from presentity ORDER BY received_time DESC; | id | username | priority | expires | received_time | body | 84481 | 3918 | 40 | 1464340716 | 1464337116 | <?xml version='1.0' encoding='utf-8' ?><presence entity="sip:3918@192.168.77.43" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpidf="urn:ietf:params:xml:ns:pidf:rpid" xmlns:op="urn:oma:xml:prs:pidf:oma-pres" xmlns:opd="urn:oma:xml:pde:pidf:ext"><tuple id="com1"><status><basic>open</basic></status><op:willingness><opd:basic>ope n</opd:basic></op:willingness></tuple><dm:person id="per123"><dm:note>22222</dm:note><op:overriding-willingness><opd:basic>op en</opd:basic></op:overriding-willingness></dm:person></presence> | | 84480 | 3918 | 40 | 1464340703 | 1464337103 | <?xml version='1.0' encoding='utf-8' ?><presence entity="sip:3918@192.168.77.43" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpidf="urn:ietf:params:xml:ns:pidf:rpid" xmlns:op="urn:oma:xml:prs:pidf:oma-pres" xmlns:opd="urn:oma:xml:pde:pidf:ext"><tuple id="com1"><status><basic>open</basic></status><op:willingness><opd:basic>ope n</opd:basic></op:willingness></tuple><dm:person id="per123"><dm:note>11111</dm:note><op:overriding-willingness><opd:basic>op en</opd:basic></op:overriding-willingness></dm:person></presence> | | 84479 | 2025 | 40 | 1464340682 | 1464337082 | <?xml version='1.0' encoding='utf-8' ?><presence entity="sip:2025@192.168.77.43" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpidf="urn:ietf:params:xml:ns:pidf:rpid" xmlns:op="urn:oma:xml:prs:pidf:oma-pres" xmlns:opd="urn:oma:xml:pde:pidf:ext"><tuple id="com1"><status><basic>open</basic></status><op:willingness><opd:basic>ope n</opd:basic></op:willingness></tuple><dm:person id="per123"><op:overriding-willingness><opd:basic>open</opd:basic></op:overr iding-willingness></dm:person></presence> | 4. Then finally, the first 3918 changes his mood to "33333". The status in >Presentity< table is refreshed under the right id=84480. BUT, to 2025 is sent NOTIFY with mood "22222". "received_time" timestamp of status with mood "33333" is higher than timestamp of status with mood "22222". BUT the "id" attribute of "presentity" table of status with mood "33333" is smaller. I think that statuses with equal "priority" are retrieved by "id" and not by "received_time"!? mysql> select id,username,priority,expires,received_time,body from presentity ORDER BY received_time DESC; | id | username | priority | expires | received_time | body | 84480 | 3918 | 40 | 1464340729 | 1464337129 | <?xml version='1.0' encoding='utf-8' ?><presence entity="sip:3918@192.168.77.43" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpidf="urn:ietf:params:xml:ns:pidf:rpid" xmlns:op="urn:oma:xml:prs:pidf:oma-pres" xmlns:opd="urn:oma:xml:pde:pidf:ext"><tuple id="com1"><status><basic>open</basic></status><op:willingness><opd:basic>ope n</opd:basic></op:willingness></tuple><dm:person id="per123"><dm:note>33333</dm:note><op:overriding-willingness><opd:basic>op en</opd:basic></op:overriding-willingness></dm:person></presence> | | 84481 | 3918 | 40 | 1464340716 | 1464337116 | <?xml version='1.0' encoding='utf-8' ?><presence entity="sip:3918@192.168.77.43" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpidf="urn:ietf:params:xml:ns:pidf:rpid" xmlns:op="urn:oma:xml:prs:pidf:oma-pres" xmlns:opd="urn:oma:xml:pde:pidf:ext"><tuple id="com1"><status><basic>open</basic></status><op:willingness><opd:basic>ope n</opd:basic></op:willingness></tuple><dm:person id="per123"><dm:note>22222</dm:note><op:overriding-willingness><opd:basic>op en</opd:basic></op:overriding-willingness></dm:person></presence> | | 84479 | 2025 | 40 | 1464340682 | 1464337082 | <?xml version='1.0' encoding='utf-8' ?><presence entity="sip:2025@192.168.77.43" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpidf="urn:ietf:params:xml:ns:pidf:rpid" xmlns:op="urn:oma:xml:prs:pidf:oma-pres" xmlns:opd="urn:oma:xml:pde:pidf:ext"><tuple id="com1"><status><basic>open</basic></status><op:willingness><opd:basic>ope n</opd:basic></op:willingness></tuple><dm:person id="per123"><op:overriding-willingness><opd:basic>open</opd:basic></op:overr iding-willingness></dm:person></presence> | The Wireshark trace is attached. Can anyone help. Please? We have a big installation in the Gazprom's (Rusia) network, where Kamailio is used as a Presence server only. Thanks Ales

2 8

Yealink BLF
by Bill 05 Jun '16

05 Jun '16

Has anyone been successful in getting Yealink BLF's to work with Kamailio presence It appears that the xml format being sent is not compatible with Yealink, butr I can't find how to change the format of the body. Bill

1 0

rtpproxy ipv6 problem
by aman jeff 03 Jun '16

03 Jun '16

Hi Team, The SIP packets are fine ,we are able to establish SIP session , As long the RTP Proxy not get involved and we are not able to hear audio,So How to do the proper configuration RTP Proxy,what to do in RTP Proxy to get the voice for the connected sip session Regards Arnab

1 0

Kamailio 4.4.1 REGISTER Parse Error in PCSCF
by Mihail Dakov 02 Jun '16

02 Jun '16

Hi Folks, I have the strangest problem just after installing latest stable version. I had tried both ways using the repo (http://deb.kamailio.org/kamailio44) and downloading code with local compile (https://www.kamailio.org/w/2016/05/kamailio-v4-4-1-released/) The initial scenario is to get M2M with commercial IP and subsequently use the Rx interface to do the same. We are blocked because we cannot even achieve clean register for the UEs. UE1(galaxy s4 zoiper) -------- Kamailio IMS aio | UE2(galaxy s4 zoiper) -------- | What happens is that the first register message triggers some quasi-endless loop in the PCSCF. Hence it cannot be forwarded to SCSCF, ICSCF and HSS. But this doesn't always happen and between all the parse problems there are some register messages which get passed to the scscf icscf and hss. Somehow the parser feeds itself from the previous parse run and thus loops quite intensively. The message I get is SIP/2.0 400 CSeq method does not match request method. I must say that reiterating the registration process sometimes i get the register and sometimes i don't. Has anyone encounter similar issue for similar use case and can share some thoughs about a fix? I attached relevant log file, pcaps and configuration files. Let me know if more config files are needed.

3 15

Nonce extract from SIP Meesage
by Sergio Serrano 02 Jun '16

02 Jun '16

Hi all, I just try integrate Kamailio with external software and We want taht kamailio doesn't authenticate users. I want that the autehtication process would be done by external software. I want To receive a REGISTER for a user. Then I send a 407 and then when I receive all paramters for autehtincation(nonce, response, digestURI, etc, etc etc) send that data to external software and receive if data is correct, then sent 200 OK and save location in external software. Bu t I can see any pseudovariable or enything else to obtain that dato to provide to external software. Anyone could think that this process could be done? Regards, Sergio

3 2

NSQ libevbuffsock dependency
by smititelu 02 Jun '16

02 Jun '16

Hi, Upon trying to compile latest Kamailio from sources I got an: " nsq.h:9:24: fatal error: evbuffsock.h: No such file or directory #include <evbuffsock.h> ^ " Looking on the new NSQ module documentation I see the libevbuffsock dependency. Is there other recommended way, to install the library besides compiling from sources at [1]? Thx, Stefan [1] https://github.com/mreiferson/libevbuffsock

2 2

sip trace
by Jeremy Betts 01 Jun '16

01 Jun '16

I have a new carrier that wants sip traces of certain scenarios. They also require IPSEC tunnelling for all SIP messages. So when I use tcpdump or any other similar tool I can see the incoming SIP messages but outgoing SIP messages are ESP encoded. I've played with the siptace module but can't seem to get it to log replies. Does anyone have any solutions?

2 2

Dynamic advertised ip for rtpengine/rtpproxy
by Serge Yuriev 01 Jun '16

01 Jun '16

Hi, We have several exits from our network and all of them natted. Could we set adv ip for rtpproxy/rtpengine from db in realtime or we must define them on daemon run time only? -- Wbr, Serge via mobile -- Serge via mobile

4 3

Send Register from Kamailio
by Grant Bagdasarian 01 Jun '16

01 Jun '16

Hello, Which module do I need to have Kamailio send REGISTER messages to our carrier? Is it the UAC module or REGISTRAR module? It just needs to register with our carrier in order to receive inbound calls. Regards, Grant

2 4

advertise xxx.xxx.xxx.xxx:5060 is weird
by Sébastien Brice 01 Jun '16

01 Jun '16

Hi guys, I have a local proxy running on private ip 192.168.3.97 and advertising the wan ip of my box (THEPUBLICIPOFMYBOX) this local Kamailio is a stateful proxy that TM REGISTERS and INVITES to a Kamailio registrar(PUBLICIPKAMAILIOREGISTRAR) (please have a look at the kamailio.cfg attached to this email.) I want to force the signalling and sessions through this local proxy (some kind of outbound proxy) I have jitsi and xlite behind (ip 192.168.3.39 and 192.168.3.16) that tries to call themselves. user agents keep retransmitting 200 OK because they never get the ACK following the 200 OK pickup. If i remove the advertise THEPUBLICIPOFMYBOX of the kamailio.cfg the three handshake is working ok and the call is no longer dropped after a few second! i have a true bidirectionnal session and that's working ok. The bad thing is RTP flows straight through router (thanks to rtpproxy i guess) but i want incoming and outgoing RTP traffic to go on central local node 192.168.3.97 How can i achieve that? thanks. PS: i am not using path header because i am not skilled enough and kamailio is really hard horse to train

2 5

DHCP renewal conflicts with kamailio?
by Joonas Harjumäki 01 Jun '16

01 Jun '16

Hi! The attachment contains a log trace of kamailio 3.3.2. There is some errors in the log and I wonder if it is normal operation? The errors appeared in the middle of DHCP lease renewal and I wonder if the configuration (of dhclient/kamailio) could be improved to prevent such errors? Or to make sure kamailio retries udp_send sufficiently many times. The kamailio operated normally before renewal procedure began and after the procedure finished. Joonas Harjumäki

2 1

how to set kamctl to find ctl anf fifo that are not at /var/run
by Uri Shacked 01 Jun '16

01 Jun '16

Hi, I am using centos 7. By default on centos 7, the var/run file system is tmpfs. So, when kamailio runs with user and group other than root, it can not create the fifo file. I set the kamailio service and config files to work on usr/local/var/run/kamailio. Now, how do i set the kamctl to work with the fifo on /usr/local/var/run/kamailio ? Uri

3 3

Want dialog.default_timeout NOT to reset on subseq. req
by Andrey Utkin 01 Jun '16

01 Jun '16

Hi! I'd like to restrict max call duration. Currently we have something that partially works: dialog.default_timeout. But, as described in "How it works" section http://www.kamailio.org/docs/modules/devel/modules/dialog.html#idp2029360 "The dialog timeout is reset each time a sequential request is processed." So this option works for us when the useragents don't send sequential UPDATEs or reINVITEs. I am not sure if it is adequate to ask for this feature in dialog module. I am aware of possibility to terminate any call by management command, and of cnxcc module which has call duration control. I would love to find something that simple - a single option for max call duration, without external mechanism (cronjob etc.) or full-blown billing. Any suggestions are very welcome. Thanks in advance. Please reply to all recipients, I am not subscribed to this maillist.

2 1

513 Message too big for VIDEO calls
by Varghese Paul 31 May '16

31 May '16

Hi all, I am trying to setup WSS on opensips-2.2. I am getting "513 Message too big " when i am trying to make video calls. Audio calls and Registration is working fine.I tried to change the message_chunk_size in WSS module but still i am getting the message. Client : SIPML5 and JSSIP Opensip version : 2.2 I am attaching the sipml5 console logs. Please let me know is there any settings i can apply to resole this. Thanks in Advance Varghese Paul

2 1

Extending has_body() with additional mime types
by Giacomo Vacca 31 May '16

31 May '16

Hi all, apologies if this is an old topic but I've noticed that has_body() can provide false positives and would like to discuss a solution. As a practical example, has_body("application/vq-rtcpxr") was returning 1 for Content-Type "application/dialog-info+xml". The interpretation is that they are both unknown and so they match. Using 'if ($cT == "application/vq-rtcpxr)" provides a work around, but I wonder what's the suggested solution for this? a. Add the missing mime types (from RFC 6035 and 4235 respectively) to the list b. Instead of returning 1, return another code, e.g. 2, when the mime type is unknown c. Switch to string comparison when the mime type is unknown d. Something else? Thanks, Giacomo

2 1

Accept INVITE with transport=tcp
by Nahum Nir 31 May '16

31 May '16

Hi, I have a client that sends INVITE with transport=tcp. How can I make Kamailio to work with that? Thanks and regards, Nahum

2 1

Issue with LCR an failover routers
by Alberto Sagredo 30 May '16

30 May '16

Hi Im using LCR module and trying to handle 503/408 errors on provider to route to a second one. Actually im using if(!load_gws(1, $rU, $var(caller_uri))) { xlog("Gateways no disponibles"); sl_send_reply("500", "Server Internal Error - No gateway"); exit; } else { xlog("GW Seleccionado '$avp(i:709)'\n"); xlog("Dominio de destino: $dd\n"); xlog("To URI: $tu\n"); } if(!next_gw()) { xlog("No hay GW de Backup"); sl_send_reply("503", "Service not available - No gateway"); exit; } else { xlog("Enviado por primer GW\n"); xlog("ruri_user_avp: '$avp(i:500)'\n"); xlog("To URI after next_gw: $tu\n"); xlog("Request URI: $rU\n"); t_on_failure("FAILBACK"); } failure_route[FAILBACK] { if (t_is_canceled()) { exit; } if (t_check_status("408|50[34]")) { route(LCR_OUT_OP2); } } And using for OP2 same routing block with id=3... This second providers use AUTH so using: t_on_failure("TRUNKAUTH"); failure_route[TRUNKAUTH] { if (t_is_canceled()) { exit; } if(t_check_status("401|407")) { if (to_gw("3")) { xlog("Autenticación "); $avp(auser) = "user"; $avp(apass) = "pass"; } uac_auth(); t_relay(); exit; } But getting this error: May 30 09:59:05 kamailio /usr/local/sbin/kamailio[21602]: ERROR: tm [t_fwd.c:1704]: t_forward_nonack(): ERROR: t_forward_nonack: no branches for forwarding May 30 09:59:05 kamailio /usr/local/sbin/kamailio[21602]: ERROR: tm [tm.c:1466]: _w_t_relay_to(): ERROR: w_t_relay_to: t_relay_to failed Any idea how to resolve that?

2 2

converting integer to string
by Henry Fernandes 30 May '16

30 May '16

In Kamailio 4.2, I’m having problems converting an AVP integer value into a string. My Kamailio config stores the return code from auth_check() and outputs it. $avp(auth_code) = auth_check("$fd", "subscriber", "1"); xlog("L_NOTICE", “response = $avp(auth_code)”); When I check the AVP against an integer, it behaves normally. However, I cannot convert it to a string. The return value -3 (invalid user) gets converted to the string “18446744073709551613” when outputting it. My hunch is that Kamailio is treating the AVP as an unsigned integer (rather than a signed integer) when it converts it to a string. If I try this with an AVP with a positive integer, it works as expected. Can someone tell me how to convert a negative integer into a string in Kamailio? -H

3 3

Variable in dialog event routes
by Minh Phan 30 May '16

30 May '16

Hi List, I'm trying to have an external application to write CDR to a database by doing this: event_route[dialog:start] { evapi_relay("{'cdr_id':* $dlg_var(cdr_id),* 'event': 'answered'}"); } event_route[dialog:end] { evapi_relay("{'cdr_id': *$dlg_var(cdr_id)*, 'event': 'ended'}"); } event_route[dialog:failed] { evapi_relay("{'cdr_id': *$dlg_var(cdr_id)*, 'event': 'failed'}"); } $dlg_var(cdr_id) is set inside a request route. However, in these event routes, its value is always null. Its seems that $dlg_var variables are not available in those event routes. What are the variables available in dialog event route? Any suggestion on how to achieve what I'm trying to do? Best regards, Minh

2 3

http_async_client segfault
by Jason Penton 30 May '16

30 May '16

Hi, Has anyone experienced segfault using http_async_query from http_async_client module? Here is BT: #0 __GI_fgets_unlocked (buf=0x7fca23ffe660 "127.0.0.1", n=n@entry=1032, fp=0x7fca23ffe660) at iofgets_u.c:53 53 iofgets_u.c: No such file or directory. (gdb) bt #0 __GI_fgets_unlocked (buf=0x7fca23ffe660 "127.0.0.1", n=n@entry=1032, fp=0x7fca23ffe660) at iofgets_u.c:53 #1 0x00007fca68f08afa in get_contents (stream=<optimized out>, len=1032, linebuf=0x7fca23ffe660 "127.0.0.1") at nss_files/files-XXX.c:201 #2 internal_getent (result=0x7fca23ffe5c0, buffer=buffer@entry=0x7fca23ffe640 "\177", buflen=buflen@entry=1064, errnop=errnop@entry=0x7fca23ffebb0, herrnop=herrnop@entry=0x7fca23ffec00, af=af@entry=0, flags=flags@entry=0) at nss_files/files-XXX.c:246 #3 0x00007fca68f09ca3 in _nss_files_gethostbyname4_r (name=name@entry=0x7fca2e790fa0 "localhost", pat=pat@entry=0x7fca23ffebe0, buffer=buffer@entry=0x7fca23ffe640 "\177", buflen=buflen@entry=1064, errnop=errnop@entry=0x7fca23ffebb0, herrnop=herrnop@entry=0x7fca23ffec00, ttlp=ttlp@entry=0x0) at nss_files/files-hosts.c:402 #4 0x00007fca79eb7757 in gaih_inet (name=<optimized out>, name@entry=0x7fca2e790fa0 "localhost", service=<optimized out>, req=req@entry=0x7fca341c7490, pai=pai@entry=0x7fca23ffed10, naddrs=naddrs@entry=0x7fca23ffed00) at ../sysdeps/posix/getaddrinfo.c:850 #5 0x00007fca79ebae0d in __GI_getaddrinfo (name=0x7fca2e790fa0 "localhost", service=0x7fca23ffeec0 "18000", hints=0x7fca341c7490, pai=0x7fca23ffee78) at ../sysdeps/posix/getaddrinfo.c:2406 #6 0x00007fca6e0f80f4 in ?? () from /usr/lib/x86_64-linux-gnu/libcurl.so.4 #7 0x00007fca6e104fa4 in ?? () from /usr/lib/x86_64-linux-gnu/libcurl.so.4 #8 0x00007fca6e10284b in ?? () from /usr/lib/x86_64-linux-gnu/libcurl.so.4 #9 0x00007fca723c9182 in start_thread (arg=0x7fca23fff700) at pthread_create.c:312 #10 0x00007fca79ee547d in clone () at ../sysdeps/unix/sysv/linux/x86_64/clone.S:111 Camille, are you maintaining this module now?

2 2

Multiple async operations in single transaction fail (using http_async_client)
by Thom Seddon 30 May '16

30 May '16

Hi All, I've been doing some work this week with the new http_async_client module, it seems that with it's default behaviour it's not possible to execute another async query in the block immediately following the response, for example the following config: request_route { if (!t_newtran()) { sl_reply_error(); exit(); } http_async_query("http://localhost:8090/req_1", '{"user":"$fU"}', "REQ_1"); } route[REQ_1] { xlog("L_INFO", "route[REQ_1]: INIT\n"); if ($http_ok) { xlog("L_INFO", "route[REQ_1]: status $http_rs\n"); xlog("L_INFO", "route[REQ_1]: Sending REQ_2\n"); http_async_query("http://localhost:8090/req_2", '{"user":"$fU"}', "REQ_2"); } else { xlog("L_INFO", "route[REQ_1]: error $http_err)\n"); } } route[REQ_2] { xlog("L_INFO", "route[REQ_2]: INIT\n"); if ($http_ok) { xlog("L_INFO", "route[REQ_2]: status $http_rs\n"); } else { xlog("L_INFO", "route[REQ_2]: error $http_err)\n"); } } Generates the following error: 10(11832) INFO: <script>: route[REQ_1]: INIT 10(11832) INFO: <script>: route[REQ_1]: status 200 10(11832) INFO: <script>: route[REQ_1]: Sending REQ_2 10(11832) WARNING: tm [t_suspend.c:186]: t_continue(): transaction is not suspended [53195:560306836] Looking at the source, http_async_client calls t_suspend, then goes away and performs the request in the background. On response it calls t_continue and then this block <https://github.com/kamailio/kamailio/blob/4.4/modules/tm/t_suspend.c#L270-L…> in t_continue will continue execution, the second query will cause http_async_client to call t_suspend whilst sill executing from the previous t_continue. Once they query is sent, the first t_continue will complete, removing the T_ASYNC_SUSPENDED flag. When the second query returns it doesn't find the flag and so fails as shown. I'm far from an expect on this so would really appreciate some pointers, it seems to fundamental to be a race condition in tm but the usage in http_async_client seems to match usage in other modules so I'm not sure on the best tactic. I've tried peppering append_branch and async_route call's in various places to no avail. Full debug log here: http://pastebin.com/raw/rFmm0fmf Any help appreciated, *Thom Seddon*

2 1

username of Contact header
by Al S 27 May '16

27 May '16

Hi, I am trying to read username portion of Contact header: My Contact header content has a URI value such as: sip:8131110000@10.10.10.10.:5060 and I am trying to read : 8131110000 I tried the following perl similar RE and it didn't work: ($var(main_number2)) = $ct =~ /sip:(.*)@/; Thanks,AS

2 1

4.4: Kamailio crashes at start - sometimes
by Sebastian Damm 27 May '16

27 May '16

Hi, we have a machine running 16 Kamailio instances, and while upgrading to 4.4.1 (from 4.3.5), 8 of them wouldn't start. When downgrading to 4.3.5, they all start again. All of them have pretty identical configuration files, except IPs, ports and some code enabled or disabled via defines. After comparing a working and non-working configuration and adjusting setting by setting, we finally ended up with a working configuration. The difference is, that it won't start when a part of the code DOES NOT get included. If it gets included, it will start. This is the mentioned part of our main route: #!ifdef ENABLE_INV_RATELIMIT # Check for INVITE limit if (is_method("INVITE") && $au == $null && !($ua =~ "sipgate") ) { $var(invcount) = $shtcn(invcount=>%~$fU); xlog("L_INFO", "INVITE Requests from $fU in last 30 seconds: $var(invcount)\n"); if ($var(invcount) < 12) { $var(uniqcid) = $ci + $Ts + $ft; $var(tkey) = $fU + '-' + $(var(uniqcid){s.md5}{s.substr,0,10}); $sht(invcount=>$(var(tkey))) = 1; $var(uniqcid) = $null; $var(tkey) = $null; } if ($var(invcount) > 10) { if ($var(invcount) == 11 ) { xlog("L_NOTICE", "User $fU ($var(domain2use)) over ratelimit for new calls, rejecting.\n"); } # Enable this only after evaluating the impact! append_to_reply("Retry-After: 30\r\n"); sl_send_reply("503", "Call Rate Limit Exceeded"); exit; } } #!endif If we put this line at the top of the configuration file, everything works: #!define ENABLE_INV_RATELIMIT If we delete this line, startup does not work. It just sits in ps for one minute without forking, and then gets terminated. We enabled a bit of debugging, and this is apparently the error causing Kamailio to shutdown: May 25 14:50:15 kammel /usr/sbin/kamailio[24989]: DEBUG: <core> [sr_module.c:920]: init_mod_child(): rank 53: nathelper May 25 14:50:15 kammel /usr/sbin/kamailio[24987]: DEBUG: <core> [local_timer.c:61]: init_local_timer(): timer_list between 0x9f0428 and 0xa34428 May 25 14:50:15 kammel /usr/sbin/kamailio[24987]: DEBUG: <core> [io_wait.h:376]: io_watch_add(): DBG: io_watch_add(0x9f0240, 82, 1, (nil)), fd_no=0 May 25 14:50:15 kammel /usr/sbin/kamailio[24987]: ERROR: <core> [io_wait.h:459]: io_watch_add(): epoll_ctl failed: Bad file descriptor [9] May 25 14:50:15 kammel /usr/sbin/kamailio[24987]: CRITICAL: <core> [tcp_read.c:1747]: tcp_receive_loop(): failed to add tcp main socket to the fd list May 25 14:50:15 kammel /usr/sbin/kamailio[24987]: CRITICAL: <core> [tcp_read.c:1815]: tcp_receive_loop(): exiting... I have no idea, how this part of the code could lead to this error, but it is reproducable, that at least on this system setting or disabling this define fixes or breaks the startup. Does anybody have an idea, what's happening there? Best Regards, Sebastian

2 4

Multihomed kamailio - one IP for each account
by Attila Megyeri 27 May '16

27 May '16

Hi Folks, I am looking for a solution to do the following, basically an SBC-like solution. I have an internal asterisk system that should make outbound calls towards a provider, but that provider authenticates the customer based on IP address. And I have many such customers. The provider is not willing to make changes to his practice... Asterisk is bound to a single IP, so there is not much I can do. I was thinking about creating several outbound accounts for asterisk, and then a multi-homed Kamailio would do the following: If request comes from user1, Kamailio would send the request to the provider from IP1. If request comes from user2, Kamailio would send the request to the provider from IP2. And so on, for as many IPs as many customers I want to handle. Is there a more elegant way to do this? If not, how should I implement this? Thanks in advance, Attila

2 4

Kamailio Via: header issue
by Laura 27 May '16

27 May '16

Good morning list, I need to ask you a question about a problem we are fighting on our VoIP platform based on Kamailio 4.3.5. Our platform is made by more nodes over Europe countries, and we are encountering a problem with the size of the SIP package which, exceeds the physiological limit of about 1350 bytes. The real problem is caused from Via: Header (inside the SIP packet) which is added to each element of the system for transit on the call. >From what we have analyzed, the problem is not so much due to the number of entered Via: records but, by the fact that they contain a branch parameter extremely long. Here the call flow example from CARRIER A —> Kamailio1 —> Kamailio2 —> CARRIER B Extract from SIP message sent from CARRIER A to K1 INVITE sip:999912341234@192.168.158.42 SIP/2.0. Via: SIP/2.0/UDP 192.168.220.141:5060;branch=z9hG4bK49e4301c;rport. Extract from SIP message sent from K1 to K2 INVITE sip:999912341234@192.168.127.244 SIP/2.0. Record-Route: <sip:192.168.158.42;lr;did=194.a0f1;nat=yes>. Via: SIP/2.0/UDP 192.168.158.42;branch=z9hG4bK0635.7c1e97c2d5c01ea98e8d0e7fa23a3822.0. Via: SIP/2.0/UDP 192.168.158.42;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.cuOBV4OBVLMGZfWxF-W.y6Mx1LgRWIC9gIgx4fzxj7MBP7MBVAOBF4M.1jmxuqC93X3heroEWvH9vsCFN43qd4zRj4Mlyf3l1LNSQLpx4uMx3A. Extract from SIP message sent from K2 to CARRIER B INVITE sip:999912341234@192.168.65.248 SIP/2.0. Record-Route: <sip:192.168.127.244;lr;did=194.5122;nat=yes>. Via: SIP/2.0/UDP 192.168.127.244;branch=z9hG4bK0635.f843472880b380acde3a33732975fde9.0. Via: SIP/2.0/UDP 192.168.127.244;branch=z9hG4bKsr-j4IPOlV7MGQKatycW.F7MBj4OBFuzGZ4MB1LNSQLpx4uMx3AzuaVHRaYpB1JNEt736cQkBIvalaJmly6Mlj7W6Mfg.qw3leqWRMAMRKrz.rIzSPAg.pE3.Vl3.MZMBV7My**. Via: SIP/2.0/UDP 192.168.127.244;branch=z9hG4bKsr-j4IPOlV7MGQKatycW.F7MBj4OBFuzGZ4MB1JNEt736cQkBIvalaJmwWLORv4mK0Hot3w.jpam6t4kRWWOEWu.4eRWFQGKqfWauYEKwrSOKN7k.gWkxtMgue9mjMsg4IhkxaEkhrbW4uGjxpWPIg0.4eSWF47MRY1kUKfPlqlRxWvg9e5aKp6mxI6N4WS.BPlNRP4kIebWFudkR3loxtM.IWa.U0ZWUKWkxW0. Is there anyone here with the idea how to solve this problem ? Best regards

3 8

rtpproxy module question
by gmele 27 May '16

27 May '16

Hello, we are using an RTP Proxy from rtpproxy.org as media relay to establish communication between our mobile phones. Of course, we are using the kamailio rtpproxy module to modify the SDP payload and control the proxy. In our Kamailio configuration, we have 1 kamailio configured as Proxy and one kamailio configured as Registrar. So calls go through the Proxy and then to the Registrar who will update the SDP header and select an available rtp proxy. We have noticed that sometimes, the rtp udp flow between the phones isn't routed properly by the rtpproxies, ending in the communication drop (all the SIP nego is working well, and the SDP are correctly patched with the rtp proxy address and port). Analyzing the RTP proxy packets, we have found that the Kamailio registrar gives the Kamailio proxy ip address in the RTP proxy create session command, but keeps the original sdp port. command looks like this: Uc96,101 DC -PbO~Wnm <proxy_ip> <port from original sdp phone packet> PZU5OITCW;1 We are using rtpproxy_manage() without any flags. It seems to us that this ip and port are used as default forward route as long as the callee hasnt connected to the rtpproxy. *Is it correct? * If its true, Its seems to us that this cant work as we are mixing the proxy sip address with a udp port open on the phone ? *Is our analysis correct ?* Can we use some option in rtpproxy_manage to replace the proxy ip by the phone ip as seen in the via route ? Thx for your help Giovanni -- View this message in context: http://sip-router.1086192.n5.nabble.com/rtpproxy-module-question-tp148850.h… Sent from the Users mailing list archive at Nabble.com.

2 3

Kamailio not sending reply
by Jeremy Betts 26 May '16

26 May '16

Hello, I'm having an issue where kamailio is not sending the failure reply as I would expect when receiving calls from a certain provider. Kamailio sends the 403 response on calls from other providers. The big difference I see is that the "problem" provider is using compact headers. I am trying to force the 403 response to be sent for testing purposes but I just can't get it to reply to this one provider. I've included traces and configuration excerpts below. Any help is much appreciated! Problem Call (No Reply Sent): U 63.79.178.192:5060 -> 184.171.164.100:5060 INVITE sip:+19727289377@184.171.164.100;transport=UDP;user=phone SIP/2.0. v: SIP/2.0/UDP 63.79.178.192:5060;branch=z9hG4bK9a2b97bc805e6a3b73b43e3de4150da5.1d819013. Record-Route: <sip:63.79.178.192;lr>. f: <sip:+17143258018@199.173.94.144:5060;user=phone>;tag=-45026-41c7bce-729616d4-41c7bce. t: <sip:+19727289377@63.79.178.192:5060;user=phone>. i: b03a96e8905eadc713c441c7bcef439f1124b7ca791c2679c0-0086-5719. CSeq: 1 INVITE. Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK. v: SIP/2.0/UDP SCR9:5060;maddr=199.173.94.144;branch=z9hG4bK-41c7bce-f439f11-21b78f96;received=199.173.94.144. Max-Forwards: 27. m: <sip:199.173.94.144:5060;transport=UDP>. k: 100rel, resource-priority, replaces. c: application/sdp. l: 235. P-Asserted-Identity: <sip:+17143258018@63.79.178.192;user=phone>. Privacy: none. . v=0. o=PVG 1464277233580 1464277233580 IN IP4 199.173.68.106. s=-. p=+1 6135555555. c=IN IP4 199.173.68.106. t=0 0. m=audio 55380 RTP/AVP 18 0 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=fmtp:18 annexb=no. # U 63.79.178.192:5060 -> 184.171.164.100:5060 ACK sip:+19727289377@184.171.164.100;transport=UDP;user=phone SIP/2.0. v: SIP/2.0/UDP 63.79.178.192:5060;branch=z9hG4bK9a2b97bc805e6a3b73b43e3de4150da5.1d819013. f: <sip:+17143258018@199.173.94.144:5060;user=phone>;tag=-45026-41c7bce-729616d4-41c7bce. t: <sip:+19727289377@63.79.178.192:5060;user=phone>;tag=b1eb89aa72a4b2a406f6fb21bbd3e03f.1aac. i: b03a96e8905eadc713c441c7bcef439f1124b7ca791c2679c0-0086-5719. CSeq: 1 ACK. l: 0. Max-Forwards: 27. . Successful Call (Reply Sent): # U 64.136.173.31:5060 -> 184.171.164.100:5066 INVITE sip:7146466334@184.171.164.100:5066 SIP/2.0. Via: SIP/2.0/UDP 64.136.173.31:5060;branch=z9hG4bK1sansay3408006464rdb14345. Record-Route: <sip:sansay3408006464rdb14345@64.136.173.31:5060;lr;transport=udp>. To: <sip:7146466334@184.171.164.100>. From: <sip:17143258018@64.136.173.31>;tag=sansay3408006464rdb14345. Call-ID: 1089333243-0-3123809354(a)64.136.173.226. CSeq: 1 INVITE. Contact: <sip:17143258018@64.136.173.31:5060>. Supported: timer. Session-Expires: 1800;refresher=uac. Min-SE: 90. P-Asserted-Identity: <sip:17143258018@192.168.20.76>. Privacy: none. Expires: 120. Max-Forwards: 67. Content-Type: application/sdp. Content-Length: 274. . v=0. o=Sansay-VSXi 188 1 IN IP4 64.136.173.31. s=Session Controller. c=IN IP4 69.85.185.142. t=0 0. m=audio 37970 RTP/AVP 0 18 101. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=ptime:20. # U 184.171.164.100:5066 -> 64.136.173.31:5060 SIP/2.0 403 DID Lookup Failed. Via: SIP/2.0/UDP 64.136.173.31:5060;branch=z9hG4bK1sansay3408006464rdb14345. To: <sip:7146466334@184.171.164.100>;tag=b1eb89aa72a4b2a406f6fb21bbd3e03f.d2ec. From: <sip:17143258018@64.136.173.31>;tag=sansay3408006464rdb14345. Call-ID: 1089333243-0-3123809354(a)64.136.173.226. CSeq: 1 INVITE. Server: x-Freevoice SIP Proxy 4.21. Content-Length: 0. . # U 64.136.173.31:5060 -> 184.171.164.100:5066 ACK sip:7146466334@184.171.164.100:5066 SIP/2.0. Via: SIP/2.0/UDP 64.136.173.31:5060;branch=z9hG4bK1sansay3408006464rdb14345. To: <sip:7146466334@184.171.164.100>;tag=b1eb89aa72a4b2a406f6fb21bbd3e03f.d2ec. From: <sip:17143258018@64.136.173.31>;tag=sansay3408006464rdb14345. Call-ID: 1089333243-0-3123809354(a)64.136.173.226. CSeq: 1 ACK. Max-Forwards: 70. Content-Length: 0. Kamailio Config: request_route { # per request initial checks route(REQINIT); # NAT detection #route(NATDETECT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { setflag(FLT_ACCFAILED); if (t_check_trans()) t_relay(); exit; } t_check_trans(); if(is_method("INVITE")||is_method("MESSAGE")){ route(ORIGINATE); #route(FROMAST); } route[ORIGINATE] { sl_send_reply("403", "DID Lookup Failed"); exit; } Jeremy Betts <https://www.youtube.com/user/FreevoiceLLC> <https://www.youtube.com/user/FreevoiceLLC>

2 2

webrtc tlsv1 alert access denied
by Arsen 26 May '16

26 May '16

Hi guys! I am trying to configure kamailio with WSS. We have trusted certificate installed SIP over TCP/TLS works fine. But when I try WSS I got error: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS read:error:14094419:SSL routines:SSL3_READ_BYTES:tlsv1 alert access denied ERROR: <core> [tcp_read.c:1303]: tcp_read_req(): ERROR: tcp_read_req: error reading Before above error it was 'bad certificate', so I have imported CA in the firefox and now I get these errors.. I have tried sipml5 and tryit.jssip.net same issue with both clients, also it seems I have these errors only when I use firefox, when I use chrome it even doesn't show me an error.. Any ideas? Thanks! -- Regards, Arsen.

2 3

SRV and dialog stickiness
by Alex Balashov 26 May '16

26 May '16

Hello, Sorry if this is a tired, worn question, but I've not dealt much with Kamailio's SRV support before: If a registrant has a contact binding whose domain component is subject to an SRV lookup with load-balanced or weighted entries, how does one solve the problem of ensuring that subsequent in-dialog requests go to the same host as the initial INVITE? Does Kamailio offer some facility for doing this? Is it somehow accommodated by SIP? -- Alex -- Alex Balashov | Principal | Evariste Systems LLC 1447 Peachtree Street NE, Suite 700 Atlanta, GA 30309 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

3 5

rtpengine and SAVPF
by Ján Füri 26 May '16

26 May '16

Hi all, Im struggling with incoming calls to webrtc clients. I am trying to change INVITE's SDP using rtpengine from RTP/AVP or RTP/SAVP to RTP/SAVPF towards my webrtc clients. This works well. But if I send INVITE with SDP that includes RTP/AVP and RTP/SAVP as well, rtpengine creates incorrect SDP. It includes twice m=audio RTP/SAVPF and my WebRTC client does not like this INVITE and forbiddeds it by 488 Not Acceptable Here. Example of my incorrect INVITE : Content-Type: application/sdp Content-Disposition: session Content-Length: 1393 v=0 o=SBC 1464238673 1464238674 IN IP4 <MyPublicIP> s=SBC c=IN IP4 <MyPublicIP> t=0 0 m=audio 30454 RTP/SAVPF 102 9 8 0 3 18 a=rtpmap:102 SILK/8000 a=fmtp:102 useinbandfec=1; usedtx=0 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=ptime:20 a=sendrecv a=rtcp:30455 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:5X2wgU83bIwrwkHSymjnw48SiMGa1n+eIl50yr99 a=setup:actpass a=fingerprint:sha-1 11:76:2D:2A:F7:0D:5A:23:9D:F6:0C:E7:4C:DF:1E:CB:BF:5D:76:88 a=ice-ufrag:iglHZqry a=ice-pwd:eyVXbFNoviYYlu5uuRZlnixFwQ a=candidate:cE0FGbXWIfwj6OGD 1 UDP 2130706431 <MyPublicIP> 30454 typ host a=candidate:cE0FGbXWIfwj6OGD 2 UDP 2130706430 <MyPublicIP> 30455 typ host m=audio 30484 RTP/SAVPF 102 9 8 0 3 18 a=rtpmap:102 SILK/8000 a=fmtp:102 useinbandfec=1; usedtx=0 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=ptime:20 a=sendrecv a=rtcp:30485 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:ozDJ3G/wmaYedQbcTafbhaTIt6raIJa6ugrLUC99 a=setup:actpass a=fingerprint:sha-1 11:76:2D:2A:F7:0D:5A:23:9D:F6:0C:E7:4C:DF:1E:CB:BF:5D:76:88 a=ice-ufrag:gM1viWqu a=ice-pwd:zdS4tP1Mj4hLTe6huBWYrhtI1s a=candidate:cE0FGbXWIfwj6OGD 1 UDP 2130706431 <MyPublicIP> 30484 typ host a=candidate:cE0FGbXWIfwj6OGD 2 UDP 2130706430 <MyPublicIP> 30485 typ host My setup : incoming call -> MediaServer (freeswitch) -> SIP & Websocket proxy (Kamailio + Rtpengine) -> webRTC clients Is there anybody who could help me with this issue or explain me how to remove the second line of m=audio RTP/SAVPF ? -- Jan

1 0

Sequential calls messes up presence with pua-dialog info
by Ali Pey 25 May '16

25 May '16

Hello, We are using pua-dialoginfo and presence modules for BLF. When the first call comes in or goes out, every thing works great; however, a second call to the same key/extension (e.g a call waiting which is not answered) would turn off the key even though the original call/dialog is still active. Has anyone had experienced such problem? How can I fix this? Any constructive suggestion would be appreciated. Regards, Ali Pey

4 5

nh_reload_rtpp not available
by david 25 May '16

25 May '16

dear all i'm using kamailio 4.4.1 and i'm not seeing nh_reload_rtpp as a fifo exported function i'm trying to setup the dburl for the rtpengine sets using db_text but when i do kamctl fifo which i see all the commands of the module except the reload_rtpp nh_enable_rtpp nh_show_rtpp nh_ping_rtpp nh_show_hash_total /usr/local/kamailio/sbin/kamctl fifo nh_reload_rtpp 500 command 'nh_reload_rtpp' not available these are the config lines i have for that #!define DBURLRTP "text:///var/log/rtpengine_conf" loadmodule "db_text.so" modparam("rtpengine", "db_url", DBURLRTP) modparam("rtpengine", "table_name", "rtpproxy") and the database is like root@test:/var/log/rtpengine_conf# ls rtpproxy version root@test:/var/log/rtpengine_conf# cat version table_name(str) table_version(int) rtpproxy:1 root@test:/var/log/rtpengine_conf# cat rtpproxy set_id(int) url(str) weight(int) disable(int) 1:udp\:x.x.x.x\:x:100:0 2:udp\:x.x.x.x\:x:100:0 is there something i missed here? do i must use mysql or postgres drivers for this? (i use a test machine to old to install a driver of those with kam 4.4 maybe?) best regards david

1 1

doubt using rate limit
by david escartin 25 May '16

25 May '16

Hello Daniel i had the call rate for some minutes actually. maybe 4-5min and all the time the behaviour was like that. it's true that in that kamailio there wa no any other traffic, only those test calls i set with a sipp to simulate the load best regards david Hello, for how long you got this 20req/second? The module is not counting like continuous time, but more like traffic on each second, iirc. Cheers, Daniel On 23/05/16 11:24, david escartin wrote: > Hello all > > i have a quick question about using module ratelimit on kamailio 4.2 > or 4.4. > on tests made before running it on production, i set the RED > algorithm with a limit of 10 calls per second, but i saw the call > limitation was not done until we got more than 20 calls per second or > so on the queue. we didnt get any INVITE discarded. > after surpassing around 20 INV/sec, it's true the algorithm starts to > work fine and only 10INV/sec are passed more or less > > on production i have no easy way to check this behaviour, besides it's > a feature barely used. > > so i only want to know if this behaviour is known, or if i'm missing > soemthing > > my config is like this > > modparam("ratelimit", "timer_interval", 5) > modparam("ratelimit", "queue", "4:INVITE") > modparam("ratelimit", "pipe", "4:RED:10") > > route[RATE_LIMIT] { > $var(limitation)="4"; > if (!rl_check("$var(limitation)")) { > send_reply("505","Limiting"); > xlog("L_INFO","Call $ci / Call-ID $ci: ratelimit was reached, > rejecting with 505\n"); > route(CLEAR); > exit; > } > } > > > thanks a lot and regards > david escartin > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users at lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://www.asipto.com - http://www.kamailio.org http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Kamailio for sip routing architecture
by Loic Chabert 25 May '16

25 May '16

Hello Kamailio users, I'm working on a sip infrastructure, with opensource technologies, and of course Kamailio will run on my new infrastructure :) Kamailio will be my central routing engine, between SBC (running on asterisk), and somes boxes (asterisk too) where customers will be registred. Please find below my future infrastructure: ---------- -------------- -------- ---------------- | SBC1 | - - - - - | - - - -| kamailio1 |- - - - - | - - - - - | CB1 |-----------------| SIPPHONE | ---------- | -------------- | -------- ---------------- | | ---------- | | -------- | SBC2 | - - - - - | | - - - - - | CB2 | ---------- | | -------- | -------------- | ---------- | - - - -| kamailio2 |- - - - -| -------- | SBCn |- - - - - | -------------- | - - - - - | CBn | ---------- | | -------- Kamailio high availabality will be done with loopback address and BGP in two datacenters. SIP carriers will be connected to SBC and customer to one CB. I have made some tests with kamailio, and i use LCR to route call from CB to SBC, using wheigt and prorities to send call to carriers. Then, when a call arrived from carriers, i use DNS NATPER lookup to find the right CB box. CB HA is done using DNS SRV. So user is registred at one location, and only one. No location sharing beetwen boxes. I have some question concerning my architecture, and technologies used to routing call. Is LCR is the best option for routing call between multiple carriers ? DNS NAPTER is a modern solution to find an endpoint on this architecture ? Existing a better way to route call with kamailio, for exemple rt-json and curl ? In my point of view, kamailio should be a routing engine, using internal technologies, to maximise performances. Less work with kamailio and external app, best performances in sip routing. Is it the good vision ? Thanks in advance.

2 4

Kamailio routing to different carriers based on prefix
by José Seabra 25 May '16

25 May '16

Hi there, I'm looking for a solution to route calls based on prefix, my routing table might reach a long list of prefixs. I looked to carrierroute and it seems to be nice for this purpose but i have some doubts to how to handle correctly with carrierroute. IfI understood correctly, if I set the parameter avoid_failed_destinations to 1, can i use the same carrier id and domain on failure route to select another destination that matches the prefix? Can I have the same carrier ID with diferent domains sharing the same prefixs but mapping it to different destinations? Thank you for your help. Best regards -- José Seabra

2 1

call count / and work with despatcher
by jaflong jaflong 24 May '16

24 May '16

Hi Everyone, What is a accurate way to get current call count. I am only relaying calls. Initialy I tried the code below. It seemed to work, it reported the correct number of calls I had dialed. I get the result through kamcmd htable get command. ------------------------------------------ under route[RELAY] # # COUNT # if (is_method("BYE")) { $var(n) = $shtdec(callcount=>active); sht_rm_name_re("callcount=>$ft"); } -------------------------------------------- # Manage incoming replies onreply_route[MANAGE_REPLY] { # # COUNT # if (status=="200") { if (is_method("INVITE")) { # On start if($sht(callcount=>active) == $null) { $sht(callcount=>active) = 0; } # New call if ($sht(callcount=>$ft::from_tag)== $null) { $sht(callcount=>$ft::from_tag) = $ft; $var(n) = $shtinc(callcount=>active); } } xlog("L_INFO", "This is a 200 OK reply\n"); } xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") { route(NATMANAGE); } } ------------------------------------------------------- However I wanted to distribute calls to several servers I further added the despatcher code below # round robin dispatching on gateways group '1' if(!ds_select_dst("1", "4")) { send_reply("404", "No destination"); exit; } xlog("L_INFO", "-----RELAYING going to <$ru> via <$du> time [$Tf] -----\n"); This makes my counting code above give incorrect call count. How can I make call count work with dispatcher ? Do advise how I can do this Regards

2 1

Create "Contac Groups" on the server Kamailio
by ALFONSO MARTINEZ GOMEZ 24 May '16

24 May '16

Hi, I'm working on a VoIP communication application with Kamailio as SIP server and using the SIREMIS interface. As an administrator, I can add "Subscribers" which later will be registered on the server so that they can make the necessary negotiations when they want to make a call. My question is, how you can make groups of "contacts" so that such calls can be made only with each other "contacts" in the same group? that is, we can have many "subscribers" on the server and belong to two different groups and that they can only talk to each other contact each group. Thank you so much and greetings!

1 0

doubt using rate limit
by david escartin 24 May '16

24 May '16

Hello all i have a quick question about using module ratelimit on kamailio 4.2 or 4.4. on tests made before running it on production, i set the RED algorithm with a limit of 10 calls per second, but i saw the call limitation was not done until we got more than 20 calls per second or so on the queue. we didnt get any INVITE discarded. after surpassing around 20 INV/sec, it's true the algorithm starts to work fine and only 10INV/sec are passed more or less on production i have no easy way to check this behaviour, besides it's a feature barely used. so i only want to know if this behaviour is known, or if i'm missing soemthing my config is like this modparam("ratelimit", "timer_interval", 5) modparam("ratelimit", "queue", "4:INVITE") modparam("ratelimit", "pipe", "4:RED:10") route[RATE_LIMIT] { $var(limitation)="4"; if (!rl_check("$var(limitation)")) { send_reply("505","Limiting"); xlog("L_INFO","Call $ci / Call-ID $ci: ratelimit was reached, rejecting with 505\n"); route(CLEAR); exit; } } thanks a lot and regards david escartin

2 1

Please help me setting up Kamailio for android application
by Kunal Dhanda 24 May '16

24 May '16

I wanted to create my own voip application using kamailio . I am not able to find any rest integration with kamailio for my server to use for user registration and other task.Please help in developing my appplication. -- Thanks & Regards, Kunal Dhanda, Software Developer E-mail : < "kunaliitkgp09(a)gmail.com" >, Ph. No. (M) : +91 - 8884818805

2 1

Found bug at ACC
by Pavel Eremin 24 May '16

24 May '16

Hi all, when we set syslog facility for ACC to LOCAL3, log_write_cdr() will write at LOG_DAEMON anyway. You have to recompile module acc to user another default syslog facility to save cdr with another syslog facility.

2 1

kamctl stats shmem error changing shared memory
by Mr.Gab 23 May '16

23 May '16

Hi, i have a problem with my kamailio installation. Actually i set the shared memory to 128MB and it's all fine. When i increase the shared memory to 256Mb and more i get the following error while checking kamctl commands (like kamctl stats shmem) ERROR: Error opening Kamailio's FIFO /var/run/kamailio/kamailio_fifo ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name", "/var/run/kamailio/kamailio_fifo")' in your config ERROR: and also have loaded the mi_fifo module. If i revert to 128Mb kamailio is able to create the kamailio_fifo file and all works fine. Can somebody help me to solve? Thanks

3 3

Re: [SR-Users] Change CANCEL reason when fr_timer expires
by Bruno Emer 22 May '16

22 May '16

Hello, Alex! Fisrt of all, thanks for you help with this problem! I don't think that this will help me. I can see that this parameter is enabled by default and in fact it is working in some cases. One case where this works is when the INVITE is sended for more than one endpoint (when the same user is registered in many devices). When someone answers the INVITE, Kamailio sends a CANCEL message to the other endpoints and it has the Reason header showing that the call was answered elsewhere. I want to add the CANCEL Reason header when the timer expires, so kamilio will not receive a final replay to that transaction. *Bruno Emer* Mobile: +55 11 96540-0044 email: brunoemer(a)gmail.com <https://www.facebook.com/bruno.emer.5> <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> <https://twitter.com/brunoemer_> <https://www.flickr.com/photos/122070309@N03/> <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_> 2016-05-20 15:42 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: > Hello all! > > I have a situation here where when I send a INVITE to some specific > user-agents in my environment I must change the fr_timer to a value lower > than the default. > > I can acomplish this using the "t_set_fr" commands in my cfg file, and it > works like a charm. > > The point is that when the timer reaches its limit, the kamailio server > sends a CANCEL message to the user-agent. I know that this is the normal > behavior, but I want to add the Reason field on it, so the user-agent > application can know that the call wasn't answered on time. > > I was looking the TM module documentation to understand how to do this, > but I couldn't figure out a way to get this working. > > Is there a way to add the Reason in CANCEL messages generated by the > fr_timer? > > > Thanks. > > > *Bruno Emer* > > Mobile: +55 11 96540-0044 > email: brunoemer(a)gmail.com > <https://www.facebook.com/bruno.emer.5> > <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> > <https://twitter.com/brunoemer_> > <https://www.flickr.com/photos/122070309@N03/> > <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_> >

2 1

Change CANCEL reason when fr_timer expires
by Bruno Emer 21 May '16

21 May '16

Hello all! I have a situation here where when I send a INVITE to some specific user-agents in my environment I must change the fr_timer to a value lower than the default. I can acomplish this using the "t_set_fr" commands in my cfg file, and it works like a charm. The point is that when the timer reaches its limit, the kamailio server sends a CANCEL message to the user-agent. I know that this is the normal behavior, but I want to add the Reason field on it, so the user-agent application can know that the call wasn't answered on time. I was looking the TM module documentation to understand how to do this, but I couldn't figure out a way to get this working. Is there a way to add the Reason in CANCEL messages generated by the fr_timer? Thanks. *Bruno Emer* Mobile: +55 11 96540-0044 email: brunoemer(a)gmail.com <https://www.facebook.com/bruno.emer.5> <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> <https://twitter.com/brunoemer_> <https://www.flickr.com/photos/122070309@N03/> <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_>

2 1

extra characters in re-invite "TO" field
by Al S 20 May '16

20 May '16

Hi, Kamailio adds extra characters in the re-invite "TO" field during session timer. There is no header manipulation in our route[WITHINDLG] block. Re-INVITE received by KamailioTo: <sip:8131111111@?.?.?.?;otg=1>;tag=1c1048632956 Re-INVITE forwarded by KamailioTo: <sip:8132222222@?.?.?.?;T\033\023Z\f1>;tag=1c1048632956 Your help would be greatly appreciated. Thanks,AS

2 2

Question about testing RTPENGINE
by Dmitry 20 May '16

20 May '16

Hello I am trying to make a test setup to test the RTPENGINE. Now my setup is Kazoo . i.e.: Subsriber A ---Asterisk (192.168.175.132)-----via NAT-------------Kamailio(10.10.11.104) ----FreeSwitch(10.10.11.104) Subscriber B is connected directly to Kamailio (this subscriber is in the same network) First WITHOUT RTPENGINE : When A calls to B - the sound both ways, so it is ok. Second WITH RTPENGINE: When A calls to B - the sound both ways, so it is ok. Actually what I want is to make such setup that calls without the RTPENGINE are impossible.

1 0

WebRTC
by Moacir Ferreira 20 May '16

20 May '16

Hi, I am trying to use Kamailio with WebTRC to make and receive calls from the browser. Using the rpms from the Kamailio repository I have installed and tried the websocket config example from the source code. Using the first debug example from here https://www.kamailio.org/wiki/tutorials/tls/testing-and-debugginga I can see that my TLS wss setup is working. But I try JSCommunicator and sipML5 I get no success, they fail to register in Kamailio. From the logs I can see that something goes wrong with TLS. Doing some research, it is more like that several products are currently broken and not working... So can anyone help me (share a config and tricky) or point to a document that would help me to get it running? Thanks, Moacir

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Re: [SR-Users] Kamailio notify conflict when a second call lands on a phone
by Pejman Rajabian 19 May '16

19 May '16

Is there is reason why this post does not get approved? From: Pejman Rajabian [mailto:pejman@italkip.com] Sent: Wednesday, May 18, 2016 12:59 PM To: 'sr-users(a)lists.sip-router.org' Subject: Kamailio notify conflict when a second call lands on a phone I am having a dilemma with manipulation of the light status when multiple calls land on the same extension. Let's say a call is in progress on extension 107 and Kamailio has already sent notifies to all subscribers who watch the status of extension 107 and they all have a red light for the blf button assigned to 107 on their phones. When a second call is destined for extension 107 while the first call is in progress, the Kamailio server sends a new notify to the subscribers to 107. If one of the two calls hangs up while the other one is still in progress, Kamailio sends a new notify to subscribers to report idle status and the subscribers phone blf light for extension 107 turns green while there is an active call on this extension and it should be red. I guess what Im trying to say is that I need a way to tell Kamilio to reset back to the last status instead of just sending a idle status. Can someone please help with this matter? Pejman Rajabian Director of Product Development iTalk IP Networks LLC Office: (703) 550-8888 x.107 Cell: (571) 723-2224 Fax: (703) 439-1350 pejman(a)italkip.com

2 1

Sipcapture on SPARC
by Spencer Thomason 18 May '16

18 May '16

Hello, I’m attempting to deploy a homer sip capture server on SPARC and I’m seeing some strange date related errors: ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on query: Incorrect datetime value: '2052-06-12 16:21:06' for column 'date' at row 1 (1292) Is there some sore of endianness issue? Thanks for any assistance! Regards, Spencer

2 3

[kamailio 5.0] upgrade VS rollback/backward compatibility
by Alex Lutay 18 May '16

18 May '16

Hi, We are heavily using Kamailio in our product Sipwise NGCP and currently testing possibility to provide rollback for the customers with the huge systems as the last resort functionality in case if upgrade went wrong. We hope never use it on practice but prefer to have such possibility. The main logic here is to change DB in 'backward compatible' way. So OLD code must work with NEW DB schema (but NOT a vice verse). We have some rules which helps us care about it, like "Do not drop columns, just discontinue them" and "All new fields should have defaults", etc. Also we have separated 'code', 'configs' and 'data' parts of our product and rollback 'code'/'configs' only, while 'data' (e.g. MySQL DB) stays in new state. Otherwise rollback will require a lot if time to revert all DB statements and sometimes it is just not possible. While every time we have to upgrade Kamailio we have a problems with rollback possibility. The point here that Kamailio checks DB table_version and doesn't start if the version mismatch found. Q: What about to add Kamailio possibility to work in backward compatible mode? Sample for upgrade Kamailio 4.3.x -> 4.4.x (the versions here as an example only!) https://www.kamailio.org/wiki/install/upgrade/4.3.x-to-4.4.0 We have 3 DB statements there which are adding new columns to 3 tables, like > -- table: active_watchers > ALTER TABLE active_watchers > ADD COLUMN flags INT(11) NOT NULL DEFAULT '0', > ADD COLUMN user_agent VARCHAR(255) NOT NULL; > DELETE FROM version WHERE TABLE_NAME='active_watchers'; > INSERT INTO version (`table_name`, `table_version`) VALUES ('active_watchers','12'); Problem here is missed default for column 'user_agent' which can be empty value as for me. So, lets assume we have it there. Then we can add new column to table 'version', like that: > mysql> select * from version where table_name='active_watchers'; > +-----------------+---------------+---------------------+ > | table_name | table_version | table_compatibility | > +-----------------+---------------+---------------------+ > | active_watchers | 12 | 11 | > +-----------------+---------------+---------------------+ > 1 row in set (0.00 sec) Currently: - module presence version 4.4.x starts check required version and refuse to load if table_version!=12 Proposed behaviour: - module presence version 4.4.x starts check required version and refuse to load if table_version!=12 and compatibility option is not set - if compatibility option is set, check value of table_compatibility and refuse to load if current version less then table_compatibility value >From the sample above I would say module presence in Kamailio 4.4.x is compatible with Kamailio 4.3.x if we define default value for the field 'user_agent'. Having that will allow us to rollback on previous NGCP release and _temporary_ work on Kamailio 4.3.x having new Kamailio DB 4.4.x P.S. This is just a question to check Kamailio community vision here and reality is more complex as described, so some additional checks/limitations will be in place. P.P.S Compatibility Kamailio option can/should be disabled by default. Field 'table_compatibility' can be outside table 'version', e.g. in separate table. Complete compatibility logic can be some separate Kamailio 'compatibility' module. Thank you! -- Alexander Lutay Head of Quality Assurance Sipwise GmbH, Campus 21/Europaring F15 AT-2345 Brunn am Gebirge Office: +43(0)13012036 Email: alutay(a)sipwise.com Website: https://www.sipwise.com Meet us @ ANGACOM: Hall 10.1 / booth N10 Exhibition and Congress for Broadband, Cable & Satellite: 07 – 09 June, 2016 in Cologne

4 3

Kamailio notify conflict when a second call lands on a phone
by Pejman Rajabian 18 May '16

18 May '16

I am having a dilemma with manipulation of the light status when multiple calls land on the same extension. Let's say a call is in progress on extension 107 and Kamailio has already sent notifies to all subscribers who watch the status of extension 107 and they all have a red light for the blf button assigned to 107 on their phones. When a second call is destined for extension 107 while the first call is in progress, the Kamailio server sends a new notify to the subscribers to 107. If one of the two calls hangs up while the other one is still in progress, Kamailio sends a new notify to subscribers to report idle status and the subscribers phone blf light for extension 107 turns green while there is an active call on this extension and it should be red. I guess what Im trying to say is that I need a way to tell Kamilio to reset back to the last status instead of just sending a idle status. Can someone please help with this matter? Pejman Rajabian Director of Product Development iTalk IP Networks LLC Office: (703) 550-8888 x.107 Cell: (571) 723-2224 Fax: (703) 439-1350 <mailto:pejman@italkip.com> pejman(a)italkip.com

1 0

invalid route error when turning on Roles in kamailio kazoo
by Dmitry 18 May '16

18 May '16

hello When I uncomment in /etc/kazoo/kamailio/local.cfg#!trydef TRAFFIC-FILTER-ROLE (it was commented previously) And the kamailio does not start and the log gives this error:-Invalid route satement The structure of files is simple:the trafic-filter-role is in the separate file. and in the main file (default.cfg #!ifdef TRAFFIC_FILTER-ROLEinclude_file "traffic-filter-role.cfg" and that's it. But it gives INVALID ROUTE ERROR

2 2

Re: [SR-Users] Multiple Registrations - Overwrite oldest contact
by Bruno Emer 17 May '16

17 May '16

Hello, David. Sorry for the time to reply you. I was a little busy here in the last few days. I am not sure about the "==". We had tried it here, but it only worked when we used with =~. We are currently running kamailio 4.2, so it may be a problem with our version. About the $hdr variable, it is used to get information from the SIP Headers. I am not sure, but I dont think that there is an header with Instance-ID, so you must try to find how can you get this ID in kamailio, as I don't know where is it saved. *Bruno Emer* Mobile: +55 11 96540-0044 email: brunoemer(a)gmail.com <https://www.facebook.com/bruno.emer.5> <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> <https://twitter.com/brunoemer_> <https://www.flickr.com/photos/122070309@N03/> <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_> 2016-05-13 12:24 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: > Hello all! > > After some time I was able to get this working here, and now I am coming > back to share my configuration with you guys (in case someone needs to get > this working). > > Here is what we have done: > > route[REGISTRAR] { > if (is_method("REGISTER")) > { > if(isflagset(FLT_NATS)) > { > setbflag(FLB_NATB); > # uncomment next line to do SIP NAT pinging > ## setbflag(FLB_NATSIPPING); > } > > xlog("L_INFO", "checking $tu registered with > User-Agent:$hdr(User-Agent)\n"); > if(reg_fetch_contacts("location", "$tu", "caller")) > { > xlog("L_INFO", "$(ulc(caller=>count)) records > found for $tu\n"); > $var(i) = 0; > while($var(i) < $(ulc(caller=>count))) > { > > xlog("L_INFO", "record $var(i) with > ruid:$(ulc(caller=>ruid)[$var(i)]) - > User-Agent:$(ulc(caller=>user_agent)[$var(i)])"); > if > ($hdr(User-Agent)=~$(ulc(caller=>user_agent)[$var(i)])) > { > $var(ruid) = > $(ulc(caller=>ruid)[$var(i)]); > xlog("L_INFO", "unregistering $tu > ruid:$var(ruid)"); > unregister("location", "$tu", > "$var(ruid)"); > } > $var(i) = $var(i) + 1; > } > } > if (!save("location", "0x00")) > sl_reply_error(); > > exit; > } > } > > > There are some logs over there because we used it to see what was going on > while testing the configurations. > > > With this we can get just one register per User-Agent. > > > And again, thanks Olle for your help in this problem > > *Bruno Emer* > > Mobile: +55 11 96540-0044 > email: brunoemer(a)gmail.com > <https://www.facebook.com/bruno.emer.5> > <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> > <https://twitter.com/brunoemer_> > <https://www.flickr.com/photos/122070309@N03/> > <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_> > > 2016-04-01 17:37 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: > >> Sure it is! >> >> This is exactly what I was looking for! Actually I wasn't able to see >> that there was a way to perform this action! >> >> Thanks for your help, Olle! >> >> I will try to create this, and when I have it done, I will post my script >> here, so if someone else need the same problem I can help. >> >> Now, I have just one more question about my scenario: when I register a >> user, is there a way to create something like a "custom field" on the >> location? To explain better: If possible, I can add something like a custom >> header that says if the user is registered from a phone device or the web >> interface, then I can perform the loop and replace only the oldest contact >> using that device... >> >> >> >> *Bruno Emer* >> >> Mobile: +55 11 96540-0044 >> email: brunoemer(a)gmail.com >> >> 2016-04-01 17:18 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: >> >>> This is my problem... I want to have 2 contacts for AOR, and not only >>> one. >>> >>> When the 3rd arrives, it must remove the oldest and continue with 2... >>> >>> *If I am using the 0x04 parameter, it will save just one contact, and >>> the user will not be able to get the Invite both, in the mobile device and >>> web interface.* >>> >>> *Bruno Emer* >>> >>> Mobile: +55 11 96540-0044 >>> email: brunoemer(a)gmail.com >>> <https://www.facebook.com/bruno.emer.5> >>> <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> >>> <https://twitter.com/brunoemer_> >>> <https://www.flickr.com/photos/122070309@N03/> >>> <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_> >>> >>> 2016-04-01 17:07 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: >>> >>>> Hello, *Olle!* >>>> >>>> *Thanks for your help in this case, and don't worry about the time >>>> (actually I wasn't even expecting to get an answer today).* >>>> >>>> >>>> *I know the max_contacts parameter. But the problem is that if I set >>>> the max_contacts to 2, when the user tries to register again, he gets an >>>> error message 503. The point is that I want to get the user registered, and >>>> replace the oldest location entry for the newest one.* >>>> >>>> >>>> >>>> *Bruno Emer* >>>> >>>> Mobile: +55 11 96540-0044 >>>> email: brunoemer(a)gmail.com >>>> >>>> 2016-04-01 16:31 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: >>>> >>>>> Hello all. >>>>> >>>>> I have problem here and I tried to find a solution and search over >>>>> internet, but without success. >>>>> >>>>> My scenario is the following: I have an application that must be >>>>> registered in Kamailio when a user logs in the web interface, so he can get >>>>> calls (something like a web softphone using webrtc). At this point, we are >>>>> OK, and everything is working fine. >>>>> >>>>> To get these register functions I am using the parameter >>>>> "save("location", "0x04")" as described in the REGISTRAR module >>>>> documentation, so if a user logs in another web browser or computer, only >>>>> the last one will continue registered and all calls will be forwarded to >>>>> him. >>>>> >>>>> The point is that now we are creating a phone app that will do almost >>>>> the same thing as the web interface, allowing users to receive calls using >>>>> the mobile device, and here is my problem: I want to allow my users to be >>>>> registered on two devices at the same time, but if a user logs into another >>>>> device, I don't want to reply with a 503. I want to allow the user to >>>>> register again, deregistering the oldest contact. >>>>> >>>>> I saw that there is a module named "ims_usrloc_scscf" and on its >>>>> description it says "implemented overwrite oldest contact behaviour", >>>>> but I couldn't find any documentation about it. >>>>> >>>>> So, is there a way to get this working today? >>>>> >>>>> >>>>> *Bruno Emer* >>>>> >>>>> Mobile: +55 11 96540-0044 >>>>> email: brunoemer(a)gmail.com >>>>> >>>> >>>> >>> >> >

1 0

Kamailio 4.3.3 and HUAWEI HG8245. Port 1 call port 2. 481 Call/Transaction Does Not Exist. Warning: 399 No Dialog match.
by Mx R 17 May '16

17 May '16

Hi there. In our network we used several ONT Huawei models. But with one model HUAWEI-EchoLife HG8245 / V1R003C00S205 we have the following problem. When connecting to this device in the POTS ports of the two phones. From the first port we can not call to the second and from the second port can not call to the first port. On the other ONT Huawei devices, there is no such problem. Earlier we had set OpenSIP and this device did not have a problem. The problem is likely in the configuration because it works at the default settings. kamailio -v Version: kamailio 4.3.3 (x86_64/linux) f69971 /var/log/kamailio May 17 10:53:21 kamailio /usr/local/sbin/kamailio[17875]: INFO: <script>: Callee '123456' is local - M=INVITE RURI=sip:123456@10.10.10.200 F= sip:654321@10.10.10.200 T=sip:123456@10.10.10.200 IP=10.14.197.13 5 ID=lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 May 17 10:53:21 kamailio /usr/local/sbin/kamailio[17875]: INFO: <script>: Call control: dst user '123456' currently has [0/2] active calls - M=INVITE RURI=sip:123456@10.10.10.200 F=sip:654321@10.10.10.200 T=si p:123456@10.10.10.200 IP=10.10.10.100 ID= lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 May 17 10:53:21 kamailio /usr/local/sbin/kamailio[17875]: INFO: <script>: Normalize source: sip:654321@10.10.10.200 -> sip:654321@10.10.10.200 May 17 10:53:21 kamailio /usr/local/sbin/kamailio[17875]: INFO: <script>: Request leaving server - M=INVITE RURI=sip:123456@10.10.10.100:5060;transport=udp F=sip:654321@10.10.10.200 T=sip:123456@178.168.4. 37 IP=10.10.10.100 ID=lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 May 17 10:53:22 kamailio /usr/local/sbin/kamailio[17873]: INFO: <script>: Reply - S=100 D=Trying F=sip:654321@10.10.10.200 T=sip:123456@10.10.10.200 IP=10.10.10.100 ID=lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.1 4.197.135 May 17 10:53:22 kamailio /usr/local/sbin/kamailio[17861]: INFO: <script>: Reply - S=481 D=Call/Transaction Does Not Exist F=sip:654321@10.10.10.200 T= sip:123456@10.10.10.200 IP=10.10.10.100 ID= lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 sngrep raw 2016/05/17 10:53:21.514702 10.10.10.100:5060 -> 10.10.10.200:5060 INVITE sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bKz4j0izm0i8jcn2pmlkm4c24cz Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 1 INVITE Contact: <sip:654321@10.10.10.100:5060;transport=udp> Supported: 100rel,replaces,timer,privacy,in-dialog Session-Expires: 1800 Accept: application/sdp,application/simservs+xml Min-SE: 90 Accept-Encoding: identity Accept-Language: en Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Max-Forwards: 70 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Session-ID: 20zlkckj1kz4ll1i1ic2pl1csk80kli1 Content-Length: 318 Content-Type: application/sdp v=0 o=huawei 1 1 IN IP4 10.10.10.100 s=- c=IN IP4 10.10.10.100 t=0 0 m=audio 50000 RTP/AVP 8 0 18 9 97 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:9 G722/8000 a=ptime:20 a=rtpmap:97 telephone-event/8000/1 a=sendrecv a=fmtp:97 0-15,32-36 2016/05/17 10:53:21.515342 10.10.10.200:5060 -> 10.10.10.100:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bKz4j0izm0i8jcn2pmlkm4c24cz Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 1 INVITE Server: sip.local Content-Length: 0 2016/05/17 10:53:21.517879 10.10.10.200:5060 -> 10.10.10.100:5060 SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bKz4j0izm0i8jcn2pmlkm4c24cz Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=9477874460dbcfddfdce8a8a24ab1a42.1303 CSeq: 1 INVITE Proxy-Authenticate: Digest realm="10.10.10.200", nonce="VzrPHVc6zfEYftiGlJYC9XLb4ixCZtQB", qop="auth" Server: sip.local Content-Length: 0 2016/05/17 10:53:21.564365 10.10.10.100:5060 -> 10.10.10.200:5060 ACK sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bKz4j0izm0i8jcn2pmlkm4c24cz Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=9477874460dbcfddfdce8a8a24ab1a42.1303 CSeq: 1 ACK Max-Forwards: 70 Content-Length: 0 2016/05/17 10:53:21.590008 10.10.10.100:5060 -> 10.10.10.200:5060 INVITE sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Contact: <sip:654321@10.10.10.100:5060;transport=udp> Supported: 100rel,replaces,timer,privacy,in-dialog Session-Expires: 1800 Accept: application/sdp,application/simservs+xml Min-SE: 90 Proxy-Authorization: Digest username="654321",realm="10.10.10.200",nonce="VzrPHVc6zfEYftiGlJYC9XLb4ixCZtQB",uri=" sip:123456@10.10.10.200 ",response="55e03b278f1cbdda52852004bbc8d7c9",algorithm=MD5,cnonce="0f31d b",qop=auth,nc=00000006 Accept-Encoding: identity Accept-Language: en Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Max-Forwards: 70 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Session-ID: 20zlkckj1kz4ll1i1ic2pl1csk80kli1 Content-Length: 318 Content-Type: application/sdp v=0 o=huawei 1 2 IN IP4 10.10.10.100 s=- c=IN IP4 10.10.10.100 t=0 0 m=audio 50000 RTP/AVP 8 0 18 9 97 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:9 G722/8000 a=ptime:20 a=rtpmap:97 telephone-event/8000/1 a=sendrecv a=fmtp:97 0-15,32-36 2016/05/17 10:53:21.590808 10.10.10.200:5060 -> 10.10.10.100:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Server: sip.local Content-Length: 0 2016/05/17 10:53:21.642831 10.10.10.200:5060 -> 10.10.10.100:5060 INVITE sip:123456@10.10.10.100:5060;transport=udp SIP/2.0 Record-Route: <sip:10.10.10.200;lr=on;ftag=pk4nzz3j;nat=yes;did=925.a5a1;vst=AAAAAAkLCwgLCnEDBHgfBg4WHwIWHQMuMzc-;vsf=AAAAAAkLAwgLAnEDBHgfBg4WHwIWHQMuMzc-> Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0 Via: SIP/2.0/UDP 10.10.10.100:5060;rport=5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321" <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Contact: <sip:654321@10.10.10.100:5060;transport=udp> Supported: 100rel,replaces,timer,privacy,in-dialog Session-Expires: 1800 Accept: application/sdp,application/simservs+xml Min-SE: 90 Accept-Encoding: identity Accept-Language: en Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Max-Forwards: 69 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Content-Length: 334 Content-Type: application/sdp P-Asserted-Identity: <sip:654321@10.10.10.200> v=0 o=huawei 1 2 IN IP4 10.10.10.200 s=- c=IN IP4 10.10.10.200 t=0 0 m=audio 57816 RTP/AVP 8 0 18 9 97 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:9 G722/8000 a=ptime:20 a=rtpmap:97 telephone-event/8000/1 a=sendrecv a=fmtp:97 0-15,32-36 a=nortpproxy:yes 2016/05/17 10:53:22.121872 10.10.10.200:5060 -> 10.10.10.100:5060 INVITE sip:123456@10.10.10.100:5060;transport=udp SIP/2.0 Record-Route: <sip:10.10.10.200;lr=on;ftag=pk4nzz3j;nat=yes;did=925.a5a1;vst=AAAAAAkLCwgLCnEDBHgfBg4WHwIWHQMuMzc-;vsf=AAAAAAkLAwgLAnEDBHgfBg4WHwIWHQMuMzc-> Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0 Via: SIP/2.0/UDP 10.10.10.100:5060;rport=5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321" <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Contact: <sip:654321@10.10.10.100:5060;transport=udp> Supported: 100rel,replaces,timer,privacy,in-dialog Session-Expires: 1800 Accept: application/sdp,application/simservs+xml Min-SE: 90 Accept-Encoding: identity Accept-Language: en Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Max-Forwards: 69 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Content-Length: 334 Content-Type: application/sdp P-Asserted-Identity: <sip:654321@10.10.10.200> v=0 o=huawei 1 2 IN IP4 10.10.10.200 s=- c=IN IP4 10.10.10.200 t=0 0 m=audio 57816 RTP/AVP 8 0 18 9 97 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:9 G722/8000 a=ptime:20 a=rtpmap:97 telephone-event/8000/1 a=sendrecv a=fmtp:97 0-15,32-36 a=nortpproxy:yes 2016/05/17 10:53:22.990080 10.10.10.100:5060 -> 10.10.10.200:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0,SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il;rport=5060 Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321"<sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Content-Length: 0 2016/05/17 10:53:22.996850 10.10.10.100:5060 -> 10.10.10.200:5060 SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0,SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il;rport=5060 Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321"<sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=m2scp1cm CSeq: 2 INVITE Warning: 399 10.10.10.100 "SS100006F143L3557[05552] No Dialog match" User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Content-Length: 0 2016/05/17 10:53:22.997476 10.10.10.200:5060 -> 10.10.10.100:5060 ACK sip:123456@10.10.10.100:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0 Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321" <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=m2scp1cm CSeq: 2 ACK Max-Forwards: 69 Content-Length: 0 2016/05/17 10:53:36.282450 10.10.10.100:5060 -> 10.10.10.200:5060 CANCEL sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 CANCEL Max-Forwards: 70 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Session-ID: 20zlkckj1kz4ll1i1ic2pl1csk80kli1 Content-Length: 0 2016/05/17 10:53:36.283389 10.10.10.200:5060 -> 10.10.10.100:5060 SIP/2.0 200 ok -- no more pending branches Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=b79688aec473b5156b1a9f919edc0e49-0b87 CSeq: 2 CANCEL Server: sip.local Content-Length: 0 2016/05/17 10:53:53.040976 10.10.10.200:5060 -> 10.10.10.100:5060 SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il;rport=5060 Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=m2scp1cm CSeq: 2 INVITE Warning: 399 10.10.10.100 "SS100006F143L3557[05552] No Dialog match" User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Content-Length: 0 2016/05/17 10:53:53.051396 10.10.10.100:5060 -> 10.10.10.200:5060 ACK sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=m2scp1cm CSeq: 2 ACK Max-Forwards: 70 Content-Length: 0

1 0

Enum module and parallel branching
by Loic Chabert 17 May '16

17 May '16

Hello, I'm trying to use enum module at core level to route my numbers to end-point asterisk servers. As i read (and validated by tests), when a phone number has two DNS NAPTER entries, enum module create two branch, one for each entries, and send request to asterisk servers (in the same time). Do you know any tricks or functions to disable parallel branching, and make serial branching instead (following responses returned by asterisk) ? I have also change preference and order in dns config file, same results. Thanks in advance. Regards.

2 3

MySQL database for kamailio as S-CSCF
by Christoph Valentin 17 May '16

17 May '16

Hi Community, I'm doing very first steps to use kamailio as an S-CSCF. Now I'm confused about some defines in the examples/scscf/scscf.cfg file. #!define DB_URL "con1=>mysql://scscf:heslo@xxx.xxx.xxx.xxx/scscf" ##!define DB_URL2 "con2=>mysql://scscf:heslo@10.1.27.217/scscf" Tutorials/ims/Installation-howto tell me only I-CSCF needs a database, I do not find anything about database for S-CSCF. However, when I comment out the defines ##!define DB_URL "con1=>mysql://scscf:heslo@xxx.xxx.xxx.xxx/scscf" ##!define DB_URL2 "con2=>mysql://scscf:heslo@10.1.27.217/scscf" then the kamctl start results in a segmentation fault. I have installed kamailio-4.4.0 from RPMs. Is there any hint about database for S-CSCF? Thanks in advance Christoph

1 1

Kamailio 4.3.3 and HUAWEI HG8245. Port 1 call port 2. 481 Call/Transaction Does Not Exist. Warning: 399 No Dialog match.
by M R 17 May '16

17 May '16

Hi there. In our network we used several ONT Huawei models. But with one model HUAWEI-EchoLife HG8245 / V1R003C00S205 we have the following problem. When connecting to this device in the POTS ports of the two phones. From the first port we can not call to the second and from the second port can not call to the first port. On the other ONT Huawei devices, there is no such problem. Earlier we had set OpenSIP and this device did not have a problem. The problem is likely in the configuration because it works at the default settings. kamailio -v Version: kamailio 4.3.3 (x86_64/linux) f69971 /var/log/kamailio May 17 10:53:21 kamailio /usr/local/sbin/kamailio[17875]: INFO: <script>: Callee '123456' is local - M=INVITE RURI=sip:123456@10.10.10.200 F= sip:654321@10.10.10.200 T=sip:123456@10.10.10.200 IP=10.14.197.13 5 ID=lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 May 17 10:53:21 kamailio /usr/local/sbin/kamailio[17875]: INFO: <script>: Call control: dst user '123456' currently has [0/2] active calls - M=INVITE RURI=sip:123456@10.10.10.200 F=sip:654321@10.10.10.200 T=si p:123456@10.10.10.200 IP=10.10.10.100 ID= lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 May 17 10:53:21 kamailio /usr/local/sbin/kamailio[17875]: INFO: <script>: Normalize source: sip:654321@10.10.10.200 -> sip:654321@10.10.10.200 May 17 10:53:21 kamailio /usr/local/sbin/kamailio[17875]: INFO: <script>: Request leaving server - M=INVITE RURI=sip:123456@10.10.10.100:5060;transport=udp F=sip:654321@10.10.10.200 T=sip:123456@178.168.4. 37 IP=10.10.10.100 ID=lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 May 17 10:53:22 kamailio /usr/local/sbin/kamailio[17873]: INFO: <script>: Reply - S=100 D=Trying F=sip:654321@10.10.10.200 T=sip:123456@10.10.10.200 IP=10.10.10.100 ID=lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.1 4.197.135 May 17 10:53:22 kamailio /usr/local/sbin/kamailio[17861]: INFO: <script>: Reply - S=481 D=Call/Transaction Does Not Exist F=sip:654321@10.10.10.200 T= sip:123456@10.10.10.200 IP=10.10.10.100 ID= lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 sngrep raw 2016/05/17 10:53:21.514702 10.10.10.100:5060 -> 10.10.10.200:5060 INVITE sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bKz4j0izm0i8jcn2pmlkm4c24cz Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 1 INVITE Contact: <sip:654321@10.10.10.100:5060;transport=udp> Supported: 100rel,replaces,timer,privacy,in-dialog Session-Expires: 1800 Accept: application/sdp,application/simservs+xml Min-SE: 90 Accept-Encoding: identity Accept-Language: en Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Max-Forwards: 70 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Session-ID: 20zlkckj1kz4ll1i1ic2pl1csk80kli1 Content-Length: 318 Content-Type: application/sdp v=0 o=huawei 1 1 IN IP4 10.10.10.100 s=- c=IN IP4 10.10.10.100 t=0 0 m=audio 50000 RTP/AVP 8 0 18 9 97 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:9 G722/8000 a=ptime:20 a=rtpmap:97 telephone-event/8000/1 a=sendrecv a=fmtp:97 0-15,32-36 2016/05/17 10:53:21.515342 10.10.10.200:5060 -> 10.10.10.100:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bKz4j0izm0i8jcn2pmlkm4c24cz Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 1 INVITE Server: sip.local Content-Length: 0 2016/05/17 10:53:21.517879 10.10.10.200:5060 -> 10.10.10.100:5060 SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bKz4j0izm0i8jcn2pmlkm4c24cz Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=9477874460dbcfddfdce8a8a24ab1a42.1303 CSeq: 1 INVITE Proxy-Authenticate: Digest realm="10.10.10.200", nonce="VzrPHVc6zfEYftiGlJYC9XLb4ixCZtQB", qop="auth" Server: sip.local Content-Length: 0 2016/05/17 10:53:21.564365 10.10.10.100:5060 -> 10.10.10.200:5060 ACK sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bKz4j0izm0i8jcn2pmlkm4c24cz Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=9477874460dbcfddfdce8a8a24ab1a42.1303 CSeq: 1 ACK Max-Forwards: 70 Content-Length: 0 2016/05/17 10:53:21.590008 10.10.10.100:5060 -> 10.10.10.200:5060 INVITE sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Contact: <sip:654321@10.10.10.100:5060;transport=udp> Supported: 100rel,replaces,timer,privacy,in-dialog Session-Expires: 1800 Accept: application/sdp,application/simservs+xml Min-SE: 90 Proxy-Authorization: Digest username="654321",realm="10.10.10.200",nonce="VzrPHVc6zfEYftiGlJYC9XLb4ixCZtQB",uri=" sip:123456@10.10.10.200 ",response="55e03b278f1cbdda52852004bbc8d7c9",algorithm=MD5,cnonce="0f31d b",qop=auth,nc=00000006 Accept-Encoding: identity Accept-Language: en Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Max-Forwards: 70 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Session-ID: 20zlkckj1kz4ll1i1ic2pl1csk80kli1 Content-Length: 318 Content-Type: application/sdp v=0 o=huawei 1 2 IN IP4 10.10.10.100 s=- c=IN IP4 10.10.10.100 t=0 0 m=audio 50000 RTP/AVP 8 0 18 9 97 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:9 G722/8000 a=ptime:20 a=rtpmap:97 telephone-event/8000/1 a=sendrecv a=fmtp:97 0-15,32-36 2016/05/17 10:53:21.590808 10.10.10.200:5060 -> 10.10.10.100:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Server: sip.local Content-Length: 0 2016/05/17 10:53:21.642831 10.10.10.200:5060 -> 10.10.10.100:5060 INVITE sip:123456@10.10.10.100:5060;transport=udp SIP/2.0 Record-Route: <sip:10.10.10.200;lr=on;ftag=pk4nzz3j;nat=yes;did=925.a5a1;vst=AAAAAAkLCwgLCnEDBHgfBg4WHwIWHQMuMzc-;vsf=AAAAAAkLAwgLAnEDBHgfBg4WHwIWHQMuMzc-> Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0 Via: SIP/2.0/UDP 10.10.10.100:5060;rport=5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321" <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Contact: <sip:654321@10.10.10.100:5060;transport=udp> Supported: 100rel,replaces,timer,privacy,in-dialog Session-Expires: 1800 Accept: application/sdp,application/simservs+xml Min-SE: 90 Accept-Encoding: identity Accept-Language: en Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Max-Forwards: 69 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Content-Length: 334 Content-Type: application/sdp P-Asserted-Identity: <sip:654321@10.10.10.200> v=0 o=huawei 1 2 IN IP4 10.10.10.200 s=- c=IN IP4 10.10.10.200 t=0 0 m=audio 57816 RTP/AVP 8 0 18 9 97 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:9 G722/8000 a=ptime:20 a=rtpmap:97 telephone-event/8000/1 a=sendrecv a=fmtp:97 0-15,32-36 a=nortpproxy:yes 2016/05/17 10:53:22.121872 10.10.10.200:5060 -> 10.10.10.100:5060 INVITE sip:123456@10.10.10.100:5060;transport=udp SIP/2.0 Record-Route: <sip:10.10.10.200;lr=on;ftag=pk4nzz3j;nat=yes;did=925.a5a1;vst=AAAAAAkLCwgLCnEDBHgfBg4WHwIWHQMuMzc-;vsf=AAAAAAkLAwgLAnEDBHgfBg4WHwIWHQMuMzc-> Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0 Via: SIP/2.0/UDP 10.10.10.100:5060;rport=5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321" <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Contact: <sip:654321@10.10.10.100:5060;transport=udp> Supported: 100rel,replaces,timer,privacy,in-dialog Session-Expires: 1800 Accept: application/sdp,application/simservs+xml Min-SE: 90 Accept-Encoding: identity Accept-Language: en Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Max-Forwards: 69 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Content-Length: 334 Content-Type: application/sdp P-Asserted-Identity: <sip:654321@10.10.10.200> v=0 o=huawei 1 2 IN IP4 10.10.10.200 s=- c=IN IP4 10.10.10.200 t=0 0 m=audio 57816 RTP/AVP 8 0 18 9 97 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:18 G729/8000 a=ptime:20 a=rtpmap:9 G722/8000 a=ptime:20 a=rtpmap:97 telephone-event/8000/1 a=sendrecv a=fmtp:97 0-15,32-36 a=nortpproxy:yes 2016/05/17 10:53:22.990080 10.10.10.100:5060 -> 10.10.10.200:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0,SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il;rport=5060 Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321"<sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 INVITE Content-Length: 0 2016/05/17 10:53:22.996850 10.10.10.100:5060 -> 10.10.10.200:5060 SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0,SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il;rport=5060 Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321"<sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=m2scp1cm CSeq: 2 INVITE Warning: 399 10.10.10.100 "SS100006F143L3557[05552] No Dialog match" User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Content-Length: 0 2016/05/17 10:53:22.997476 10.10.10.200:5060 -> 10.10.10.100:5060 ACK sip:123456@10.10.10.100:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bKd55d.f4d5d1ee3d090fe847af101d0a47ac5e.0 Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: "654321" <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=m2scp1cm CSeq: 2 ACK Max-Forwards: 69 Content-Length: 0 2016/05/17 10:53:36.282450 10.10.10.100:5060 -> 10.10.10.200:5060 CANCEL sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200> CSeq: 2 CANCEL Max-Forwards: 70 User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Session-ID: 20zlkckj1kz4ll1i1ic2pl1csk80kli1 Content-Length: 0 2016/05/17 10:53:36.283389 10.10.10.200:5060 -> 10.10.10.100:5060 SIP/2.0 200 ok -- no more pending branches Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=b79688aec473b5156b1a9f919edc0e49-0b87 CSeq: 2 CANCEL Server: sip.local Content-Length: 0 2016/05/17 10:53:53.040976 10.10.10.200:5060 -> 10.10.10.100:5060 SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il;rport=5060 Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=m2scp1cm CSeq: 2 INVITE Warning: 399 10.10.10.100 "SS100006F143L3557[05552] No Dialog match" User-Agent: HUAWEI-EchoLife HG8245/V1R003C00S205 Content-Length: 0 2016/05/17 10:53:53.051396 10.10.10.100:5060 -> 10.10.10.200:5060 ACK sip:123456@10.10.10.200 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK0lz4ll1ikm1k02p20lz4ll1il Route: <sip:10.10.10.200:5060;transport=udp;lr> Call-ID: lps4s2204mz4ll1i4mz4ll1ici0lc81i(a)10.10.10.100 From: <sip:654321@10.10.10.200>;tag=pk4nzz3j To: <sip:123456@10.10.10.200>;tag=m2scp1cm CSeq: 2 ACK Max-Forwards: 70 Content-Length: 0

1 0

Debian, dnssec.so, and missing dnsval.conf
by David Holl 17 May '16

17 May '16

In the interest of sharing and promoting DNSSEC, I was recently stumped on a problem where I noticed I could not call sip:test.echo@sip5060.net without disabling the dnssec.so module in my kamailio.cfg. The last time I had tried the sip5060 echo test was several months ago, and in the intervening time, I had applied the numerous debian package updates. Thus, I could not isolate the cause of the problem to one of the many package updates such as kamailio 4.4.x, libval, my own DNSSEC dns server, etc... My initial only clue was kamailio's syslog messages: May 12 18:02:49 tensor /usr/sbin/kamailio[24128]: ERROR: tm [ut.h:296]: uri2dst2(): ERROR: uri2dst: failed to resolve "sip5060.net" :unresolvable A or AAAA request (-7) May 12 18:02:49 tensor /usr/sbin/kamailio[24128]: ERROR: tm [t_fwd.c:1712]: t_forward_nonack(): ERROR: t_forward_nonack: failure to add branches May 12 18:02:49 tensor /usr/sbin/kamailio[24128]: ERROR: sl [sl_funcs.c:363]: sl_reply_error(): ERROR: sl_reply_error used: Unresolvable destination (478/SL) To summarize the problem, I found that kamailio's dnssec.so was failing queries to domains that were not DNSSEC protected such as sip5060.net. This behavior is a bug, since sip5060.net is in this case, provably not participating in DNSSEC from the toplevel "net." zone [1]. The fix was to provide a default configuration file to libval, which was somehow not included in Debian's libval15 package. I created the following configuration from the default dnsval.conf in the source distribution: https://dnssec-tools.org/download/dnsval-2.2.tar.gz https://dnssec-tools.org/download/dnsval-2.2.tar.gz.asc as well as their wiki: https://www.dnssec-tools.org/wiki/index.php/Dnsval.conf cat << 'EOF' > /etc/dnssec-tools/dnsval.conf ################################## # Global Options ################################## global-options trust-oob-answers yes # I'm running a DNSSEC nameserver locally, # so 4 KiB queries to localhost should be OK: edns0-size 4096 env-policy disable app-policy disable rec-fallback no ; : trust-anchor . DS 19036 8 2 49AAC11D7B6F6446702E54A1607371607A1A41855200FD2CE1CDDE32F24E8FB5 ; : zone-security-expectation . validate ; : provably-insecure-status . trusted ; : clock-skew . 0 ; EOF In summary: I find it odd that a default dnsval.conf was not included with my system's normal installation of the "libval15:amd64" package, version 2.2-4. After providing dnsval.conf, kamailio's dnssec.so was happy again. Side notes: * To identify the directory for dnsval.conf, run libval-config --dnsval-config On Debian, libval-config is in the libval-dev pacakge, and this package includes a man page for dnsval.conf as well. * To test your libval and dnsval.conf, try dt-validate -s On Debian, dt-validate is in the libval-bin package. - David [1] -- Authenticated denial of existence: https://tools.ietf.org/html/rfc5155 https://tools.ietf.org/html/rfc7129 And for sip5060.net, there's an NSEC3 record in the top level net. zone proving non-existance of sip5060.net DS records: http://dnsviz.net/d/sip5060.net/dnssec/ Translation: sip5060.net provably not enabled DNSEC.

2 1

kamailio not authenticate a subscribed username
by Al S 16 May '16

16 May '16

Hi,1) A generic subscriber is added to kamailio database:kamctl add pbx password2) Our UAC simulator (sipp) sends a call from a non subscribed username=8181111111@local_domain using a subscribed username for authentication (username=pbx, realm=local_domain) but kamailio does not authenticated the call:UAC =====> INVITE (From username=818111111@local_domain) ===> kamailioUAC <===== 407 <===== KamailioUAC =====> INVITE (AUTH username=pbx, realm=local_domain, nonce=???, response=????) =====> kamailio(kamailio route[AUTH} module does not authenticate it)The correct password is being sent to kamailio.Your assistance is greatly appreciatedThanks,AR

1 0

kamailio not authenticate a call with subscribed username
by Al S 16 May '16

16 May '16

Hi, 1) A generic subscriber is added to kamailio database:kamctl add pbx password 2) Our UAC simulator (sipp) sends a call from a non subscribed username=8181111111@local_domain using a subscribed username for authentication (username=pbx, realm=local_domain) but kamailio does not authenticated the call: UAC =====> INVITE (From username=818111111@local_domain) ===> kamailioUAC <===== 407 <===== KamailioUAC =====> INVITE (AUTH username=pbx, realm=local_domain, nonce=???, response=????) =====> kamailio(kamailio route[AUTH} module does not authenticate it) The correct password is being sent to kamailio. Your assistance is greatly appreciated Thanks,AR

1 0

Re: [SR-Users] Another question about RTPengine module
by Dmitry 16 May '16

16 May '16

the problem is: rtpengine was compiled with USE_SCTP,but kamailio that was installed via RPM is without this flag. How to tell compiler NOT TO " USE USE_SCTP" flag? +++++++++++++++++++++++ Makefile.defs:SCTP ?= 0 - and it compiles without sctp. Now kamailio.log:ERROR: rtpengine [rtpengine.c:1113]: child_init(): Name or service not knownERROR: <core> [sr_module.c:929]: init_mod_child(): Error while initializing module rtpengine (/usr/lib64/kamailio/modules/rtpengine.so) View this message in context: Re: Another question about RTPengine module Sent from the Users mailing list archive at Nabble.com. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list [hidden email] http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list [hidden email] http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users If you reply to this email, your message will be added to the discussion below: http://sip-router.1086192.n5.nabble.com/Another-question-about-RTPengine-mo… To unsubscribe from Another question about RTPengine module, click here. NAML -- View this message in context: Re: Another question about RTPengine module Sent from the Users mailing list archive at Nabble.com. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list [hidden email] http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list [hidden email] http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users If you reply to this email, your message will be added to the discussion below: http://sip-router.1086192.n5.nabble.com/Another-question-about-RTPengine-mo… To unsubscribe from Another question about RTPengine module, click here. NAML -- View this message in context: Re: Another question about RTPengine module Sent from the Users mailing list archive at Nabble.com. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users On Sunday, May 15, 2016 6:21 PM, Dmitry <mbike2000ru(a)yahoo.com> wrote: yes, I wrote that modparam are recognized.MODPARAM must be placed i the sae place where the RTPPROXY resides in the config.+++++++++++++++++++++++++++++++ Another question:kazoo kamailio KAMAILIO-V :4.3.5:kamailio -vversion: kamailio 4.3.5 (x86_64/linux)flags: STATS: Off, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RESADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MBpoll method support: poll, epoll_lt, epoll_et, sigio_rt, select.id: unknowncompiled on 17:51:55 Mar 8 2016 with gcc 4.4.7 To add the RTPENGINE - I found Kamailio 4.3.5 sources https://www.kamailio.org/pub/kamailio/4.3.5/src/ When I start kamailio I get the following error:ERROR: <core> [sr_module.c:431]: version_control(): module compile flags mismatch for /usr/lib64/kamailio/modules/rtpengine.so I did the following to add missing modules: #make include_modules="rtpengine topoh" cfg #make all cp /usr/src/kamailio-4.3.5/modules/rtpengine/rtpengine.so /usr/lib64/kamailio/modules/rtpengine.so How to fix it? The sources version seems to be the same On Saturday, May 14, 2016 1:21 AM, ycaner <yasin.caner(a)netgsm.com.tr> wrote: Hello; i am not sure. i downloaded from git kamailio. add rtpengine instead of rtpproxy . it works fine. it is hard to solve problem without log. Cheers. try to run with kamailio -E -dd , maybe it gives information. check kamailio.cfg is that rtpengine is loaded like loadmodule "rtpengine.so" 2016-05-13 11:35 tarihinde Dmitry [via SIP Router] yazdı: one more question about rtpengine. I installed Kamailio (centos 6.7) from source as You recommended make include_modules="db_mysql utils regex dnssec jansson lcr xmlrpc rtpengine " cfg no rtpengine daemon (it is not installed) but the picture is the same: modparam ("rtpengine,....) kamailio -c loading modules under config path: /usr/local/lib64/kamailio/modules/ 0(19523) ERROR: <core> [modparam.c:150]: set_mod_param_regex(): No module matching <rtpengine> found it means that it does not depend upon rtpengine daemon. # single rtproxy modparam("rtpengine", "rtpengine_sock", "udp:localhost:12221") I copied it from http://www.kamailio.net/docs/modules/4.2.x/modules/rtpengine.html STRACE kamctl start -says nothing anout rtpengine What else is it possible to do to find the cause of the problem? On Friday, May 13, 2016 11:45 AM, ycaner <[hidden email]> wrote: Hello; rtpengine module isnt default. you need to add with "make include_modules". after make && make install , automatically it will be copied to lib/modules. Cheers. 2016-05-13 09:42 tarihinde Dmitry [via SIP Router] yazdı: And one more question on it: If the Kamailio was installed from RPM and rtpengine.so is missing - May I add rtpengine.so the following way: make include_modules="rtpengine" cfg make all and then copy rtpengine.so to /modules/ directory Is this way (adding missing modules) is correct? On Friday, May 13, 2016 11:25 AM, Dmitry <[hidden email]> wrote: hello I tested on Centos 6.7 as well as Ubuntu server - the problem is the same. RTPENGINE was installed both from dpkg and from Source (manual compilation) So I need a clarification on it. RTPENGINE daemon is started. The Kamailio module itself is taken from /usr/src/kamailio, where the source was compiled. mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules/" cp /usr/src/kamailio/modules/rtpengine/rtpengine.so /usr/lib/x86_64-linux-gnu/kamailio/modules/rtpengine.so sudo /etc/init.d/kamailio restart | grep rtp ERROR: <core> [modparam.c:163]: set_mod_param_regex(): set_mod_param_regex: No module matching <rtpengine> found version: kamailio 5.0.0-dev4 (x86_64/linux) Any help is appreciated. On Thursday, May 12, 2016 6:04 PM, Dmitry <[hidden email]> wrote: Hi, I installed Kamailio from GIT. I did make include_modules="db_mysql utils regex dnssec jansson lcr xmlrpc rtpengine " cfg Actually loadmodule "rtpengine.so" does not affect kamailio start/restart. modparam("rtpengine", "rtpengine_sock", "udp:localhost:12221") causes this failure: [modparam.c:150]: set_mod_param_regex(): No module matching <rtpengine> found On Thursday, May 12, 2016 12:26 PM, ycaner <[hidden email]> wrote: Hello; on bash , kamailio -V gives Version , kamailio -c gives debug result. you can ask from here https://groups.google.com/forum/#!forum/2600hz-dev for kazoo rpm. if there is something wrong , they will fix it. you can install from source code from here => https://www.kamailio.org/w/ and "make include_modules="db_mysql utils regex dnssec jansson lcr xmlrpc rtpengine " cfg " in kamailio folder before 'make all' , helps you add modules. if there is a dependecies is missing , gives error. Cheers. 2016-05-11 19:51 tarihinde Dmitry [via SIP Router] yazdı: I installed Kamailio : yum install -y kazoo-kamailio https://2600hz.atlassian.net/wiki/display/Dedicated/via+RPM git clone https://github.com/sipwise/rtpengine.git the latest version of rtpengine and I do not know how to view the version of Kazoo-Kamailio On Wednesday, May 11, 2016 6:31 PM, ycaner <[hidden email]> wrote: hello ; maybe some devel are missing. which version using?i will try. cheers. -- View this message in context: http://sip-router.1086192.n5.nabble.com/Another-question-about-RTPengine-mo… Sent from the Users mailing list archive at Nabble.com. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list [hidden email] http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list [hidden email] http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users If you reply to this email, your message will be added to the discussion below: http://sip-router.1086192.n5.nabble.com/Another-question-about-RTPengine-mo… To unsubscribe from Another question about RTPengine module, click here. NAML --

1 4

Another question about RTPengine module
by Dmitry 15 May '16

15 May '16

I will clarify the question in my previous post.I installed Kamailio from 2600hz repo via yum (it is for kazoo). But when cp /usr/src/kamailio/modules/rtpengine/rtpengine.so /usr/lib64/kamailio/modules/rtpengine.so the kamailio does not see this moduleNo module matching <rtpengine> found Is there something like modules.list (it is used when I compile Kamailio manually) when the kamailio was installed via RPM?

2 12

WebRTC integration
by Moacir Ferreira 15 May '16

15 May '16

Hi, I am trying to use Kamailio with WebTRC to make and receive calls from the browser. Using the rpms from the Kamailio repository I have installed and tried the websocket config example from the source code. Using the first debug example from here https://www.kamailio.org/wiki/tutorials/tls/testing-and-debugginga I can see that my TLS wss setup is working. But I try JSCommunicator and sipML5 I get no success, they fail to register in Kamailio. From the logs I can see that something goes wrong with TLS. Doing some research, it is more like that several products are currently broken and not working... So can anyone help me (share a config and tricky) or point to a document that would help me to get it running? Thanks, Moacir

1 0

Re: [SR-Users] Multiple Registrations - Overwrite oldest contact
by Bruno Emer 13 May '16

13 May '16

Hello all! After some time I was able to get this working here, and now I am coming back to share my configuration with you guys (in case someone needs to get this working). Here is what we have done: route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } xlog("L_INFO", "checking $tu registered with User-Agent:$hdr(User-Agent)\n"); if(reg_fetch_contacts("location", "$tu", "caller")) { xlog("L_INFO", "$(ulc(caller=>count)) records found for $tu\n"); $var(i) = 0; while($var(i) < $(ulc(caller=>count))) { xlog("L_INFO", "record $var(i) with ruid:$(ulc(caller=>ruid)[$var(i)]) - User-Agent:$(ulc(caller=>user_agent)[$var(i)])"); if ($hdr(User-Agent)=~$(ulc(caller=>user_agent)[$var(i)])) { $var(ruid) = $(ulc(caller=>ruid)[$var(i)]); xlog("L_INFO", "unregistering $tu ruid:$var(ruid)"); unregister("location", "$tu", "$var(ruid)"); } $var(i) = $var(i) + 1; } } if (!save("location", "0x00")) sl_reply_error(); exit; } } There are some logs over there because we used it to see what was going on while testing the configurations. With this we can get just one register per User-Agent. And again, thanks Olle for your help in this problem *Bruno Emer* Mobile: +55 11 96540-0044 email: brunoemer(a)gmail.com <https://www.facebook.com/bruno.emer.5> <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> <https://twitter.com/brunoemer_> <https://www.flickr.com/photos/122070309@N03/> <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_> 2016-04-01 17:37 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: > Sure it is! > > This is exactly what I was looking for! Actually I wasn't able to see that > there was a way to perform this action! > > Thanks for your help, Olle! > > I will try to create this, and when I have it done, I will post my script > here, so if someone else need the same problem I can help. > > Now, I have just one more question about my scenario: when I register a > user, is there a way to create something like a "custom field" on the > location? To explain better: If possible, I can add something like a custom > header that says if the user is registered from a phone device or the web > interface, then I can perform the loop and replace only the oldest contact > using that device... > > > > *Bruno Emer* > > Mobile: +55 11 96540-0044 > email: brunoemer(a)gmail.com > > 2016-04-01 17:18 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: > >> This is my problem... I want to have 2 contacts for AOR, and not only >> one. >> >> When the 3rd arrives, it must remove the oldest and continue with 2... >> >> *If I am using the 0x04 parameter, it will save just one contact, and the >> user will not be able to get the Invite both, in the mobile device and web >> interface.* >> >> *Bruno Emer* >> >> Mobile: +55 11 96540-0044 >> email: brunoemer(a)gmail.com >> <https://www.facebook.com/bruno.emer.5> >> <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> >> <https://twitter.com/brunoemer_> >> <https://www.flickr.com/photos/122070309@N03/> >> <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_> >> >> 2016-04-01 17:07 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: >> >>> Hello, *Olle!* >>> >>> *Thanks for your help in this case, and don't worry about the time >>> (actually I wasn't even expecting to get an answer today).* >>> >>> >>> *I know the max_contacts parameter. But the problem is that if I set the >>> max_contacts to 2, when the user tries to register again, he gets an error >>> message 503. The point is that I want to get the user registered, and >>> replace the oldest location entry for the newest one.* >>> >>> >>> >>> *Bruno Emer* >>> >>> Mobile: +55 11 96540-0044 >>> email: brunoemer(a)gmail.com >>> >>> 2016-04-01 16:31 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: >>> >>>> Hello all. >>>> >>>> I have problem here and I tried to find a solution and search over >>>> internet, but without success. >>>> >>>> My scenario is the following: I have an application that must be >>>> registered in Kamailio when a user logs in the web interface, so he can get >>>> calls (something like a web softphone using webrtc). At this point, we are >>>> OK, and everything is working fine. >>>> >>>> To get these register functions I am using the parameter >>>> "save("location", "0x04")" as described in the REGISTRAR module >>>> documentation, so if a user logs in another web browser or computer, only >>>> the last one will continue registered and all calls will be forwarded to >>>> him. >>>> >>>> The point is that now we are creating a phone app that will do almost >>>> the same thing as the web interface, allowing users to receive calls using >>>> the mobile device, and here is my problem: I want to allow my users to be >>>> registered on two devices at the same time, but if a user logs into another >>>> device, I don't want to reply with a 503. I want to allow the user to >>>> register again, deregistering the oldest contact. >>>> >>>> I saw that there is a module named "ims_usrloc_scscf" and on its >>>> description it says "implemented overwrite oldest contact behaviour", >>>> but I couldn't find any documentation about it. >>>> >>>> So, is there a way to get this working today? >>>> >>>> >>>> *Bruno Emer* >>>> >>>> Mobile: +55 11 96540-0044 >>>> email: brunoemer(a)gmail.com >>>> >>> >>> >> >

2 1

Changing the SDP connection-address to a value in a pseudovariable
by Kevin McAllister 13 May '16

13 May '16

counter to the docs for nathelper fix_nated_sdp the config parser rejects any pseudovariables in the second parameter claiming they need to be constant. Does anyone know another way to change the "c=" line in an SDP? Or do I have to patch the nathelper module to include a free_fixup_function for the fix_nated_sdp stuff? This is on kamailio 4.2 Thanks, Kevin

1 1

Kamailio Crash
by Sunil More 13 May '16

13 May '16

Hello All, Kamailio instance just crashed with the following logs May 13 06:05:34 P172 /usr/local/kamailio/sbin/kamailio[12675]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received May 13 06:05:34 P172 /usr/local/kamailio/sbin/kamailio[12674]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received May 13 06:05:34 P172 /usr/local/kamailio/sbin/kamailio[12682]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received May 13 06:05:34 P172 /usr/local/kamailio/sbin/kamailio[12684]: INFO: <core> [main.c:809]: sig_usr(): signal 15 received May 13 06:05:34 P172 /usr/local/kamailio/sbin/kamailio[12667]: INFO: <core> [mem/f_malloc.c:598]: fm_free(): freeing a free fragment (0x7f23ff39cb88/0x7f23ff39cbd0) - ignore to add i have collected a gdb backtrace as below, root@IP:/home/corefiles# gdb /usr/local/kamailio/sbin/kamailio /home/c/core.kamailio.sig11.12671 chetan/ corefiles/ root@P172:/home/corefiles# gdb /usr/local/kamailio/sbin/kamailio /home/c/core.kamailio.sig11.12671 chetan/ corefiles/ root@P172:/home/corefiles# gdb /usr/local/kamailio/sbin/kamailio /home/corefiles/core.kamailio.sig11.12671 The version of Kamilio used by me is as below;. version: *kamailio 4.4.1 (x86_64/linux) 90be8b* flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 90be8b Thanking You, Sunil More Ph : 9503338275

2 2

Blacklists and DB_Cassandra
by SamyGo 13 May '16

13 May '16

Hi, I am tasked to make use of blacklist module for about 4 to 6 million numbers. I am thinking of using Cassandra for the purpose but reading through the documentatiom of module and recent mailing list discussion made me a bit hesitant. I am looking for advise on this whether this is going to perform as expected or is not even going to work with the BlackListing module ! I am using Cassandra 33x and Kamailio 4.4 over Ubuntu 14.04. Looking for suggestions here. Regards, Sammy

2 4

DB_Cassandra + UserBlacklist = ERROR
by SamyGo 12 May '16

12 May '16

Hi List, I'm trying to hook up userblacklist module with db_cassandra. I've kamailio keyspace configured with tables for userblacklist , globalblacklist, and version are created with some data in there. Here is setup info: version: kamailio 4.4.1 *kamailio.cfg* ... loadmodule "db_cassandra.so" modparam("db_cassandra", "schema_path","/etc/kamailio/kamailio") loadmodule "userblacklist.so" modparam("userblacklist", "db_url", "cassandra://:@127.0.0.1:9160/kamailio") modparam("userblacklist", "userblacklist_table", "userblacklist") modparam("userblacklist", "globalblacklist_table", "globalblacklist") *SCHEMA PATH:* root@whit-list:/etc/kamailio/kamailio# ls userblacklist version root@whit-list:/etc/kamailio/kamailio# cat version/version table_name(string) table_version(int) table_name root@whit-list:/etc/kamailio/kamailio# cat userblacklist/userblacklist id(int) username(string) domain(string) prefix(string) whitelist(int) id username *CASSANDRA DB* root@whit-list:/etc/kamailio/kamailio# cqlsh Connected to Test Cluster at 127.0.0.1:9042. [cqlsh 5.0.1 | Cassandra 3.5 | CQL spec 3.4.0 | Native protocol v4] Use HELP for help. cqlsh> use kamailio ... ; cqlsh:kamailio> describe tables; globalblacklist version userblacklist cqlsh:kamailio> select * from version; table_name | table_version -----------------+--------------- uacreg | 2 version | 1 globalblacklist | 1 userblacklist | 1 (4 rows) Every time I start up Kamailio I see the following debug logs: 0(13877) DEBUG: db_cassandra [dbcassa_table.c:572]: dbcassa_read_table_schemas(): Full name= /etc/kamailio/kamailio/ 0(13877) DEBUG: db_cassandra [dbcassa_table.c:589]: dbcassa_read_table_schemas(): Full dir name= /etc/kamailio/kamailio/version 0(13877) DEBUG: db_cassandra [dbcassa_table.c:609]: dbcassa_read_table_schemas(): Found database version 0(13877) DEBUG: db_cassandra [dbcassa_table.c:620]: dbcassa_read_table_schemas(): database table version 0(13877) DEBUG: db_cassandra [dbcassa_table.c:630]: dbcassa_read_table_schemas(): File path= /etc/kamailio/kamailio/version/version 0(13877) DEBUG: db_cassandra [dbcassa_table.c:200]: dbcassa_load_file(): loading file [/etc/kamailio/kamailio/version/version] 0(13877) DEBUG: db_cassandra [dbcassa_table.c:122]: dbcassa_table_new(): mtime is 1463068672 0(13877) DEBUG: db_cassandra [dbcassa_table.c:240]: dbcassa_load_file(): new col [table_name] 0(13877) DEBUG: db_cassandra [dbcassa_table.c:253]: dbcassa_load_file(): column[0] is STR! 0(13877) DEBUG: db_cassandra [dbcassa_table.c:283]: dbcassa_load_file(): column[0] is actually STRING! 0(13877) DEBUG: db_cassandra [dbcassa_table.c:240]: dbcassa_load_file(): new col [table_version] 0(13877) DEBUG: db_cassandra [dbcassa_table.c:253]: dbcassa_load_file(): column[1] is STR! 0(13877) DEBUG: db_cassandra [dbcassa_table.c:283]: dbcassa_load_file(): column[1] is actually STRING! 0(13877) DEBUG: db_cassandra [dbcassa_table.c:350]: dbcassa_load_file(): col [table_name] in primary key 0(13877) DEBUG: db_cassandra [dbcassa_table.c:589]: dbcassa_read_table_schemas(): Full dir name= /etc/kamailio/kamailio/userblacklist 0(13877) DEBUG: db_cassandra [dbcassa_table.c:609]: dbcassa_read_table_schemas(): Found database userblacklist 0(13877) DEBUG: db_cassandra [dbcassa_table.c:620]: dbcassa_read_table_schemas(): database table userblacklist 0(13877) DEBUG: db_cassandra [dbcassa_table.c:630]: dbcassa_read_table_schemas(): File path= /etc/kamailio/kamailio/userblacklist/userblacklist 0(13877) DEBUG: db_cassandra [dbcassa_table.c:200]: dbcassa_load_file(): loading file [/etc/kamailio/kamailio/userblacklist/userblacklist] 0(13877) DEBUG: db_cassandra [dbcassa_table.c:122]: dbcassa_table_new(): mtime is 1462985556 0(13877) DEBUG: db_cassandra [dbcassa_table.c:240]: dbcassa_load_file(): new col [id] 0(13877) DEBUG: db_cassandra [dbcassa_table.c:258]: dbcassa_load_file(): column[0] is INT! 0(13877) DEBUG: db_cassandra [dbcassa_table.c:240]: dbcassa_load_file(): new col [username] 0(13877) DEBUG: db_cassandra [dbcassa_table.c:253]: dbcassa_load_file(): column[1] is STR! 0(13877) DEBUG: db_cassandra [dbcassa_table.c:283]: dbcassa_load_file(): column[1] is actually STRING! 0(13877) DEBUG: db_cassandra [dbcassa_table.c:240]: dbcassa_load_file(): new col [domain] 0(13877) DEBUG: db_cassandra [dbcassa_table.c:253]: dbcassa_load_file(): column[2] is STR! 0(13877) DEBUG: db_cassandra [dbcassa_table.c:283]: dbcassa_load_file(): column[2] is actually STRING! 0(13877) DEBUG: db_cassandra [dbcassa_table.c:240]: dbcassa_load_file(): new col [prefix] 0(13877) DEBUG: db_cassandra [dbcassa_table.c:253]: dbcassa_load_file(): column[3] is STR! 0(13877) DEBUG: db_cassandra [dbcassa_table.c:283]: dbcassa_load_file(): column[3] is actually STRING! 0(13877) DEBUG: db_cassandra [dbcassa_table.c:240]: dbcassa_load_file(): new col [whitelist] 0(13877) DEBUG: db_cassandra [dbcassa_table.c:258]: dbcassa_load_file(): column[4] is INT! 0(13877) DEBUG: db_cassandra [dbcassa_table.c:350]: dbcassa_load_file(): col [id] in primary key 0(13877) DEBUG: db_cassandra [dbcassa_table.c:350]: dbcassa_load_file(): col [username] in primary key ..... 0(12780) DEBUG: db_cassandra [dbcassa_base.cpp:149]: db_cassa_new_connection(): opening connection: cassa:// xxxx:xxxx@127.0.0.1:9160/kamailio 0(12780) DEBUG: <core> [mem/f_malloc.c:439]: fm_malloc(): fm_malloc(0x7f4e36140010, 48) called from db_cassandra: dbcassa_base.cpp: db_cassa_new_connection(155) 0(12780) DEBUG: <core> [mem/f_malloc.c:514]: fm_malloc(): fm_malloc(0x7f4e36140010, 48) returns address 0x7f4e361bb5f0 0(12780) DEBUG: db_cassandra [dbcassa_base.cpp:161]: db_cassa_new_connection(): 0x7f4e361bb5f0=pkg_malloc(48) 0(12780) DEBUG: db_cassandra [dbcassa_base.cpp:117]: dbcassa_open(): Opened connection to Cassandra cluster 127.0.0.1:9160 0(12780) DEBUG: db_cassandra [dbcassa_base.cpp:955]: db_cassa_query(): query table=version 0(12780) ERROR: db_cassandra [dbcassa_base.cpp:449]: *cassa_translate_query(): table version does not exist!* 0(12780) ERROR: db_cassandra [dbcassa_base.cpp:962]: db_cassa_query(): Failed to query Cassandra cluster 0(12780) ERROR: <core> [db.c:397]: db_table_version(): error in db_query 0(12780) ERROR: <core> [db.c:436]: db_check_table_version(): querying version for table userblacklist 0(12780) ERROR: userblacklist [db_userblacklist.c:86]: userblacklist_db_init(): during table version check. 0(12780) DEBUG: <core> [db_pool.c:100]: pool_remove(): removing connection from the pool 0(12780) DEBUG: <core> [mem/f_malloc.c:575]: fm_free(): fm_free(0x7f4e36140010, 0x7f4e361bb5f0), called from db_cassandra: dbcassa_base.cpp: db_cassa_free_connection(203) 0(12780) DEBUG: <core> [mem/f_malloc.c:594]: fm_free(): fm_free: freeing block alloc'ed from db_cassandra: dbcassa_base.cpp: db_cassa_new_connection(155) 0(12780) DEBUG: <core> [mem/f_malloc.c:575]: fm_free(): fm_free(0x7f4e36140010, 0x7f4e361bb318), called from core: db.c: db_do_close(356) 0(12780) DEBUG: <core> [mem/f_malloc.c:594]: fm_free(): fm_free: freeing block alloc'ed from core: db.c: db_do_init2(298) 0(12780) ERROR: <core> [sr_module.c:968]: init_mod(): Error while initializing module userblacklist (/usr/local/lib64/kamailio/modules/userblacklist.so) ERROR: error while initializing modules Any help would be highly appreciated. Regards, Sammy

2 2

route_if_exists crash / breakage
by Alex Balashov 12 May '16

12 May '16

Hi, I'm running 4.4.0:ea2033 and am seeing a crash when using route_if_exists() on a nonexistent route: Core was generated by `/usr/local/sbin/kamailio -P /var/run/kamailio.pid -m 512 -M 8 -u evaristesys -g'. Program terminated with signal 11, Segmentation fault. #0 0x0000000000456751 in do_action (h=0x7ffcce5bc4c0, a=0xf0f0f0f0, msg=0x7fba440abc80) at action.c:337 337 switch ((unsigned char)a->type){ gdb) print a $1 = (struct action *) 0xf0f0f0f0 (gdb) print a->type Cannot access memory at address 0xf0f0f108 The invocation looks like this: if(check_route_exists("AUX_RTPENGINE_OFFER_INBOUND")) route_if_exists("AUX_RTPENGINE_OFFER_INBOUND"); else { set_rtpengine_set("1"); rtpengine_offer("replace-origin replace-session-connection ICE=remove"); } #route[AUX_RTPENGINE_OFFER_INBOUND] { # xlog("L_INFO", 'action=R-AUX-RTPENGINE-OFFER-INBOUND'); #} Uncommenting the nonexistent route has another curious effect: script execution appears to break after route[AUX_RTPENGINE_OFFER_INBOUND] executes, and nothing further happens. Changing route_if_exists("AUX_RTPENGINE_OFFER_INBOUND"); to route(AUX_RTPENGINE_OFFER_INBOUND); resolves both issues (the crash and the script execution halting). But, of course, if the route is commented out - its normal state - then Kamailio won't start since it won't allow a nonexistent identifier to be passed to route(). -- Alex Balashov | Principal | Evariste Systems LLC 1447 Peachtree Street NE, Suite 700 Atlanta, GA 30309 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 6

Kamailio - Manage several kamailio instances with Siremis
by José Seabra 12 May '16

12 May '16

Hi list, Anyone here knows if i can use one siremis instance to manage more than one Kamailio Instance? BR -- Cumprimentos José Seabra

1 0

Kamailio / issue with registration
by Ali Taher 12 May '16

12 May '16

Hello, I installed Homer with kamailio as capture agent in order to capture sip packets. The first test I did was using the default kamailio.cfg config located in homer-docker directory , where all Register and Invite packets were saved in the database and I was able to monitor them from web homer. After that , I tried to enable registration using the save(location) function in registrar module , yet I'm getting the below error in the log : 2016-05-11T18:48:36.443231+03:00 ubuntu /usr/sbin/kamailio[4759]: ERROR: <core> [udp_server.c:550]: udp_send(): sendto(sock,0x7f64702b1d98,767,0,192.168.26.3:3806,16): Bad file descriptor(9) 2016-05-11T18:48:36.443239+03:00 ubuntu /usr/sbin/kamailio[4759]: ERROR: sl [../../forward.h:201]: msg_send_buffer(): udp_send failed 2016-05-11T18:48:36.443247+03:00 ubuntu /usr/sbin/kamailio[4759]: ERROR: registrar [reply.c:734]: reg_send_reply(): failed to send 200 OK Can you please help with resolving this issue , noting that the registration works good if I remove the routing logic set for sipcapture. Any hint would be appreciated. Cheers, Ali Taher

1 1

Wiki update notice
by Mikko Lehto 12 May '16

12 May '16

Hello Some wiki work: 1. "What's new in 4.4" -content is no more present in devel version wiki page 2. based on recent mailing list discussion, I added note about github pull requests I think these make sense, please yell if not :) [1] http://kamailio.org/wiki/features/new-in-devel [2] http://kamailio.org/wiki/devel/github-contributions?rev=1456916916&do=diff -- Mikko

2 2

T_RELAY_TO distribute local proxy to a central one
by Sébastien Brice 12 May '16

12 May '16

Hi guys, i have a kamailio running on the wan (my central kamailio) and i have phones on a private network range 172.20.0.0/16 behind a NAT. i also run another kamailio locally to collect all sip phones on my network and this one use the DB of the remote central kamailio. the local kamailio does t_relay_to() the ip of the central kamailio route { if (uri==myself) { if (method=="REGISTER") { route(2); exit; }; route[2] { rewritehost("public_remote_kamailio"); if(!t_relay_to("udp:public_remote_kamailio:5060")) { sl_reply_error(); }; exit; } the bad thing is once the phones get registered with non-routable ip (like in the attached picture) trying to INVITE them directly by requesting public_kamailio_ip result in non-working situation. how should I rewrite my dst_domain when registring in a way INVITE triggers to the local kamailio. My local kamailio runs with the listen udp:private_local_kamailio advertise udp:public_local_kamailio I try to be explicit but my understanding is not perfect, it would be nice you could give me a hand. I put my local kamailio.cfg attached. thx

2 1

4.4 compiler warnings on debian jessie
by Juha Heinanen 12 May '16

12 May '16

debian jessie c compiler complains about these in kamailio 4.4: CC (gcc) [sip-proxy] mem/tlsf_malloc.o mem/tlsf_malloc.c: In function 'tlsf_malloc_init_pkg_manager': mem/tlsf_malloc.c:1353:16: warning: assignment from incompatible pointer type ma.xmalloc = tlsf_malloc; ^ mem/tlsf_malloc.c:1355:16: warning: assignment from incompatible pointer type ma.xrealloc = tlsf_realloc; ^ mem/tlsf_malloc.c:1358:16: warning: assignment from incompatible pointer type ma.xavailable = tlsf_available; ^ mem/tlsf_malloc.c: In function 'tlsf_malloc_init_shm_manager': mem/tlsf_malloc.c:1501:20: warning: assignment from incompatible pointer type ma.xmalloc_unsafe = tlsf_malloc; ^ it would be highly appreciated to get rid of them. -- juha

5 12

calling rpc command in cfg
by Yasin CANER 12 May '16

12 May '16

3 2

speeddial, owners
by Filippo Graziola 11 May '16

11 May '16

Hello everybody, I have a kamailio running on debian, and it works well with a little modification of the kamailio-advanced file. I would like to have some speed dial short numbers for all local users, because I used a random generator script and usernames and password are 25 digits (not only numbers) long. My problem is with the ownership of a every speeddial number. Example: SIP accounts: 5001,5002,5003 SIP domain: mydomain.com add a speeddial number: kamctl speeddial add 5002(a)mydomain.com 51(a)mydomain.com sip:5001@mydomain.com I can call from user 5002 to user 5001 typing 51 correctly. what I would like to do is that sd-id 51(a)mydomain.com could be called from every user registered at mydomain.com. I read the module documentation, kamctl help but I'm not able to find an answer. Could you please help me? Thank you in advance

2 2

kamailio does not see rtpengine module
by Dmitry 11 May '16

11 May '16

Hello when I compiled rtpengine for kamailio (in Kazoo)? there were no errors while compiling, but when restarting kamailio - I saw:kamailio: ERROR: <core> [modparam.c:150]: set_mod_param_regex(): No module matching <rtpengine> found [root@ip-192-168-130-136 kamailio]# locate rtpengine.so/usr/lib64/kamailio/modules/rtpengine.so default.cfg:mpath="/usr/lib64/kamailio/modules/" Does it mean that something wrong with rtpengine compilation? (I compiled manually) Appreciate any help.

1 0

Skype-like service, RTP flows strangely
by Sébastien Brice 11 May '16

11 May '16

Hi guys i did the "running skype-like service in less than one hour" and got it running ok. Kamailio run remotely in a datacenter. but i am trying to get the communication working locally between two devices behind the same NAT. I know the #!define WITH_NAT shoud get the call working bidirectionnaly but that is not the case, I've got one way dial every time i'm attempting call ! running a network capture (please find attached to this email )reveal rtp does not flow correctly: the device A 192.168.3.33 send RTP straight to device B 192.168.3.98 ---> OK device B does not behave like that, rather to talk to A (192.168.3.33) it sends RTprotocol to the public_ip of my internet box (refered as public_ip red text in my .png capture) and the WAN leg of my box answers ICMP 242 Destination unreachable (port unreachable) So comes blind communicatios. But that is not supposely act like that, right ? RTP should be relayed by the rtpproxy running on the remote Kamailio and we should have: device A -----> remote_kamailio_ip ----> device B device B -----> remote_kamailio_ip ----> device A thx you for any help PS: see attached file

2 1

VersionUp for CentOS7's
by 中島宗治 11 May '16

11 May '16

Please version up for CentOS7's rpm. -- Muneharu Nakasima <mune(a)vasara.net> <vasaralamp(a)gmail.com>

2 1

Kamailio and Oracle database
by gmele 11 May '16

11 May '16

Hello, we are trying to connect our Kamailio SIP to an Oracle database. We tried both db_oracle and db_unixodbc modules to connect to the DB. With db_oracle, Kamailio core dumps when trying to access the active_watchers table (caused by the oracle CLOB?). Here is the kamailio logs: Apr 29 13:54:53 newvm1 /usr/sbin/kamailio[23665]: INFO: <core> [db_query.c:131]: db_do_query_internal(): DB-QUERY: select presentity_uri,callid,to_tag ,from_tag,event from active_watchers where updated=:1 AND event=:2 Apr 29 13:54:53 newvm1 /usr/sbin/kamailio[23665]: INFO: db_oracle [res.c:238]: get_columns(): COLUMN PRESENTITY_URI TYPE 4 SIZE 129 Apr 29 13:54:53 newvm1 /usr/sbin/kamailio[23665]: INFO: db_oracle [res.c:238]: get_columns(): COLUMN CALLID TYPE 4 SIZE 256 Apr 29 13:54:53 newvm1 /usr/sbin/kamailio[23665]: INFO: db_oracle [res.c:238]: get_columns(): COLUMN TO_TAG TYPE 4 SIZE 65 Apr 29 13:54:53 newvm1 /usr/sbin/kamailio[23665]: INFO: db_oracle [res.c:238]: get_columns(): COLUMN FROM_TAG TYPE 4 SIZE 65 Apr 29 13:54:53 newvm1 /usr/sbin/kamailio[23665]: INFO: db_oracle [res.c:238]: get_columns(): COLUMN EVENT TYPE 4 SIZE 65 Apr 29 13:54:53 newvm1 /usr/sbin/kamailio[23665]: INFO: db_oracle [res.c:378]: get_rows(): OCIStmtFetch2() returned 100 Apr 29 13:54:53 newvm1 /usr/sbin/kamailio[23667]: CRITICAL: <core> [pass_fd.c:275]: receive_fd(): EOF on 16 Apr 29 13:54:53 newvm1 /usr/sbin/kamailio[23667]: CRITICAL: <core> [pass_fd.c:275]: receive_fd(): EOF on 14 What is really strange is that this table is accessed several times before the crash, without any problem. The other point is that this table is absolutely empty... With db_unixodbc, all requests having dates check fail because because the date format is not understood by oracle. The date should be converted using the oracle to_date(date, format) funtion to work... Here are the Kamilio traces: Apr 29 14:10:38 newvm1 /usr/sbin/kamailio[17021]: ERROR: usrloc [dlist.c:183]: get_all_db_ucontacts(): query error Apr 29 14:10:38 newvm1 /usr/sbin/kamailio[17021]: ERROR: nathelper [nathelper.c:2089]: nh_timer(): failed to fetch contacts Apr 29 14:10:39 newvm1 /usr/sbin/kamailio[17021]: ERROR: db_unixodbc [dbase.c:133]: db_unixodbc_submit_query(): rv=-1. Query= select received,contact,socket,cflags,path,ruid from$ Apr 29 14:10:39 newvm1 /usr/sbin/kamailio[17021]: ERROR: db_unixodbc [connection.c:220]: db_unixodbc_extract_error(): unixodbc:SQLExecDirect=HY000:1:933:[Oracle][ODBC][Ora]ORA-00$ Apr 29 14:10:39 newvm1 /usr/sbin/kamailio[17021]: ERROR: <core> [db_query.c:132]: db_do_query_internal(): error while submitting query I tried the option modparam("db_unixodbc", "use_escape_common", 1), but without success. Has anyone been able to make Kamailio 4.4.0 work with Oracle 11g, either with db_oracle or with db_unixodbc module? Regards Giovanni -- View this message in context: http://sip-router.1086192.n5.nabble.com/Kamailio-and-Oracle-database-tp1479… Sent from the Users mailing list archive at Nabble.com.

3 28

Garbage when decoding uri
by Glenn O Larsen 11 May '16

11 May '16

During a re-invite, it looks I get a strange decode error if the Route header is to long, or is there something else I have missed... 23(23945) DEBUG: rr [rr_cb.c:95]: run_rr_callbacks(): callback id 0 entered with <transport=tls;r2=on;lr=on;ftag=as03d8c4db;vsf=AAAAAEFVRVNSRxtbXFhFITUbBwcLEQBaQAFVDkVFUXI9cGhvbmU-> 23(23945) DEBUG: uac [replace.c:527]: restore_uri(): getting 'vsf' Route param 23(23945) DEBUG: uac [replace.c:535]: restore_uri(): route param is 'AAAAAEFVRVNSRxtbXFhFITUbBwcLEQBaQAFVDkVFUXI9cGhvbmU-' (len=52) 23(23945) DEBUG: uac [replace.c:609]: restore_uri(): decoded uris are: new=[sip:xxxxxxxxxx@u*5?%"9tu3{6dB @hone] old=[sip:xxxx.xxxx@10.0.0.1:5060] Kamailio 4.3.4 is used. -- Best regards Glenn Larsen

1 0

Kamailio Simple Config File Request
by Ali Dakroub~Vanrise Technical Support 10 May '16

10 May '16

Hello, I am a newbie in Kamailio configuration. I would be very grateful if you can provide me with a simple Kamailio.cfg file that achieves the below scenario: 1. Machines A and B are able to register on Kamailio Sip Server. 2. Machine A is able to call Machine B through Kamailio SIP server using x-lite. Noting that Machines A and B are on the same network. Thanks in advance. Regards, Ali

1 0

Setting source interface on relay
by Phil Lavin 10 May '16

10 May '16

Kamailio receives a fairly arbitrary out of dialog NOTIFY on an interface with a private IP address (eth1). It runs it through the following code which relays it to an IP/port combo pulled from a hash table. The IP that it is relaying to is a public address and should be addressed out of the interface with a public IP address (eth0). It is, however, sent on the private interface (eth1) and thus never reaches its target. route[PHONEREBOOT] { # Only handle check-sync (Polycom) NOTIFY if (!is_method("NOTIFY") || $hdr(Event)!="check-sync") return; # Look up ip:port combo in the agentloc hash table if ($sht(agentloc=>$rU)==$null) return; # Set SIP URI to point to the agent $ru = "sip:" + $rU + "@" + $sht(agentloc=>$rU); route(RELAY); exit; } Is there a way to explicitly set the source interface to relay on? Thanks Phil Lavin Telecoms Systems Manager CloudCall by SYNETY www.cloudcall.com<http://www.cloudcall.com/> T: +44 (0) 330 335 0000 / +1 617 982 1600 D: +44 (0) 116 424 4790 / +1 716 242 6604 SM: LinkedIn<https://uk.linkedin.com/pub/phil-lavin/25/422/750> READ OUR BLOG FOR SMARTER COMMUNICATIONS<http://t.sidekickopen03.com/e1t/c/5/f18dQhb0S7lC8dDMPbW2n0x6l2B9nMJW7t5XX43…> Confidentiality: This e-mail transmission, including any attachments, is intended only for the named recipient(s) and may contain information that is privileged, confidential and/or exempt from disclosure under applicable law. If you have received this transmission in error, or are not the named recipient(s), please notify the sender immediately by return e-mail and permanently delete this transmission, including any attachments. Security: This e-mail and any attachments are believed to be free from any virus but it is the responsibility of the recipient to ensure this is so. E-mail is not a 100% secure communication.

3 4

Kamailio v4.4.1 Released
by Daniel-Constantin Mierla 10 May '16

10 May '16

Hello, Kamailio SIP Server v4.4.1 stable release is out. This is a maintenance release of the latest stable branch, 4.4, that includes fixes since release of v4.4.1. There is no change to database schema or configuration language structure that you have to do on previous installations of v4.4.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.4.1. For more details about version 4.4.1 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2016/05/kamailio-v4-4-1-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Looking forward to meeting many of them at Kamailio World Conference 2016 next week in Berlin! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

extension of auth module
by JBF 10 May '16

10 May '16

Hello, We would like to use some functionalities provided in ims_auth. But our deployment uses auth/auth_db (we have another server with credentials and we would like to avoid adding a diameter interface to it) In particular, we want to add an Authentication-Info header to the response Do we have to patch the auth module to achieve this, or is there some other way ? If we patch it, would you merge it to the master ? its important to us, so we can easily upgrade to the next versions of kamailio. -- View this message in context: http://sip-router.1086192.n5.nabble.com/extension-of-auth-module-tp148332.h… Sent from the Users mailing list archive at Nabble.com.

1 0

Planning the release of Kamailio v4.4.1
by Daniel-Constantin Mierla 10 May '16

10 May '16

Hello, I am considering to do a minor release from branch 4.4 next week, likely on Tue or Wed, May 10 or 11, 2016. If there are issues you are aware of and not reported yet to the tracker, do it as soon as possible to try to fix what we can by the date. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com

1 2

WSS configuration sample needed
by Sunil More 09 May '16

09 May '16

Hello All, I am trying to connect Kamailio over WSS and the error I get is "tls_accept: client did not present a certificate" . It works fine for WS, however Please guide me to connect over WSS. the following are the logs i See.. DEBUG: tls [tls_domain.c:703]: sr_ssl_ctx_info_callback(): SSL handshake started DEBUG: <core> [tcp_main.c:2430]: tcpconn_do_send(): sending... ay 9 11:07:01 P172 /usr/local/kamailio/sbin/kamailio[32025]: DEBUG: tls [tls_domain.c:715]: sr_ssl_ctx_info_callback(): SSL handshake done May 9 11:07:01 P172 /usr/local/kamailio/sbin/kamailio[32025]: DEBUG: tls [tls_domain.c:718]: sr_ssl_ctx_info_callback(): SSL disable renegotiation May 9 11:07:01 P172 /usr/local/kamailio/sbin/kamailio[32025]: DEBUG: tls [tls_server.c:411]: tls_accept(): TLS accept successful May 9 11:07:01 P172 /usr/local/kamailio/sbin/kamailio[32025]: DEBUG: tls [tls_server.c:418]: tls_accept(): tls_accept: new connection from sunilmore.in:49703 using TLSv1/SSLv3 AES256-SHA 256 May 9 11:07:01 P172 /usr/local/kamailio/sbin/kamailio[32025]: DEBUG: tls [tls_server.c:421]: tls_accept(): tls_accept: local socket: sunilmore.in:10082 May 9 11:07:01 P172 /usr/local/kamailio/sbin/kamailio[32025]: DEBUG: tls [tls_server.c:432]: tls_accept(): tls_accept: client did not present a certificate And the websocket closes. Please help. Thanking You, Sunil More Ph : 9503338275

2 5

sdp_remove_line_by_prefix
by Alberto Sagredo 09 May '16

09 May '16

Hi Im trying to remove my Asterisk s= line on SDP but when doing: sdp_remove_line_by_prefix("s=Asterisk"); Nothing happens. Loaded sdpops.so module as well. Is there any additional needed? BR

4 10

Getting sip.instance in INVITE
by Cibin Paul 09 May '16

09 May '16

Hi, How can I get sip.instance value in the sip header of INVITE? I do get this value during REGISTER Thanks & Regards Cibin

2 2

Third party registration help
by Code Developer 09 May '16

09 May '16

Hello, I'm after some help on third-party SIP registration in Kamailio - specifically, how to perform registration of user agents that don't register themselves, but only send out occasional messages. The UAs not registering themselves are devices on a GSM network that try to save band-width by not sending REGISTER requests for themselves. They only ever send out SIP messages when necessary, to which they expect a reply message from our processing server. (Note, details like external IP addresses have been changed for security.) The configuration is as follows. We have a SIP server running Kamailio (4.3, location db enabled, nathelper enabled) on a static external IP address (50.170.158.149), a processing server on an internal network registered with the SIP server (let's say with a User agent name of '1000'), and a monitoring device in another part of the country connected to the internet via a GSM network (let's say User agent name of '2000'). The device is effectively NAT'd on the GSM provider's private network. At some point the device (UA 2000) sends a SIP message to the processing server via Kamailio (to sip:1000@50.170.158.149). The processing server is registered, so the SIP server knows where it is and forwards the message accordingly. 200/OK replies propagate back successfully to the device. The message flow appear as below. Only relevant header fields are shown: Message received by the SIP server from remote device: ======== IP Src:60.250.149.198, Dst:50.170.158.149 UDP Src Port: 59177, Dst Port: 5060 -- Request-Line: MESSAGE sip:1000@50.170.158.149 Via: SIP/2.0/UDP 10.92.240.131:5060;branch=z9hG4bK... To: <sip:1000@50.170.158.149> From: <sip:2000@10.92.240.131>;tag=7a82c48a ======== Message forwarded from SIP server, received by processing server as shown: ======== IP Src:50.170.158.149, Dst: 192.168.0.235 UDP Src Port:5060, Dst Port:5060 -- Request-Line: MESSAGE sip:1000@50.170.158.146:5060; ob SIP/2.0 Via: SIP/2.0/UDP 50.170.158.149;branch=z9hG... Via: SIP/2.0/UDP 10.92.240.131:5060;rport=59177;received=60.250.149.198;branch=z9hG... To: <sip:1000@50.170.158.149> From: <sip:2000@10.92.240.131>;tag=7a82c48a ======== The processing server now needs to send back a reply message to the remote, unregistered device which is NATd on the GSM internal network. We can do this by sending back a reply to the external IP address and rport values found in the second Via header of the message received by the processing server (60.250.149.198, port 59177). The reply can only be sent from the SIP server or the GSM network's router will ignore it - it's only a temporary reply address/port open for a minute or so as it's effectively just a NAT route to the internet. In the first instance, when the processing server attempts to send back a message to the remote device via the SIP server, the SIP server replies with a 404 as the remote device hasn't registered and the SIP server has no idea where to send it. After looking at RFC3261 my conclusion was that we must get the processing server to perform a third-party registration on behalf of the remote device before we can send a reply message back to that device. My understanding is that to perform the third party registration I should send a REGISTER request from our processing server to the SIP server: 1) Set the 'From' field to the AOR of the UA performing the registration (which is the processing server, 1000). 2) Set the 'To' field to the AOR of the UA being registered (i.e. the remote device, 2000). 3) Add a 'Contact' header field set to the IP address and port of the remote device (from Via = 60.250.149.198 port 59177). The message flow for the register is as follows (only relevant header fields shown): Processing server sends a register request to the SIP Server: ======== Request-Line: REGISTER sip:50.170.158.149 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.235:5060;rport;branch=z9hG... From: <sip:1000@50.170.158.149>;tag=e6e2d70... To: <sip:2000@50.170.158.149> Contact: <sip:2000@60.250.149.198:59177> ======== SIP server responds to the processing server with a 401 unauthorized response containing the auth header field (nonce challenge): ======== WWW-Authenticate: Digest realm="50.170.158.149", nonce="VytkJ1crIvu4ekGp/M4mnRlYmJRciF3N" ======== Processing server sends a revised register request to the SIP Server, augmented with the authorisation header for the supplied nonce: ======== Request-Line: REGISTER sip:50.170.158.149 SIP/2.0 Via: SIP/2.0/UDP 50.170.158.146:1096; rport;branch=z9hG4... From: <sip:1000@50.170.158.149>;tag=dae68a... To: <sip:2000@50.170.158.149> Contact: <sip:2000@60.250.149.198:59177> Authorization: Digest username="1000", realm="50.170.158.149", nonce="VytkJ1crIvu4ekGp/M4mnRlYmJRciF3N" response="d4d7ffba8..." ======== However, the SIP Server again responds with a 401 and won't perform the third-party registration. I haven't been able to determine why. I *think* the response hash the processing server is sending back is correct. I'm using PJSIP on our processing server to perform all SIP functionality. The processing server does obviously register itself, the difference being that the To and From fields are the same (<sip:1000@50.170.158.149>) and there is no Contact header. I'm at the point now of being unable to find any information or examples on how third-party registration requests should be performed - apart from RFC3261, which doesn't really describe the process very well. Can anyone tell me why the third-party registration would fail, despite replying with the authorisation header? Does Kamailio first need to be configured to allow third-party reg somehow? Alternatively, should I be attempting to send back SIP messages to unregistered devices in some other fashion? I hope someone can help or point me in the right direction. As an aside, when the device (UA2000) is plugged into a network via ethernet, it does then perform its own registration and everything works - messages go back and forth without issue.

2 1

LCR Module Question
by Alberto Sagredo 09 May '16

09 May '16

Hi I have it working but i have re-read documentation and do not see how to do what i need. I explain it :) Now i have only one LCR provider and i need to add a backup one. I do not know if its enough to add under same lcr_id or its better to add with different one and add several lcr_rule and lcr_rule_target Acordding to next_gw() function, what is better to be used? In my case. a primary providers is used and only a backup one for the momment. Thanks for your help Alberto

3 3

Media-Proxy & NAT
by cao tao 09 May '16

09 May '16

Hello: I use kamailio as my SIP signal proxy, and media-proxy as RTP relayer. It works well at many situations. But when it comes to two layer of NAT(a user behind two layer of NAT), I got a single way media(the sip signal layer was ok). I can’t hear the voice of user who behinds two layer of NAT, but he can hear the voice of mine. Most part of my kamailio config is default, and simply enable media-proxy by engage_media_proxy()： if (is_method("INVITE") && !has_totag()) { engage_media_proxy(); } And in my UAC/UAS side, I use PJSIP. Can anybody give me some help? Any suggestion will be appreciated!

2 1

User offline
by 王欣鑫 06 May '16

06 May '16

Hello, I use kamailio set up an IMS on a cloud server in a public network, and have alice and bob registered on the server using ImsDroid in a private network. When I use alice to call bob, I got "User offline" error. Below are .pcap file captured from 4060 port and pcscf.log/icscf.log/scscf.log and all config files(in kamailio.zip) 4060.pcap <http://sip-router.1086192.n5.nabble.com/file/n148166/4060.pcap> pcscf.log <http://sip-router.1086192.n5.nabble.com/file/n148166/pcscf.log> icscf.log <http://sip-router.1086192.n5.nabble.com/file/n148166/icscf.log> scscf.log <http://sip-router.1086192.n5.nabble.com/file/n148166/scscf.log> kamailio.zip <http://sip-router.1086192.n5.nabble.com/file/n148166/kamailio.zip> My kamailio version and debian operating system version is like below: root@iZ94lg22es7Z:~# kamailio -version version: kamailio 4.4.0 (x86_64/linux) d4f23c flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: d4f23c compiled on 13:02:46 May 5 2016 with gcc 4.9.2 root@iZ94lg22es7Z:~# root@iZ94lg22es7Z:~# cat /etc/issue Debian GNU/Linux 8 \n \l root@iZ94lg22es7Z:~# Can anybody give me some help? Any suggestion will be appreciated! -- View this message in context: http://sip-router.1086192.n5.nabble.com/User-offline-tp148166.html Sent from the Users mailing list archive at Nabble.com.

2 14

Clients are busy
by kai.ohnacker＠cbc.de 06 May '16

06 May '16

Hello Community, in my scenario works multiple user with the kamailio server and uses different clients. They work in a mobile scenario and setup the clients and break the clients in a high period. If they break the clients, sometimes they don't send a bye, but they take off the cable connection during a call. If these devices will connected again and other clients try to call them, they are busy. That is my problem :). Does the Kamailio server provides any function to send a bye message if a clients don't register at a required time? Or is there another solution available? Thanks for help. Cheers, Kai

2 1

making rtpengine gives error on centos
by Yasin CANER 05 May '16

05 May '16

4 5

Kamailio - Sip_capture Module
by Ali Taher 05 May '16

05 May '16

Hello, We are currently using kamailio sip_capture module, and then we need to create a Wireshark file (pcap format)from fields in the database. As we noticed in sip_capture table there is msg fields that contains only the sip content. And in order to create our file we need to have the entire packet and not part of it. We are hopefully asking if you can give us a hand with that , beside the Homer project, what we actually wanted is to know if there is another module in kamailio that store the entire packet captured and not only the part of it. Regards, Ali Taher

2 3

Pause a Transaction for a short while
by Sirvan Paraste 05 May '16

05 May '16

Hello all, is there anyway to pause a transaction for a short while (1 second) ? I have implemented the push notification, when the callee get register I need a short time to be sure that the app is completely start and can do its duty. otherwise it missed the call. so I want to pause the transaction. I have tested async_sleep("1"); it worked. is there any better solution ? $var(hjoin) = 0; lock("$tU"); async_sleep("1");

2 1

ACK not forwarded by Proxy when LOCATION lookup
by Al S 05 May '16

05 May '16

Hi, I am trying to simulate Inbound calls from UAC to Kamailio to UAS. Kamailio performs location lookup within route[LOCATION] and updates URI with the address of UAS. However, it does not add a new Record-Route. Therefore, ACK message sent by UAC is not routed by the proxy: UAC ---> (send INVITE) ---> Kamailio ---> (send INVITE) ---> UASUAS ---> (send 180) ---> Kamailio ---> (send 180) ---> UACUAS ---> (send 200) ---> Kamailio ---> (send 200) ---> UAC UAC ---> (send ACK) ---> Kamailio ???? ACK message has "Route" header with the address of Proxy. Thanks,AI S

2 1

You must register first with a S-CSCF/Server error on LIR select next S-CSCF
by 王欣鑫 05 May '16

05 May '16

Hello, I'm using kamailio and OpenImsCore HSS module to set up an IMS enviroment(P/I/S-CSCF and HSS). All these four modules are set up in one cloud server in public network with different ports(4060/4061 5060 6060 3869 3870) to commonicate with each other. Now I have two users Alice and Bob, they are in private network. I can register Alice and Bob with ImsDroid on this IMS, however when I use Alice to call Bob, I can see "You must register first with a S-CSCF" error. As I found this was caused by kamailio.cfg in pcscf module, I comment out these lines below. Then I got another error "Server error on LIR select next S-CSCF". And I captured packets on port 5060, I can see several "Request-Line: OPTIONS sip:sbc-2.ng-voice.com SIP/2.0" and "Request-Line: OPTIONS sip:sbc-1.ng-voice.com SIP/2.0". #if (!pcscf_is_registered("location")) { # send_reply("403","Forbidden - You must register first with a S-CSCF"); # break; #} What's wrong with my IMS? Could any body give me some help? Thank you! Here are my IMS configure files. kamailio.zip <http://sip-router.1086192.n5.nabble.com/file/n148131/kamailio.zip> Here are my hss configure files. hss_config.zip <http://sip-router.1086192.n5.nabble.com/file/n148131/hss_config.zip> -- View this message in context: http://sip-router.1086192.n5.nabble.com/You-must-register-first-with-a-S-CS… Sent from the Users mailing list archive at Nabble.com.

1 0

Using websocket server for events
by Yuriy Gorlichenko 04 May '16

04 May '16

Hi is there is a valy to send websocket events (not only SIP) on remote subscribed client from config file? I just want make shure that i need to use redis for this staff insead of native websocket events

2 3

Using Kamailio as UAC on any PBX
by Jöran Vinzens 04 May '16

04 May '16

Hi All, we are trying to use the Kamailio as an UAC on a PBX. It should behave just like a normal SIP UAC (e.g. an SNOM phone). The reason why we want that, is connecting a service to any other PBX with Client features and security (DA). Our setup looks like: Kamailio UAC w/ register <-> Kamailio OBP/SBC <-> PBX The register is working fine. We insert the credentials into DB and trigger kamailio to reload the config (kamctl uac.reload). Then the register is sent out including the unauth (401) handling. After than we are registered on the PBX. The Register is sent though the OBP as configured in DB. We are trying to make calls: Calls from PBX to Kamailio are working fine but we struggle with calls from kamailio torwards PBX. We have the l_uuid in a variable and we try to get the PV ready for the invite. First we need to do is rewrite the "from" URI. We thought we can get a proper FROM URI with the uuid we have. But the uac_replace_from just sets the $fu as given and does not make any logic with UAC credentials. To handle the RURI, we trigger uac_reg_lookup with the uuid and it replaces the $ru with user-name and domain from the data stored in DB. In this case that makes no sense. The $rU should be what the user had dialed, the $rd should be the server given in uacreg DB table. The OBP given in DB is ignored completely. After we managed to rewrite everything in SIP as needed we try to catch the 407 as it's shown in the module documentation with failure route. Here we have not found any way to get the credentials for the uuid. Now we wonder if we have missed any global setting which sets everything right. If anybody has a similar setup and a working config, we appreciate any help we can get. our config looks like: ... loadmodule "uac.so" .... modparam("uac","auth_realm_avp","$avp(arealm)") modparam("uac","auth_username_avp","$avp(auser)") modparam("uac","auth_password_avp","$avp(apass)") modparam("uac","restore_from_avp","$avp(original_uri_from)") modparam("uac","restore_to_avp","$avp(original_uri_to)") modparam("uac","credential","username:domain:password") modparam("uac", "reg_db_url","DBURI") modparam("uac", "reg_timer_interval", 60) modparam("uac", "reg_retry_interval", 300) modparam("uac", "reg_contact_addr", "1.2.3.4:5060") ... route { xlog("L_NOTICE", "INVITE: F=$fu T=$tu R=$ru\n"); route("QUERY_UUID"); # <- sets $var(UUID) to l_uuid of the DB $var(request_user) = $rU; uac_reg_lookup("$var(UUID)", "$ru"); $var(request_from_user) = $rU; $rU = $var(request_user); uac_replace_from("$var(request_from_user)","sip:$var(request_from_user)@ 1.2.3.4"); t_on_failure("TRUNKAUTH"); xlog("L_NOTICE", "INVITE: F=$fu T=$tu R=$ru $avp(auser) $avp(apass)\n"); # avp "auser and apass are "<null>" if (!t_relay()) { xlog("L_ERR", "error to <$tu> from <$fu>\n"); sl_reply_error(); }; } failure_route[TRUNKAUTH] { if (t_is_canceled()) { exit; } if(t_check_status("401|407")) { # $avp(auser) = "test"; <-should be set from the l_uuid profile anyhow # $avp(apass) = "test"; $var(request_user) = $rU; if (uac_auth()) { # We do never get here since uac_auth never returns true. $rU = $var(request_user); route("FORWARD_MESSAGE"); exit; } xlog("L_NOTICE", "DAMN!!!"); } } -- Jöran Vinzens - vinzens[at]sipgate.de <vinzens(a)sipgate.de> sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois Steuernummer: 106/5724/7147, Umsatzsteuer-ID: DE219349391 www.sipgate.de - www.sipgate.co.uk

2 1

rtpengine default set
by Alex Balashov 04 May '16

04 May '16

Hi, After upgrading to Kamailio v4.4, I receive the following error on boot: May 03 17:18:22 allegro-4.evaristesys.com /usr/local/sbin/kamailio[24972]: ERROR: rtpengine [rtpengine.c:2691]: select_rtpp_set(): script error-invalid id_set to be selected May 03 17:18:22 allegro-4.evaristesys.com /usr/local/sbin/kamailio[24972]: ERROR: rtpengine [rtpengine.c:2691]: select_rtpp_set(): script error-invalid id_set to be selected May 03 17:18:22 allegro-4.evaristesys.com /usr/local/sbin/kamailio[24972]: NOTICE: rtpengine [rtpengine.c:1788]: mod_init(): Default rtpp set 0 NOT found My rtpengine module config is: --- #!ifdef WITH_RTPENGINE #!subst "/RTPENGINE_SOCK/udp:127.0.0.1:5050/" modparam("rtpengine", "rtpengine_sock", "1 == RTPENGINE_SOCK") modparam("rtpengine", "rtpengine_disable_tout", 30) modparam("rtpengine", "rtpengine_tout_ms", 2000) modparam("rtpengine", "rtpengine_retr", 2) #!endif --- Furthermore, I have set_rtpengine_set("1") before every call to the rtpengine_* module function(s) in my route script. Subsequently to that, Kamailio starts normally and connects to rtpengine: May 03 17:18:22 allegro-4.evaristesys.com /usr/local/sbin/kamailio[24972]: INFO: rtpengine [rtpengine.c:2542]: rtpp_test(): rtp proxy <udp:127.0.0.1:5050> found, support for it enabled May 03 17:18:22 allegro-4.evaristesys.com /usr/local/sbin/kamailio[24975]: INFO: rtpengine [rtpengine.c:2542]: rtpp_test(): rtp proxy <udp:127.0.0.1:5050> found, support for it enabled May 03 17:18:22 allegro-4.evaristesys.com /usr/local/sbin/kamailio[24974]: INFO: rtpengine [rtpengine.c:2542]: rtpp_test(): rtp proxy <udp:127.0.0.1:5050> found, support for it enabled [...] It might be something on my end, but my guess is that in v4.4 there is a fixup that runs beforehand and looks for a default set. Strictly speaking, I don't need an explicitly identified set, since I'm only using one rtpengine instance. However, I explicitly specify it because I'd like the code to be open in the future to selecting a given rtpengine relay based on some criteria. -- Alex -- Alex Balashov | Principal | Evariste Systems LLC 1447 Peachtree Street NE, Suite 700 Atlanta, GA 30309 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 2

create packet
by zeinab 04 May '16

04 May '16

Hello, My name is Zeinab and I am using kamailio to capture sip packets. What I really want to do is to create a .pcap file so that I can open it in Wireshark. As I saw, that you are only saving the sip content in database. And to create the packet I need the entire frame. So I think there is something missing from my side. Can you help me with that? You can see in the attachment the field needed from wireshark And in sip_capture I found only this: If there is some manual were I can find the entire packets please let me know Thank you for your time and looking forward to hear from you. Best Regards, Zeinab

2 1

create PCAP file
by zeinab 04 May '16

04 May '16

Hello, My name is Zeinab and I am using kamailio to capture sip packets. What I really want to do is to create a .pcap file so that I can open it in Wireshark. As I saw, that you are only saving the sip content in database. And to create the packet I need the entire frame. So I think there is something missing from my side. Can you help me with that? You can see in the attachment the field needed from wireshark And in sip_capture I found only this: If there is some manual were I can find the entire packets please let me know Thank you for your time and looking forward to hear from you. Best Regards, Zeinab

1 0

Kamailio 5-0 - Executing event routes in embedded languages
by Daniel-Constantin Mierla 04 May '16

04 May '16

Hello, as announced earlier this morning, most of the routing blocks can be written directly in an embedded language (right now supported being Lua and Python). The missing blocks are (most of) the event routes. The event route from tm for branch failure route is already supported, as the name has to be set via t_on_branch_failure(...). In the native configuration, event routes are executed if they are defined. Their names (as in event_route[name]) are not valid function names for the other programming languages to be used directly corresponding callback function name. I was thinking of the alternative, next are the ones I came up with, more suggestions are welcome. 1) add parameters for event route callback names, like: modparam("htable", "evrtcb_mod_init", "ksr_htable_mod_init") would define the name of the function for the alternative to event_route[htable:mod-init]. This has a benefit in speed, if the parameter is not defined, then there is no overhead in trying to execute a callback function in an embedded language. The drawback is that there is need to code a bit of C to add those parameters to the components executing event routes. 2) do a direct mapping using some translation rules, like: - prefix with ksr_ - replace ':' and '-' with underscore '_' So event_route[htable:mod-init] will have automatically the correspondent function ksr_htable_mod_init(). The drawback is that each time the even route should be called, a lookup on embedded language symbols table needs to be done to see if it exists, then execute it. There is still some C code to do, but less than adding parameters like for 1). 3) like 2), but instead of an automatic translation, keep a mapping table: struct mapping { str kname; str ename; } table[] = { str_init("htable:mod-init"), str_init("ksr_htable_mod_init"), } Adding a new event route will require adding a new record in the mapping table. The benefit is that the function name in embedded language can be selected to be whatever, with or without ties to the event_route name. Opinions? Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com

1 0

A question about the code
by ngregory＠reprivata.com 04 May '16

04 May '16

3 4

Kamailio 5.0 - Routing blocks in Lua or Python
by Daniel-Constantin Mierla 04 May '16

04 May '16

Hello, I want to announce that the framework (named kemi) which allows writing the routing blocks in an embedded language is already in place. Lua or Python can be use as alternative to the native scripting language to write the routing blocks. More details can be found at: - https://www.kamailio.org/wiki/devel/config-engines Example of scripts that offer more or less same features as etc/kamailio-basic.cfg are available in source tree at examples/kemi. For convenience of reading, here are the links to git repository: - the common file setting global and modules parameters as well as specifying the config engine: https://github.com/kamailio/kamailio/blob/master/examples/kemi/kamailio-bas… - the part of config with routing blocks in native language: https://github.com/kamailio/kamailio/blob/master/examples/kemi/kamailio-bas… - the part of config with routing blocks in Lua: https://github.com/kamailio/kamailio/blob/master/examples/kemi/kamailio-bas… - the part of config with routing blocks in Python: https://github.com/kamailio/kamailio/blob/master/examples/kemi/kamailio-bas… You can compare the last three above and see how different or similar they look for providing same SIP routing features using different languages. A lot of modules need to export now functions to embedded config interface, which is quite straightforward: https://www.kamailio.org/wiki/devel/config-engines#exporting_functions_to_k… The main missing part is dealing with event_route blocks -- I will send a separate email to discuss that. Interesting that some incipient performance tests to compare the interpreters show good results for using especially Lua but also Python: - https://www.kamailio.org/wiki/devel/config-engines#interpreters_performances Anyone willing to help can start exporting functions from modules to embedded interface or test some more complex configs in Lua or Python. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com

1 0

t_on_branch_failure never called AND t_relay never fails in a, forked scenario
by Tristan Mahé 03 May '16

03 May '16

Hi fellow kamailio users, I'm playing actually with t_on_branch_failure, in a simultaneous forking scenario ( lookup("location") returns multiple contacts ). Any idea on how to get those "on_branch_failure" working ? Thanks in Advance ! * Test by putting failure_exec_mode: - at 1 => no branch failure being called, never. - at 0 => t_relay does not fail if a single branch answers the transaction, even if all other branches are failing. * Test by moving the t_on_branch_failure("manage_branch_failure"); line before or after the lookup("location") or on a _on_branch route to set it per branch. - The on_route branch are executed, but never the failure one. - tried also with failure_exec_mode = 0 or 1. * Using set_forward_no_connect, I can see in the logs a "Sending to branch X failed" but no failure route being executed. Without it, nothing in the logs indicating that the branch has timeout. * Config extract: # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # 3 by default # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) modparam("tm", "reparse_on_dns_failover", 0) # 1: Allow branches to have a per branch failure route # 0: t_relay should fail if a branch fail modparam("tm", "failure_exec_mode", 1) # Allow locally generated 408 to be forgotten in case a reply is received. #modparam("tm", "faked_reply_prio", 2000) # USER location service route[LOCATION] { t_on_branch_failure("manage_branch_failure"); if (!lookup("location")) { .... } # Let's get back after 2.5s if no 100/Trying received. t_set_fr(0, 2500); # set_forward_no_connect(); #t_on_branch("SET_BRANCH_FAILURE"); route(RELAY); exit; } route[RELAY] { .... if (!t_relay()) { xlog("L_WARN","$rm|$ci|$fu|$tu|RELAY_FAILED"); sl_reply_error(); } exit(); } # set failure route per branch branch_route[SET_BRANCH_FAILURE] { xlog("L_WARN","$rm|$ci|$fu|$tu|SETTING_BrANCH_FAILURE_ROUTE"); t_on_branch_failure("manage_branch_failure"); } # manage failure routing cases per branch event_route[tm:branch-failure:manage_branch_failure] { xlog("L_WARN","$rm|$ci|$fu|$tu[$avp(dest_to)]|MANAGE_BRANCH_FAILURE"); }

3 2

set_body_multipart and memory leak
by Sergey Basov 03 May '16

03 May '16

Hi, All. I have some problems with pkg_memory leak. After each call used private memory of the "udp receiver" process increases at 2500 bytes. On test environment I hav set it to 2Mbyte, but it is full after 700 calls. I need to modify SDP in invite and other packets. When I do ==================================================== if(has_body("application/sdp")) set_body_multipart(); if (msg_apply_changes()) { xlog("L_INFO", "ISUP 1 Changes Applied Succesfully"); } } record_route(); dlg_manage route(RELAY); ==================================================== Debug from kamailio: Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: textops [textops.c:2627]: has_body_f(): content type is 196611 Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: textops [textops.c:1577]: set_multibody_helper(): delimiter<17>:[unique-boundary-1] Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: textops [textops.c:1486]: generate_boundary(): adding final CRLF+CRLF Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: textops [textops.c:1714]: set_multibody_helper(): content-type<44>:[multipart/mixed;boundary="unique-boundary-1"] Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: textops [textops.c:1766]: set_multibody_helper(): set flag FL_BODY_MULTIPART Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: ERROR: <core> [msg_translator.c:1691]: get_boundary(): Content-Type hdr has no params Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: WARNING: <core> [msg_translator.c:1954]: build_req_buf_from_sip_req(): check_boundaries error Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: <core> [msg_translator.c:422]: clen_builder(): content-length: 251 (251) Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: textopsx [textopsx.c:231]: msg_apply_changes_f(): SIP message content updated - reparsing Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: <core> [parser/msg_parser.c:608]: parse_msg(): SIP Request: Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: <core> [parser/msg_parser.c:610]: parse_msg(): method: <INVITE> Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: <core> [parser/msg_parser.c:612]: parse_msg(): uri: <sip:0637304494@10.56.42.27:5070;user=phone> Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: <core> [parser/msg_parser.c:614]: parse_msg(): version: <SIP/2.0> Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: <core> [parser/parse_via.c:1254]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK09e6.2654d2382f7c4e4fb8bb62fe85f62dfa.0>; state=6 Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: <core> [parser/parse_via.c:1254]: parse_via_param(): Found param type 236, <i> = <1>; state=16 Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: <core> [parser/parse_via.c:2642]: parse_via(): end of header reached, state=5 Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: <core> [parser/msg_parser.c:496]: parse_headers(): parse_headers: Via found, flags=2 Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: <core> [parser/msg_parser.c:498]: parse_headers(): parse_headers: this is the first via Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: INFO: <script>: ISUP 1 Changes Applied Succesfully Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: <core> [parser/parse_via.c:1254]: parse_via_param(): Found param type 232, <branch> = <z9hG4bKsr-s7wTDLa0zUfsTql5zpl0zpl0z.lRD.zok.TLz.lSk2Fw1RY9PJYMxpdED.dED.aEG38LkpcsWhqMQr1v1bFAyJGCxrCoQdB71Mtceh52GpZKDpX3zRFAzVFJzhBLGpeJe.TczW7cDuT8GLs7O37vBbn6BbTfGhM2Gpl*>; state=16 Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: <core> [parser/parse_via.c:2642]: parse_via(): end of header reached, state=5 Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: <core> [parser/msg_parser.c:496]: parse_headers(): parse_headers: Via found, flags=ffffffffffffffff Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: <core> [parser/msg_parser.c:509]: parse_headers(): parse_headers: this is the second via Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: <core> [parser/parse_addr_spec.c:894]: parse_addr_spec(): end of header reached, state=10 Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: <core> [parser/msg_parser.c:173]: get_hdr_field(): DEBUG: get_hdr_field: <To> [46]; uri=[sip:0637304494@10.56.42.27:5070;user=phone] Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: <core> [parser/msg_parser.c:175]: get_hdr_field(): DEBUG: to body [<sip:0637304494@10.56.42.27:5070;user=phone>#015#012] Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: <core> [parser/msg_parser.c:153]: get_hdr_field(): get_hdr_field: cseq <CSeq>: <2> <INVITE> Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: <core> [parser/msg_parser.c:187]: get_hdr_field(): DEBUG: get_hdr_body : content_length=251 Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: <core> [parser/msg_parser.c:89]: get_hdr_field(): found end of header -- Best regards, Sergey Basov e-mail: sergey.v.basov(a)gmail.com

2 16

xmlrpc crash
by Juha Heinanen 03 May '16

03 May '16

i was sending some xmlrpc requests to kamailio and got crash. -- juha (gdb) where #0 0x00007f9c465da634 in __GI___libc_free (mem=0x17828c0) at malloc.c:2945 #1 0x00007f9c39e0b797 in collect_garbage () at xmlrpc.c:741 #2 0x00007f9c39e12669 in dispatch_rpc (msg=0x0, s1=0x17828b0 "ams", s2=0x7f9c46903628 <main_arena+8> "`\352w\001") at xmlrpc.c:2440 #3 0x000000000053c2c0 in do_action (h=0x7ffd17c562d0, a=0x7f9c46355c28, msg=0x7ffd17c563b0) at action.c:1060 #4 0x000000000053ad85 in run_actions (h=0x7f9c46903620 <main_arena>, a=0x17828b0, msg=0x7f9c46903628 <main_arena+8>) at action.c:1549 #5 0x00007f9c39e0d04b in em_receive_request (orig_msg=0x7f9c45cc43e8, new_buf=0x0, new_len=24651952) at xmlrpc.c:2303 #6 0x00007f9c39e14034 in process_xmlrpc (msg=0x7f9c45cc43e8) at xmlrpc.c:2385 #7 0x00000000004fdce3 in nonsip_msg_run_hooks (msg=0x7f9c45cc43e8) at nonsip_hooks.c:111 #8 0x0000000000549610 in receive_msg (buf=0x0, len=1171014632, rcv_info=0x7f9c3e4b0110) at receive.c:188 #9 0x00000000005cb97a in tcp_read_req (con=0x7f9c3e4b00f8, bytes_read=0x7ffd17c56e64, read_flags=0x7ffd17c56e68) at tcp_read.c:1418 #10 0x00000000005d0293 in handle_io (fm=0x7f9c3e4b00f8, events=10416, idx=-1) at tcp_read.c:1611 #11 0x00000000005d6bce in io_wait_loop_epoll (h=<optimized out>, t=<optimized out>, repeat=<optimized out>) at io_wait.h:1061 #12 tcp_receive_loop (unix_sock=1183856160) at tcp_read.c:1781 #13 0x00000000004d9df7 in tcp_init_children () at tcp_main.c:4788 #14 0x000000000050736f in main_loop () at main.c:1690 #15 0x000000000041cdbc in main (argc=0, argv=0x0) at main.c:2616

2 15

Kamailio Client-Server
by Shiv Patidar 02 May '16

02 May '16

hiii 1.We are writing our own SIP Client to test my Client i am using Kamailio server 2. So i am trying to map the identities used between Client and Server 3. If can possible can u please help me to the steps has to be follower to set identities common between Client and Server. 4. Or if u have some test code for client can u share the SIP message dump for test code so i can re-use the same for configuration my client code. Thank you

3 2

unixODBC failure/bug?
by Nagorny, Dimitry 02 May '16

02 May '16

Hi all, I am trying to use the unixODBC module to connect to MariaDB as a first step and then to test it to connect to MSSQL (please don't blame me on MSSQL...). I am using Kamailio 4.3.5 on CentOS7.2 fully patched. Here info about installed unixODBC: [root@r5dev-KAM1 ~]# yum list installed | grep unix unixODBC.x86_64 2.3.1-11.el7 @base unixODBC-devel.x86_64 2.3.1-11.el7 @base Which leads me to my possible problem that I can't start Kamailio with db_unixodbc.so because it is giving me this error: r5dev-KAM1 /usr/local/kamailio/sbin/kamailio[16307]: ERROR: <core> [db.c:203]: db_bind_mod(): Module db_unixODBC not found. Missing loadmodule? Followed by errors from modules depending on the DBURL are failing as well. Here my modules part: ... #!define DBURL "unixODBC://root:<passw>@localhost/MARIADB" ... # Module Path mpath="/usr/local/kamailio/lib64/kamailio/modules/" loadmodule "tls.so" # ----- tls params ----- modparam("tls", "config", "/usr/local/kamailio/etc/kamailio/tls.cfg") #loadmodule "db_mysql.so" loadmodule "db_unixodbc.so" ... I doublechecked that the module was compiled and is present: [root@r5dev-KAM1 ~]# ls -ll /usr/local/kamailio/lib64/kamailio/modules/ | grep unix -rwxr-xr-x. 1 root root 184451 28. Apr 08:56 db_unixodbc.so I tried to check the db.c and db_unixodbc.c but I am not familiar with the c-syntax. I somehow suspect that the error is related to Linux case sensitivity if db.c is trying to check if "unixODBC" is installed and then trying to match it with the "db_unixodbc.so" but I'm am not sure. Can someone please give further advice? Respectfully Dimitry Nagorny Trainee

2 5

SIP Notifications
by Safdar Khan 02 May '16

02 May '16

Hello, How i enable push notification in kamailio.How i proceed to configure the kamailio configuration file and which kamailio module should i look for? Thanks

3 4

dlg_bridge
by huseyin kalyoncu 29 Apr '16

29 Apr '16

Hello I am trying to implement click 2 call functionality on Kamailio with dlg_bridge. But It seems that Kamailio routes REFER messages to devices according to Contact field of location table instead of Received field. But if the the IP in the Contact field is private REFER messages are failing so does the dlg_bridge. I called dlg_bridge in the config like this: dlg_bridge("sip:$param1", "sip:$param2", "sip:IP_ADDRESS_OF_KAMAILIO:5060"); It rings the device in $param1 but after answering on that device Kamailio fails to send REFER if the Contact field for that device is Private IP. So it fails to initiate a call. How can I force Kamailio to use Received field for REFER messages instead of Contact field? Thanks Huseyin

2 11

kamailio stop to print log meesage
by Shiv Patidar 29 Apr '16

29 Apr '16

kamailio.log and syslog stop to print log message i have set debug 4 and also try with debug 2 and debug 3 and kamalio.log and syslog not print any log message what i need to do please help me

2 1

How to run MySQL queries in Kamailio
by Ahmad Fuad 28 Apr '16

28 Apr '16

Dear Support, Kindly find below my Database Table (kamailio.ext). Column names are "ext" and "ip" mysql> select * from ext -> ; +------+----------------+ | ext | ip | +------+----------------+ | 202 | 192.168.55.157 | | 203 | 192.168.55.158 | | 204 | 192.168.55.159 | +------+----------------+ 3 rows in set (0.00 sec) I want to run this type of queries in kamailio. I have tried following command but no luck so far. Although I have added following line in the configuration file as well. Thanks modparam("sqlops", "sqlcon", "cb=>mysql://root:123456@localhost/kamailio") sql_query("cb", "select ip from ext where ext=$rU", "ra"); sl_send_reply("302","$ra"); PLEASE HELP !!! when we see the traces using wireshark it shows internal error. Thanks 275.733635441 192.168.55.144 -> 192.168.55.157 SIP 431 Status: 302 Internal Server Error | -- Best Regards, Ahmad Fuad

2 1

kamailio-radius RPM for Centos
by Antonio Reale 28 Apr '16

28 Apr '16

Hi all, I see that the kamailio-radius rpm package is available only for Opensuse. Is there a particular reason that it is unavailable for Centos? Thanks. Regards. Antonio

2 10

fetch data from Mysql Using Kamailio
by Shahid Mehmood 28 Apr '16

28 Apr '16

Dear All, i am getting error when i trying to fetch data from mysql data base using kamailio. i have following code into kamailio.cfg file loadmodule "db_mysql.so" loadmodule "sqlops.so" ................................ modparam("sqlops","sqlcon","ca=>mysql://root:123456@localhost/kamailio") ............................... ............................... sql_query("ca", "select * from ext", "ra"); sql_result_free("ra"); *Apr 24 06:30:31 kamalio /usr/sbin/kamailio[1385]: INFO: rr [rr_mod.c:160]: mod_init(): outbound module not availableApr 24 06:30:31 kamalio /usr/sbin/kamailio[1385]: INFO: usrloc [hslot.c:53]: ul_init_locks(): locks array size 1024Apr 24 06:30:31 kamalio /usr/sbin/kamailio[1385]: ERROR: sqlops [sqlops.c:276]: fixup_sql_query(): invalid connection [ca]Apr 24 06:30:31 kamalio /usr/sbin/kamailio[1385]: ERROR: <core> [route.c:1189]: fix_actions(): fixing failed (code=-1) at cfg://etc/kamailio/kamailio.cfg:514Apr 24 06:30:31 kamalio /usr/sbin/kamailio[1385]: ERROR: <core> [route.c:1189]: fix_actions(): fixing failed (code=-1) at cfg://etc/kamailio/kamailio.cfg:534* -- Shahid Mehmood

2 4

fetch data from Mysql Using Kamailio
by Shahid Mehmood 28 Apr '16

28 Apr '16

Dear All, i am getting error when i trying to fetch data from mysql data base using kamailio. i have following code into kamailio.cfg file loadmodule "db_mysql.so" loadmodule "sqlops.so" ................................ modparam("sqlops","sqlcon","ca=>mysql://root:123456@localhost/kamailio") ............................... ............................... sql_query("ca", "select * from ext", "ra"); sql_result_free("ra"); *Apr 24 06:30:31 kamalio /usr/sbin/kamailio[1385]: INFO: rr [rr_mod.c:160]: mod_init(): outbound module not availableApr 24 06:30:31 kamalio /usr/sbin/kamailio[1385]: INFO: usrloc [hslot.c:53]: ul_init_locks(): locks array size 1024Apr 24 06:30:31 kamalio /usr/sbin/kamailio[1385]: ERROR: sqlops [sqlops.c:276]: fixup_sql_query(): invalid connection [ca]Apr 24 06:30:31 kamalio /usr/sbin/kamailio[1385]: ERROR: <core> [route.c:1189]: fix_actions(): fixing failed (code=-1) at cfg://etc/kamailio/kamailio.cfg:514Apr 24 06:30:31 kamalio /usr/sbin/kamailio[1385]: ERROR: <core> [route.c:1189]: fix_actions(): fixing failed (code=-1) at cfg://etc/kamailio/kamailio.cfg:534* -- Shahid Mehmood

2 1

Kamailio IMS_QOS
by José Seabra 28 Apr '16

28 Apr '16

Hello There, I'm trying using the ims_qos module with PCRF for QOS, but I'm facing some issues. When kamailio receives an initial invite (originator), it will execute the function - "Rx_AAR("ORIG_SESSION_AAR","orig","",-1)", but the PCRF is not contacted and i see the following messages: 5(12153) DEBUG: <script>: Diameter: Orig authorizing media via Rx 5(12153) DEBUG: ims_qos [mod.c:609]: w_rx_aar(): Looking for route block [ORIG_SESSION_AAR] 5(12153) DEBUG: ims_qos [mod.c:621]: w_rx_aar(): Rx AAR called 5(12153) DEBUG: ims_qos [mod.c:1345]: create_return_code(): Creating return code of [-2] for aar_return_code 5(12153) DEBUG: ims_qos [mod.c:1354]: create_return_code(): created AVP successfully : [aar_return_code] 5(12153) DEBUG: ims_qos [mod.c:627]: w_rx_aar(): Can't do AAR for call session in request. Why kamailio says that I Can't do AAR for call session in request? I have looked into source code and I found that the condition that is blocking the communication with PCRF is: //We don't ever do AAR on request for calling scenario... if (msg->first_line.type != SIP_REPLY) { LM_DBG("Can't do AAR for call session in request\n"); return result; } Anyone can help me understand why it is happening? why it can't proceed if isn't a reply? Thank you for your support BR -- José Seabra

4 15

Call http service inside config , Push notification TSILO
by Sirvan Paraste 27 Apr '16

27 Apr '16

Hello All, I am new to kamailio, but I have setup a server that is ok for call functionalities, I have write an android app it I wnat to add push notification ability to the app when it is clised or unregisterd to the sip server. I have done all push notification part using GCM service and a third party server name Uniqush-push that is connected to GCM to handle push notification, anyway the app and push server part all work well now, The problem is this: kamilio send a http request like this : curl http://127.0.0.1:9898/push -d service=myserviceName -d subscriber=6000 -d msg="Hello World" *given 6000 as a kamilio user,* I ask a nice guy and he replied me by using the TSILO as following link: http://www.kamailio.org/events/2015-KamailioWorld/Day2/20-Federico.Cabiddu-… thanks to him it made the process more easier I think. But I dont know how I can congigured kamailio to call my server as I said above in order to send push. please help me by example because I am new, I searched a lot but I get confused, I will make a tutorial for other people to help them. thanks all

2 2

A: Ongoing process of selecting a new website theme
by Daniel-Constantin Mierla 27 Apr '16

27 Apr '16

Hello, following the last developers' meeting on IRC, we are going to update the website at kamailio.org to use a responsive theme, in order to make it more friendly for mobile browsing. Until one is selected, you may see various forms of website design for short time, as variants are enabled to see how it looks with the real content. All this should not take too long, in 1-2 days the new theme should be in place. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com

1 1

force_send_socket
by Nagorny, Dimitry 26 Apr '16

26 Apr '16

Hi all, I've configured Kamailio to use TLS and it has two NICs, 192.x.x.x extern to a PSTN and 10.250.5.74 intern to FreeSWITCH. Kamailio has "listen=tls:10.250.5.74:5071" and later on in the config I fire "force_send_socket(tls:10.250.5.74:5071)". For some reason on calls from FreeSWITCH via Kamailkio to PSTN the TLS works but the other way around if Kamailio should do TLS to FreeSWITCH it throws this warning: "/kamailio[3308]: WARNING: <core> [forward.c:231]: get_send_socket2(): protocol/port mismatch (forced tls:10.250.5.74:5071, to udp:10.250.5.72:5071)" As a result of this Kamailio tries to send the PSTN INVITE via UDP and also tries to use the external IP (192.x.x.x) and of course this can't work. Why is it throwing this warning and how to understand it? Best regards Dimitry Nagorny Trainee

1 1

BLF: Turning a key on/off without a dialog
by Ali Pey 25 Apr '16

25 Apr '16

Hello, I use presence and dialog modules for my BLF keys; however, I have a key (e.g. park) that I need to turn on/off outside a dialog: Certain Invite message would need to cause the BLF key to become busy and another unrelated Invite message would need to turn the BLF key off. What's the best and easiest way to do this? Best regards, Ali Pey

2 2

Are sip calls working OK?
by Aloysius Bolskey Clark Donson 25 Apr '16

25 Apr '16

Dear Friends, I started to use my sip account few monts ago sip:4095xx@iptel.org It was working OK, but at the end of March I realised that I could not get any calls in my account. I own also a sip account in callcentric and works OK. So my zoiper version seems nothing to do on it Have you changed the setup of the account (iptel)? I got a real US phone number in IPCall associated to my sip account. IPCall is giving up, so that is the reason why the calls made to my real phone numbers ar not being forwarded to my sip account. But I have no idea why my sip:4095xx@iptel.org is not working thanks in advance looking forward your kind answer jose

2 1

Using flags in t_branch_timeout() section of failure_route{}
by Marrold 24 Apr '16

24 Apr '16

Hi all, I'm having issues using flags in a t_branch_timeout() of failure_route{} A brief summary of the code - onreply_route{ setflag(14); } failure_route { if (t_branch_timeout()) { xlog("L_ALERT","t_branch_timeout()\n$mb"); if (isflagset(14)) { xlog("L_ALERT","isflagset(14) - YES\n$mb"); } } } The call flow is below, t_branch_timeout() is being triggered when the 200 OK is not ACK'd -- INVITE -> <- 100 Trying -- <- 180 Ringing -- <- 200 OK -- <- 200 OK -- <- 200 OK -- <- 200 OK -- I can attempt to create a minimal .cfg if anyone wishes to replicate it, assuming I'm not missing something stupid. Thanks, Matthew

2 5

NO RTP when using kamailio on public IP
by Ankitt Sharma 23 Apr '16

23 Apr '16

hello, i've been trying to setup my communication setup , everything works fine when i use my kamailio setup within the local network but as soon as i assign a public IP to kamailio , there is no RTP flow between the clients i searched over internet and found that rtpengine module should be used when using the stup on public network ,and tried configureation(with some tweaks) given by https://github.com/caruizdiaz/kamailio-ws i also observed that the call still flows when the two clients are in same network. please help what should i do! thanx in advance! :)

2 1

Kamailio IMS+FHoSS: Error "Visited Network Identity not found, responding with 400"
by Трущенков Алексей 22 Apr '16

22 Apr '16

Good afternoon! I use Kamailio from BOX from Article https://www.kamailio.org/w/2016/02/kamailio-ims-getting-started-box/ The client can not register (use Boghe IMS Client as IMS client) i-cscf log 10(1991) INFO: <script>: [REGISTER] from [sip:alice@net1.test] to > [sip:alice@net1.test] > 10(1991) ERROR: <script>: REGISTER (sip:alice@net1.test (10.0.0.10:1173) > to sip:alice@net1.test, 9848450f-c6dd-5b5e-6516-addaf85a9f2f) > 10(1991) ERROR: ims_icscf [registration.c:149]: > I_perform_user_authorization_request(): ERR:I_UAR: Visited Network Identity > not found, responding with 400 > 10(1991) ERROR: tm [t_lookup.c:1497]: t_get_trans_ident(): ERROR: > t_get_trans_ident: no transaction found I would be glad of any help. Thank you. --------------------- Trushenkov Alexey

2 1

Understanding the new interface (kemi) for embedded interpreters
by Daniel-Constantin Mierla 22 Apr '16

22 Apr '16

Hello, (cross-posting because the details are useful for both devels and users) Yesterday during the IRC meeting about Kamailio, one group of topics was about Kamailio 5.0, and the new feature of using embedded interpreters for external programming languages was highly debated. So I am trying to clarify some of its aspects. The new framework is referenced as kemi, chosen by me as an abbreviation for Kamaialio EMbedded Interface. We already had embedded interpreters for many years, respectively: - Lua via app_lua module (probably most developed and most used) - Perl via app_perl module (iirc, the oldest embedded interpreter) - Python via app_python (I discovered during the last days that it has quite a lot of features, just not documented -- never used it before) - Java via app_java (this one is a bit of a blackhole for me, never looked at it, don't know what it offers/how it is supposed to be used) - .NET group of languages via app_mono (e.g., C#, but also some variants of JavaScript, Python and maybe even Java) To clarify, an embedded interpreter means that Kamailio is linking to itself the interpreter of that language and kamailio becomes the interpreter of the language. It is not launching the standard interpreter for that language, therefore it is very fast at runtime. Moreover, kamailio is extending the language with new modules, giving access to Kamailio C functions. In other words, for example with Python, Kamailio becomes equivalent to the "python" application plus two extensions (modules) named Router (developed in the past inside app_python) and KSR (developed for kemi). At this moment, the kemi was implemented in app_lua and app_python modules. To understand what is all about, I am going to give an example with a function from maxfwd module. In default kamailio.cfg next statement is present: if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } mf_process_maxfwd_header("10") is the function exported by maxfwd: - https://www.kamailio.org/docs/modules/stable/modules/maxfwd.html#maxfwd.f.m… The parameter can be as a static number (like above) or a number provided via a variable, like next: $var(p) = 10; mf_process_maxfwd_header("$var(p)") The corresponding C code for this config function is: static int w_process_maxfwd_header(struct sip_msg* msg, char* str1, char* str2) { int mfval; if (get_int_fparam(&mfval, msg, (fparam_t*) str1) < 0) { LM_ERR("could not get param value\n"); return -1; } return process_maxfwd_header(msg, mfval); } The function get_int_fparam() is practically returning the int value of the parmeter, no matter it was a static value or a variable, then process_maxfwd_header(...) is executed with the int parameter. The msg is the SIP message structure present mostly everywhere in the C code related to the classic config interpreter. To be used inside the config file, the C function has to be exported via a cmd_export_t structure added to "struct module_exports exports", respectively: static cmd_export_t cmds[]={ ... {"mf_process_maxfwd_header", (cmd_function)w_process_maxfwd_header, 1, fixup_var_int_1, 0, REQUEST_ROUTE}, ... To get the same function in a kemi interpreter (Lua, Python at this moment), following C code was added: static sr_kemi_t sr_kemi_maxfwd_exports[] = { { str_init("maxfwd"), str_init("process_maxfwd"), SR_KEMIP_INT, process_maxfwd_header, { SR_KEMIP_INT, SR_KEMIP_NONE, SR_KEMIP_NONE, SR_KEMIP_NONE, SR_KEMIP_NONE, SR_KEMIP_NONE } }, { {0, 0}, {0, 0}, 0, NULL, { 0, 0, 0, 0, 0, 0 } } }; int mod_register(char *path, int *dlflags, void *p1, void *p2) { sr_kemi_modules_add(sr_kemi_maxfwd_exports); return 0; } The structure sr_kemi_t sr_kemi_maxfwd_exports exports directly the existing function process_maxfwd_header(), which is also executed by the function from kamailio.cfg after resolving the parameter to an int value. In the case of kemi, the Lua or Python will give directly an integer value, so no need for resolving that with a fixup wrapper. The mod_register() function is something already existing, providing a way to execute code when a module was loaded (e.g., is used for many years by modules such as tls, mysql, etc.). In this case it was added, because maxfwd didn't need anything at load time before. Practically, no C code extension was written to maxfwd module from point of view of SIP routing. Just an existing function was exported. Now, what's actually the role of this kemi framework ... The app_lua already exported the kamailio.cfg function mf_process_maxfwd_header() as sr.process_maxfwd() since Kamailio v3.1, see its usage in the tutorial: - https://kb.asipto.com/kamailio:usage:k32-lua-routing#lua_script But a Lua specific wrapper had to be written: - https://github.com/kamailio/kamailio/blob/master/modules/app_lua/app_lua_ex… All the other embedded interpreters would have needed to get a wrapper as well to export this function. That's obviously costing time and maintenance resources. So kemi is a common framework to export C function to any of the embedded interpreters, meaning that once a function is exported from a module, it becomes available to all embedded interpreters, no longer being necessary to write a specif wrapper per interpreter. The app_* module needs a bit of coding to use the kemi exported functions (already done for Lua and Python), but then nothing needs to be done for new exported functions by modules. As a matter of fact, lots of functions can be just exported, as you could see in the example with maxfwd. In some cases, maybe some functions need to be split, so the fixup handling is done separately and then calls a common C function. So, what we had so far: [native kamailio cfg script] <===> [fixup framework for variables] <===> [C implementation of function] The above stays in place, no change to it. What we got extra: [embedded language script] <===> [kemi framework] <===> [C implementation of function] In addition, kemi allows to write the routing blocks all in embedded language. Core parameters, loading modules and modules' parameters stay like so far, but there is no need to write request_route{}, failure_route, etc.. blocks. Some functions with a specific names have to be written in the embedded language script, see more at: - https://www.kamailio.org/wiki/devel/config-engines In several days I expect to be possible to have routing blocks of kamailio-basic.cfg written via kemi interpreters. That will allow testing the performance impact. Right now Lua and Python use different mechanism of exporting Kamailio C functions, the one from Python should be faster, but relies on defining some C wrappers inside the module. Anyhow Lua might be faster as interpreter itself -- and at the end it can just get the same mechanism as Python. app_lua already supports reloading the routing script without restarting kamailio, one of the features asked from time to time. Besides that, of course, a big benefit is the access to a very large set of extensions already available as Lua/Python libraries. Also important, many peoples are already familiar with these languages and there is plenty of good documentation about those languages. The classic/native kamailio.cfg format (and the interpreter for it) stays there -- same common C functions are shared with kemi, so any addition in the future will be visible to everywhere. The native interpreter will remain the option for extreme optimizations for those that deal with enormous amount of SIP traffic (e.g., heavy load balancers), but I expect that at least Lua will be in pair of performances with a registrar deployment. I hope I could shed some light on various aspects of the new framework for embedded interpreters. Expect very soon news about ability to build full routing logic in Lua/Python, hopefully many of you will join to test the results and compare with native alternatives. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com

1 0

irc devel meeting summary
by Fred Posner 22 Apr '16

22 Apr '16

Greetings, Minutes from the recent devel meeting have been posted to the wiki: https://www.kamailio.org/wiki/devel/irc-meetings/2016a-minutes In summary, this was a 3+ hour meeting held on IRC #kamailio -- one of the longer meetings on record. No current "critical" issues were open, and the meeting moved to a discussion of for a testing framework and being able to load as many modules as possible in a running Kamailio system. Olle Johansson suggested a competition for Kamailio World: "The one with a kamailio with most loaded modules running at kamailio-world gets a beer." A discussion then occurred regarding the appropriateness of returning a 200 OK when no data exists during a xmlrpc request. The consensus seemed to side with how we currently return data with the suggestion of moving further discussion to the mailing list. The Kamailio team will be looking to both upgrading the Kamailio servers to Debian Jessie as well as utilizing a "responsive" template for the main website; making the site more mobile/user friendly. Kamailio 5 was discussed with a suggestion for a developers meeting in Stockholm, perhaps in June, to hammer out the framework. Daniel was thanked for moving forward with the Lua routing aspects of Kamailio 5 while ensuring that you will still be able to utilize the config file (as current) without Lua. Additional languages, such as python, will be coming. Daniel will be working on a tutorial for exporting the kamailio.cfg to an embedded interpreter in the future. As part of Kamailio 5, the source tree structure will be improved, most likely utilizing subdirectories to better organize the different elements of the software. There will also be a change in the method of checking the database schema for compatibility with the version of software running. This summary represents just an appetizer of the incredible meeting that took place. The minutes are available for you to read at your leisure and your participation is always encouraged through the mailing lists, IRC, as well as github. The meeting never ends... it just continues on the mailing lists. =) Best regards, Fred Posner http://www.palner.com

2 1

A: Open discussion about Kamailio in about 30min
by Daniel-Constantin Mierla 22 Apr '16

22 Apr '16

Hello, long time involved in VoIP and friend of Kamailio, Suzanne Bowen has organized an open discussion about Kamailio on a web platform blah.im. The link of the session is: - https://blab.im/didx-exchange-kamailio-world-talk-on-privacy-scalability-re… It is planned to start at 12:00GMT (13:00 London, 14:00 Berlin), I will join it and I expect that the platform is allowing many to watch and ask questions. Hopefully few other Kamailio folks will join the conversation. Rather short notice here, but if you have something to ask about SIP, VoIP, Kamailio, you should try to participate. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com

1 0

BYE packet looping and loose routing
by Yasin CANER 22 Apr '16

22 Apr '16

3 3

The book of "SIP Routing with Kamailio"
by Jay Li 22 Apr '16

22 Apr '16

Dear all, I'm looking for a document/book that explains the kamailio.cfg in detail. I've gone over the suggested study material "SER-Getting Started" which is such a nice book that explains ser.cfg line by line. I know Asipto has published a book named "SIP Routing with Kamailio". Just wondering if "SIP Routing with Kamailio" is written in a similar format when it comes to explain the kamailio.cfg. Hopefully I can get some information about the book before I make the purchase. Thanks. Regards,Jay

3 2

Next Kamailio IRC Devel Meeting, Apr 20, 2016
by Daniel-Constantin Mierla 21 Apr '16

21 Apr '16

Hello, I am proposing a new IRC meeting to discuss the current major issues and the plans for next Kamailio release, on Wednesday, April 20, 2016, with alternative for next day, April 21. If many developers are not available, we can postpone it to another date in the near future (make proposals if that is the case for you). I created a wiki page for it: * https://www.kamailio.org/wiki/devel/irc-meetings/2016a Add there the topics that you want to be discussed. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com

1 3

DMQ_USRLOC sync CSEQ error
by José Seabra 21 Apr '16

21 Apr '16

Hi There, I'm using dmq and dmq_usrloc between 2 kamailio(version 4.4.0) registrar servers. I'm registering 15000 subscribers with sipp and everything is ok, the registration address of record are shared between both registrar servers. The problem here is when I restart one of the registrar server and then the other registrar server starts synchronization to restarted server. The error that i see on this process is the following shown bellow: ERROR: dmq_usrloc [usrloc_sync.c:75]: add_contact(): Invalid cseq But all the contacts are synchronized, what can cause this error? and what is the impact on normal behavior of kamailio? Best regards -- José Seabra

2 2

about xml rpc response behavior
by Yasin CANER 21 Apr '16

21 Apr '16

3 4

kazoo database not bound error
by Jayesh Nambiar 21 Apr '16

21 Apr '16

Hello, Using kazoo only to connect to amqp gives an error if the db_url is not specified. If db_url is specified it tries to do presence related stuff even if not intended. The solution was to use: modparam("kazoo", "pua_mode", 0) which I found here: http://lists.sip-router.org/pipermail/sr-users/2015-September/090092.html This is not documented in the kazoo module. Would be great if this is added in the module documentation to simplify the use of module. Thanks, - Jayesh

2 1

How to distinguish 200 OK that have byed
by 张顺通 20 Apr '16

20 Apr '16

A KA B INVITE -> 200 OK(INVITE) <- ACK is lost BYE <- 200 OK(BYE) -> 200 OK(INVITE) <- 200 OK(INVITE) <- 200 OK(INVITE) <- ACK is lost AND A sent BYE cmd,B do not recv ACK and is also Retransmission 200 OK, How can I distinguish this 200 OK and drop it? thaks

3 3

A: Kamailio World 2016 - Four Weeks Before
by Daniel-Constantin Mierla 20 Apr '16

20 Apr '16

Time is passing and Kamailio World Conference 2016 is approaching at fast pace – only four weeks left till the start of the event! The schedule [1] is pretty much nailed down, with some adjustments still expected to happen. The event starts like the past edition with a half a day of technical workshops, followed by two full conference days. A larger group of speakers [2] is participating to this edition. There was a big number of speaking proposals and we wanted to highlight more of the people that had a relevant contribution to the evolution of the project. To accommodate properly, two more discussion panels were added, keeping also the classic VUC panel. The topics cover many of the interesting aspects of real time communications, from security and scalability to WebRTC and VoLTE, touching Kamailio and other open source projects like Asterisk or FreeSwitch. More details can be found on the website of the event: - https://www.kamailioworld.com Don’t forget that this year Kamailio celebrates 15 years of development, the party is at Kamailio World! We expect to fill the capacity of the conference room, if you haven’t registered yet and plan to attend, do it as soon as possible to secure your seat! [3] Many thanks to our sponsors [4] that made possible this event: FhG Fokus, Asipto, Sipwise, Matrix.org, Sipgate, Simwood, NG Voice, Digium, VoiceTel, Evariste Systems, Core Network Dynamics, Pascom, Didx.net. Thank you for flying Kamailio and looking forward to meeting many of you at Kamailio World 2016! Daniel [1] https://www.kamailioworld.com/k04/schedule/ [2] https://www.kamailioworld.com/k04/speakers/ [3] https://www.kamailioworld.com/k04/registration/ [4] https://www.kamailioworld.com/k04/sponsors/ / / -- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com

1 0

I have kamailio with drouting modulo and I want to pass only the RTP trafic to a FREESWITCH server
by Humberto H 20 Apr '16

20 Apr '16

Hello: I have kamailio with drouting and I want to pass only the RTP trafic to a FREESWITCH o what ever you think is better to manage de RTP traffic. What opcions do I have? I am a little bit confused using WITH_FREESWITCH. I will appreciate your help, Thanks

2 1

another bug with JSONRPC-S?
by Brooks Bridges 20 Apr '16

20 Apr '16

I may have stumbled across another bug with this module. I believe the htable.dump method isn't correctly recognizing that a table name is being provided: [root@server ~]# curl -s --header 'Content-Type: application/json' --data-binary '{"jsonrpc": "2.0", "method": "htable.dump", "name": "callstats"}' http://1.2.3.4:5060/jsonrpc {"jsonrpc":"2.0","error":{"code":-32000,"message":"Execution Error"}} Debug log: 2016-04-19T17:42:30.046892+00:00 server /usr/local/sbin/kamailio[21634]: DEBUG: <core> [tcp_main.c:2465]: tcpconn_do_send(): buf=#012HTTP/1.1 500 No htable name given#015#012Sia: SIP/2.0/TCP 1.2.3.4:54912#015#012Content-Type: application/json#015#012Server: kamailio (4.4.0-rc1 (x86_64/linux))#015#012Content-Length: 69#015#012#015#012{"jsonrpc":"2.0","error":{"code":-32000,"message":"Execution Error"}} Brooks Bridges | Sr. Voice Services Engineer O1 Communications 5190 Golden Foothill Pkwy El Dorado Hills, CA 95762 office: 916.235.2097 | main: 888.444.1111, Option 2 email: bbridges(a)o1.com<mailto:bbridges@o1.com> | web: www.o1.com<http://www.o1.com/>

2 4

$shtval(htable=>key)
by Brooks Bridges 19 Apr '16

19 Apr '16

What is this? It's referenced here: http://kamailio.org/docs/modules/stable/modules/htable.html#idp1912904 but I cannot find it anywhere else. The only 2 places I find any reference to it at all in the entire codebase is here: modules/htable/doc/htable_admin.xml: <emphasis>$shtval(htable=>key)</emphasis> modules/htable/README: * $shtval(htable=>key) Brooks Bridges | Sr. Voice Services Engineer O1 Communications 5190 Golden Foothill Pkwy El Dorado Hills, CA 95762 office: 916.235.2097 | main: 888.444.1111, Option 2 email: bbridges(a)o1.com<mailto:bbridges@o1.com> | web: www.o1.com<http://www.o1.com/>

3 3

Kamailio Blind Call Forwarding
by Nagorny, Dimitry 19 Apr '16

19 Apr '16

Hi all, I'm trying to make a blind call forward to "1002" if a UA calls "1" into Kamailio. At the moment I have the following config: if ( $rU=~"^[1]$" && src_ip==$sel(cfg_get.pstn.gw_ip) ) { sl_send_reply("181", "Redirecting"); $ru = "sip:1002@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port) + ";user=phone"; force_send_socket(udp:MY_EXTERN_IP:5060); route(NORMAL_RELAY); #Just t_on_branch/reply/failure if (!t_relay()) { sl_reply_error(); } exit; } The calling UA is from the PSTN as well as the called UA. Problem right now is that the PSTN's anti-Loop protection jumps in and rejects the INVITE to 1002. Question: Is it possible to tell the calling UA "Hey wait, 181..." with Kamailio, then build seperately an independent Call Leg (with new CallID?) from Kamailio to UA "1002" and finally to put them together? Any help is very much appreciated even if they are just guides/examples. I'm using Kamailio 4.3.5 on CentOS 7.2. Very Respectfully Dimitry Nagorny Trainee

5 5

Cleaning merged branches
by Daniel-Constantin Mierla 18 Apr '16

18 Apr '16

Hello, it looks that many branches still exist after being merged, discovered by running git-sweep. The list is next. If anyone from devs or users spots a branch that was not merged and needs to be reviewed, let me know. In several days I plan to remove them to keep the git repo a bit more sane. Cheers, Daniel $ git-sweep preview Fetching from the remote These branches have been merged into master: andrei/blst_send_flags andrei/cancel_reason andrei/cdefs2doc andrei/counters andrei/fixups andrei/mod_f_params andrei/mod_register andrei/module_interface andrei/path andrei/pointer_alias_warnings andrei/raw_sock andrei/rpc_async andrei/rve_f_params andrei/script_vars andrei/send_flags andrei/shm_early_init andrei/switch andrei/tcp_tls_changes andrei/to_parser_fix andrei/type_conversion co/crypto_name_collision co/tcp_closed_event_enhancements curl cvs-head ez/newstuff henning/trie henning/trie_mods janakj/missing_imports kamailio_modules lazedo-patch-1 lazedo-patch-2 lazedo/presence_improvements lazedo/presence_xml_fix_dummy luismartingil/msrp_crash mariusbucur/dmq mariuszbihlei/dnssec misi/ua-profile sd_journal_send_xavp ser_core_cvs ser_modules tirpi/cfg_framework_multivalue tirpi/script_callbacks tmp/build_local tmp/build_request tmp/core_events tmp/hpw_curl_improvements tmp/k3.0_sr_backports tmp/ruri_branch tsearle-futex-try tteras/db_sqlite -- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com

1 0

loose_route fails
by Igor Potjevlesch 18 Apr '16

18 Apr '16

Hello, I'm trying to understand why a RE-INVITE is still discard by Kamailio (4.2.6). The RE-INVITE has the last Contact URI as Request-URI which is not Kamailio himself. But there are the route headers set. What are the other checks done by the function "loose_route()"? Regards, Igor.

2 1

DISPATCHER module No destinations availables even if my Asterisk Boxes are up and running
by Sébastien Brice 18 Apr '16

18 Apr '16

Hello everyone. I'm using kamailio as a dispatcher in front of asterisk boxes and i use a failure route if asterisk box does not respond or send 503 error (please look at my attached kamailio.cfg) I use "ds_probing_mode", 1 and "ds_ping_reply_codes", "class=2;class=3;class=4" to OPTIONS ping the two asterisk instances but they do appear down both all the time. That is very annoying none of the INVITE go through Asterisk, they all are stopped because of my failure_route that sets no destination available ERROR: <script>: Destination down: INVITE sip:9980048587420715@XXX.XXX.XXX.XX (sip:ipofAsterisk1:5060) ERROR: <script>: Destination down: INVITE sip:9980048587420715@YY.YY.YYY.YY (sip:ipofAsterisk1:5060) ERROR: <script>: Destination down: INVITE sip:000442030044700@XXX.XX.X.XXX (sip:ipofAsterisk2:5060) ERROR: <script>: Destination down: INVITE sip:000442030044700@YY.YYY.YY.YYY (sip:ipofAsterisk2:5060) NOTICE: <script>: Media server sip:172.16.0.202:5060 failed to answer, selecting other one! NOTICE: <script>: No destination, sent 404 I don't know what's wrong with my setup, could someone give an eye to my kamailio.cfg and head me on the right track ? Thanks a lot. Seb

3 5

mod_com_amd only sees silence
by Michael Jepson 18 Apr '16

18 Apr '16

When placing a call and checking for an answering machine using mod_com_amd, it always sees *silent_state*. Strangely, when I stream audio TO the session, mod_com_amd registers this audio and responds accordingly. So, it seems that somehow mod_com_amd is listening to the wrong end of the session. Any help would be greatly appreciated. Best regards, Michael Jepson

2 2

Check if xavp is set
by Marrold 17 Apr '16

17 Apr '16

Hi all, What's the most efficient way to check if an xavp is set? I have tried- is_avp_set() from avpops module which resulted in- ERROR: avpops [avpops.c:915]: fixup_is_avp_set(): bad attribute name <$xavp(tm_contacts)> and if ($xavp(tm_contacts)) {} which resulted in - WARNING: <core> [rvalue.c:1007]: rval_get_int(): automatic string to int conversion for "<<xavp:0x7feac74cfcb8>>" failed To give a little background, I am using the UAC_redirect module and t_next_contacts() in conjunction with 302 redirects .However if there is only a single contact, $xavp(tm_contacts) is not created and t_next_contacts() fails, hence why I'd like to check if it exists first. Thanks

2 2

Kamailio not replay response
by Shiv Patidar 17 Apr '16

17 Apr '16

In kamailio server register is done with 200ok response and now i send invite but kamailio not give replay 200ok What i need to do configure of kamaulio file or not And kamailio.log file show some error like Parse_first_line():parse_first_line: bad message And Parse_msg:message=<INVITE ....to, from,via all filed.......> And Receive _msg(): CORE PARSING OF SIP MESSAGE FAILED

2 2

Random number function?
by Brooks Bridges 15 Apr '16

15 Apr '16

I feel like I have to be missing something obvious here, but I can't for the life of me find anything. Is there not a basic random number/string generation function in core or any of the modules anywhere? I find that incredibly unlikely that such a simple function that could have so many uses wouldn't exist, but I just can't find anything... Any pointers? Brooks Bridges | Sr. Voice Services Engineer O1 Communications 5190 Golden Foothill Pkwy El Dorado Hills, CA 95762 office: 916.235.2097 | main: 888.444.1111, Option 2 email: bbridges(a)o1.com<mailto:bbridges@o1.com> | web: www.o1.com<http://www.o1.com/>

2 1

Siremis install error
by Serge S. Yuriev 15 Apr '16

15 Apr '16

Hello, I'm trying to setup latest Siremis from GIT and stuck on DB stage. ERROR: SQLSTATE[08006] [7] FATAL: password authentication failed for user "www-data" FATAL: password authentication failed for user "www-data" Why installer tries to use web server user instead of supplied? DB is PGSQL at localhost. Kam DB created successfully via kamdbctl. -- Serge S. Yuriev Lead VoIP engineer

3 6

sipcapture module
by Ali Taher 15 Apr '16

15 Apr '16

Hello, I'm using sipcapture module in Kamailio to capture sip packets and save them into mysql database. Everything is working as expected and I got the packets registered in the database , yet I'm wondering if there is a way to open these packets directly in Wireshark , or do I need an intermediate step to convert these packets to pcap format (if possible) in order to be read by Wireshark. Any help or hint would be appreciated. Ali Taher Technical Support Engineer T&F M W A +961-5-457437 Ext 215 +961-70-010600 <http://www.vanrise.com/> http://www.vanrise.com <https://maps.google.com/maps/ms?msa=0&msid=208243362929850217065.0004cbc8d6 12c5f9b4237&ie=UTF8&t=h&vpsrc=0&z=19&iwloc=0004cbc8e4652b4edf71cC:\Users\Mar ie\Documents\My%20Received%20Files> Vanrise Building, Facing Galerie Semaan Hazmiyeh Highway - Beirut, Lebanon Join us at <http://www.facebook.com/pages/Vanrise-Solutions-Offshore/128309743915533> <http://www.linkedin.com/company/vanrise-solutions> <http://twitter.com/VanriseSolution>

2 1

sipcapture module with wireshark
by Ali Taher 15 Apr '16

15 Apr '16

Hello, I'm using sipcapture module in Kamailio to capture sip packets and save them into mysql database. Everything is working as expected and I got the packets registered in the database , yet I'm wondering if there is a way to open these packets directly in Wireshark , or do I need an intermediate step to convert these packets to pcap format (if possible) in order to be read by Wireshark. Any help or hint would be appreciated. Regards, Ali

2 1

Help with Kamailio app_java module
by Clarence Sandjon 15 Apr '16

15 Apr '16

Hi! I am trying to use the app_java module with kamailio-4.4.0 SIP server. Every time I load the app_java module, my users can't register and the sever replies with destintion unreachable (port unreachable). How can I fix this issue? Also, how can I use the Kamailio.java file since it doesn't have a main method? Thanks

2 2

should the AVPs be available in CANCEL?
by Andrew Pogrebennyk 14 Apr '16

14 Apr '16

Hi all, it seems that the AVPs not available when when processing CANCEL message, even though they have been set for this transaction initially. Is this the expected behavior? P.S. kamailio 4.3.4 Andrew

2 2

uac "restore_mode" is auto: effective when using core variables?
by Andrew Pogrebennyk 14 Apr '16

14 Apr '16

Per my understanding the uac module stores the "vsf" parameter in Record-Route and should be able to update the From/To URIs automatically in all in-dialog requests that carry this parameter. http://kamailio.org/docs/modules/stable/modules/uac.html#uac.p.restore_mode “auto” - all sequential requests and replies will be automatically updated based on stored original URI. For this option you have to set “modparam("rr", "append_fromtag", 1)”. What makes me wonder is: does that only work in From/To was changed by uac_replace_from()/uac_replace_to() or also when assigning directly to the $fu and $tu variables? I'm changing those variables and I am using restore_mode auto but that does not change anything in in-dialog ACK. I presume this is expected behavior because I'm not using uac when assigning to variables, isn't it? Thanks. Andrew

2 1

A real mystery - 1.5.x vintage edition!
by Alex Balashov 14 Apr '16

14 Apr '16

Hi, I've got an installation that is still running Kamailio 1.5.x. Yes, I know: "They're running WHAT?" As far as I'm concerned, it was EOL'd in early 2010. Nevertheless, that doesn't stop the customer from running it. They're probably the most change-resistant customer in the history of IP telecom, and, knowing this industry, that's a high bar. There's not much I can do about it. Anyway, I have a bit of a mystery: GW Proxy PBX ================================================== ------- INVITE ------> <---- 100 Trying ---- -------- INVITE --------> <------ 100 Trying ------ <- 503 Service Unavailable ---------- ACK ---------> <- 500 Service Una... -------- ACK --------> There is no basis in the configuration for the translation of the "503 Service Unavailable" reply to "500 Service Unavailable", so I'm thinking it has to be rooted in some default module behaviour. Any insight would be appreciated! -- Alex -- Alex Balashov | Principal | Evariste Systems LLC 1447 Peachtree Street NE, Suite 700 Atlanta, GA 30309 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 5

RFC6442: geolocation
by Michael Jepson 14 Apr '16

14 Apr '16

Hello everybody, Can anybody help me out with sending a geolocation in the SIP messages that Kamailio sends? In order for the emergency telephone number to work properly, we need to send the location of our SIP trunk customers along with their INVITE when calling the emergency phone numbers. Is there any module that will facilitate this, or is there another way to add this information in the INVITE message? Thanks in advance! Best regards, Michael Jepson

1 0

Kamailio setting the reqeust URI
by NITESH BANSAL 13 Apr '16

13 Apr '16

Hello,I'm using Asterisk to originate a call via Kamailio.Request-URI in SIP INVITE coming from Asterisk looks like this<sip:kamailio@x.x.x.x>But my objective is to use Kamailio to forward the call to a remote endpoint. What header should I put in the SIP INVITE from Asterisk to Kamailio to conveythat Kamailio should use this 'SIP URI' to route the call onwards.I tried 'Route' header, but it doesn't seem very clean, as kamailio doesn't updatethe Request-URI in the forwarded INVITE if I use the Route header. Thanks,Nitesh

3 4

Kamailio TLS and SIP-Package building
by Nagorny, Dimitry 13 Apr '16

13 Apr '16

Good afternoon all, I have two questions regarding my configs: kamailio.cfg<http://pastebin.com/32PCh8n0> tls.cfg<http://pastebin.com/gnWZeD9e> . I am using Kamailio 4.3.5 on Cent OS 7.2 and all relevant pakage up-to-date. 1.) When I try to start Kamailio I get the following warning: WARNING: tls [tls_mod.c:287]: mod_init(): tls support is disabled (set enable_tls=1 in the config to enable it) Q: As you can see (line 36) TLS is enabled in config. So what could be the problem? 2.) Lines 324-330 indicate that I want to redirect to 2031 if I'm called at 1. For some reason the SIP-Trace tells me its still building the INVITE with the To-Header to sip:1@PSTN:5060 and the PSTN can't find 1, because no 1 is registered. Q: Why? (Outbound is set the same way and working perfectly). Info: I tried already with uac_replace_to and uac_replace_from w/o success. I would very much appreciate any hints on solving these issues. Very Respectfully Dimitry Nagorny Trainee

2 2

Ldap authentication
by Volkan Oransoy 13 Apr '16

13 Apr '16

Hi all, I am trying to use an ldap directory for authentication and found this tutorial. https://www.kamailio.org/wiki/tutorials/mini-howto-admin/ldap-user-auth What I want is to use ldap bind with user's existing ldap password, not a seperate sip password. I would like to use same password for simplicity. There are two ways in my mind to achieve this and I need suggestions. One way is to add a new field to the ldap tree and populate it with user password's ha1 values but this needs an additional work and doesn't seems nice to me. Second way is to get clear-text password from auth request and bind to ldap server with this password but I don't know if I can access to the password or it this is a secure way. Does anybody have any suggestions or experience to share? Regards, /Volkan

1 0

Misrouting ACK packet for 200 Ok messages.
by Yasin CANER 13 Apr '16

13 Apr '16

4 3

Stable DEB
by Serge S. Yuriev 13 Apr '16

13 Apr '16

Hi, Shouldn't http://deb.kamailio.org/kamailio point to 4.4 version? And question about recent Ubuntu version support: Trusty is way to far behind latest Willy -- Serge S. Yuriev Lead VoIP engineer

2 2

Detecting calls with missing ACK (Lazy SIP scanners)
by Marrold 13 Apr '16

13 Apr '16

Hi, I have been running a couple of Asterisk honey pots to get a better understanding of the tools and methods potential hackers are using to exploit SIP servers. I have observed many attacks from the 'sipcli' user agent that don't send ACKs. At this stage I'm not sure what they're trying to achieve, whether it's a successful call to one of their test numbers, or maybe they will brute force anything that returns a 401 later, or maybe they're waiting for a 18X response. Below are three typical scenarios- ------ INVITE ------ > <--- 100 Trying --- <----- 200 OK ----- <----- 200 OK ----- <----- 200 OK ----- ( No ACK) ------ INVITE ------ > <-------- 503 -------- <-------- 503 -------- <-------- 503 -------- ( No ACK) ------ INVITE ------ > <-------- 401 -------- <-------- 401 -------- <-------- 401 -------- ( No ACK) Please could anyone point me in the right direction to detect these non completed calls with a missing ACK in Kamailio? I am unsure on the terminology I should be using to search the online documentation. Thanks

5 11

Kamailio 5.0 - Routing blocks in embedded scripting languages - Lua
by Daniel-Constantin Mierla 12 Apr '16

12 Apr '16

Hello, based on some of the plans for Kamailio 5.0 (https://www.kamailio.org/wiki/devel/kamailio-5.0-design), I started the implementation for the support to allow writing the active routing logic at runtime (respectively what are now the routing blocks) in other embedded languages than the native kamailio.cfg interpreter. Of course, besides a lot more extensive set of functions and expressions coming with the well established scripting languages, this way will allow reloading the logic without kamailio restart. I already committed the foundation for it, currently the request_route{} and reply_route{} can be implemented in Lua, relying on app_lua module. Lua was selected for first experiments given that I have some past experience with it, but other languages such as Perl, Python will hopefully follow soon, once the framework gets to a stable design. I wrote some initial docuentation on the wiki: - https://www.kamailio.org/wiki/devel/config-engines Would be helpful if people with experience on the other supported embedded languages will join the effort to design the framework to suit those languages, as well as other developers and community members to start exporting C functions. Also, there are some other things to decide on, listed in the to-do section, where suggestions/assistance are appreciated: - https://www.kamailio.org/wiki/devel/config-engines#to-do Many of them probably will be approached during the planned IRC devel meeting next week: - https://www.kamailio.org/wiki/devel/irc-meetings/2016a Any feedback is welcome! Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com

1 0

Strange behaviour in working with CNXCC
by Dirk Teurlings 12 Apr '16

12 Apr '16

Hi, Working with CNXCC for quite some time now with success. Though running into the following problem. Running Kamailio 4.4 (Debian Wheezy) from the Kamailio repository. Setting up CNXCC with a dialog start event route, a dialog end event route and a cnxcc:shutdown event route. In the start event I set some variables in $var_dlg, because I will need them later on in the dialog, could be the end event, or the shutdown event. Now the strange thing is, that in the shutdown event, when CNXCC runs out of credit for instance, I no longer have access to the $var_dlg pool. All of the variables have been lost. Another strange thing is, if I cut off the call with CNXCC after just 1 second, the variables all are still available. But after 5 seconds cutoff, all variables are lost. On another system, running a 4.2 Kamailio with a CNXCC ported from 4.1, this still works fine. I've also tested this with Kamailio 4.2 and 4.3, with the same CNXCC code from 4.4. Any insight would be greatly appriciated! Cheers, Dirk

1 0

4.4 crash at start
by Juha Heinanen 12 Apr '16

12 Apr '16

I build kamailio from latest 4.4 git source and got crash at start: (gdb) where #0 0x0000000000449af8 in add_interfaces ( if_name=0x7f0e770106e8 "192.98.102.10", family=0, port=13880, proto=1978, ai_l=0x1277600) at socket_info.c:1210 #1 0x000000000044ad77 in fix_socket_list (list=0x5, type_flags=0xffffffff) at socket_info.c:1406 #2 0x000000000044de70 in fix_all_socket_lists () at socket_info.c:1807 #3 0x000000000041b535 in main (argc=0, argv=0x7fff8cfaf1e8) at main.c:2390 -- Juha

3 9

Extracting P-Asserted-Identity value
by Grant Bagdasarian 11 Apr '16

11 Apr '16

Hello, How do I extract the value of "tel" from multiple P-Asserted-Identity headers? Example: P-Asserted-Identity: "Cullen Jennings" <sip:fluffy@cisco.com> P-Asserted-Identity: tel:+14085264000 What I got so far is: $var(p_asserted_identity) = $(hdr(P-Asserted-Identity)[*]); $var(caller_id) = $var(p_asserted_identity){param.value,'tel',':'}; But this does not work. I keep getting parser errors when doing the transformation (2nd line of code example): I'm not sure how to format the line of code. I tried multiple ways, but none work. Apr 8 18:35:20 kamailio: : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 400, column 47: syntax error Apr 8 18:35:20 kamailio: : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 400, column 47: bad command Apr 8 18:35:20 kamailio: : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 400, column 47: bad command Apr 8 18:35:20 kamailio: : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 400, column 53: '('')' expected (function call) Apr 8 18:35:20 kamailio: : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 400, column 53: bad command: missing ';'? Apr 8 18:35:20 kamailio: : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 400, column 59: '('')' expected (function call) Apr 8 18:35:20 kamailio: : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 400, column 59: bad command: missing ';'? Apr 8 18:35:20 kamailio: : <core> [cfg.y:3365]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 400, column 60-64: bad command Apr 8 18:35:20 kamailio: : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 400, column 65: bad command Apr 8 18:35:20 kamailio: : <core> [cfg.y:3365]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 400, column 66-68: bad command Apr 8 18:35:20 kamailio: : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 408, column 6: syntax error Apr 8 18:35:20 kamailio: : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 408, column 6: Regards, Grant

3 6

Help with Kamailio java module
by Bonjour Madame 11 Apr '16

11 Apr '16

Hi! I am a newbie and I need help understanding how to use the java module with Kamailio. I am trying to create a basic java application that will print the SIP message type being processed by Kamailio. I read the app_java module but can't still figure it out. Any help or reference will be greatly appreciated. Thanks Sent from Mail for Windows 10

2 3

Error in USERLOCDB
by Alberto Sagredo 11 Apr '16

11 Apr '16

Im getting this error using USERLOCDB Apr 7 22:11:56 kamailio-int /usr/local/sbin/kamailio[29717]: ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on query: Duplicate entry '0--1' for key 'connection_idx' (1062) Apr 7 22:11:56 kamailio-int /usr/local/sbin/kamailio[29717]: ERROR: <core> [db_query.c:235]: db_do_insert_cmd(): error while submitting query Apr 7 22:11:56 kamailio-int /usr/local/sbin/kamailio[29717]: ERROR: usrloc [ucontact.c:684]: db_insert_ucontact(): inserting contact in db failed xxxx (uloc-5706bebd-7415-3) Apr 7 22:11:56 kamailio-int /usr/local/sbin/kamailio[29717]: ERROR: usrloc [urecord.c:591]: insert_ucontact(): failed to insert in database Apr 7 22:11:56 kamailio-int /usr/local/sbin/kamailio[29717]: ERROR: registrar [save.c:522]: insert_contacts(): failed to insert contact My config # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "timer_interval", 60) modparam("usrloc", "db_mode", 1) modparam("usrloc", "db_timer_clean", 1) modparam("usrloc", "use_domain", MULTIDOMAIN) modparam("usrloc", "db_ops_ruid", 1) modparam("usrloc", "db_check_update", 1) #!endif I have been using it without any problems in other installations but in a new one with 4. KAMAILIO im getting this error. I can see registrations on kamctl ul show but seems that option 1 is not working . Option 2 does not update registrations from that device. Thanks

2 3

truncating sip message
by Yasin CANER 11 Apr '16

11 Apr '16

4 10

Multiple SIP-servers with SRV-records and authentication secrets
by Alfred E. Heggestad 09 Apr '16

09 Apr '16

Dear SIP-experts and DNS-SRV gurus; I have some questions to the deployers of SER/Kamailio and best current practice for multiple SIP-servers with SRV-records and authentication. This is not a question about Kamailio itself but rather experience with deployment of it in the field. The current usecase is: 1. Multiple SIP-servers are deployed for the same domain 2. The DNS is configured with SRV-records for load balancing, example: (lets call the domain "example.com") $ host -t SRV _sip._udp.example.com _sip._udp.example.com has SRV record 20 0 5080 alpha1.example.com. _sip._udp.example.com has SRV record 20 0 5080 alpha2.example.com. 3. when a SIP client registers, it resolves the domain using RFC3263 [1] and the first REGISTER request is sent to SIP-Server #1 4. SIP-server #1 replies with 401 containing the authentication challenge 5. The SIP Client adds the authentication header to the REGISTER request and re-sends it, but this time also using RFC 3263, and due to DNS rotation the request is sent to SIP-Server #2 6. Now, because the SIP-Servers are configured with _different_ secrets in the "auth" module [2], the REGISTER request fails with authentication error. Now, I know that it is common for SIP user-agents to send both requests to the same SIP-server instance. Baresip [3] is not doing that, it does a new RFC 3263 lookup for all requests (except e2e ACK/CANCEL). so here are my questions: - What is common practice in the field, to configure auth module with the same "secret" or different "secret" values? - Do you know if there is any reference to IETF documents about how this should be handled? RFC 3263 says that every request should be resolved, except: "The procedures here MUST be done exactly once per transaction, where transaction is as defined in [1]. That is, once a SIP server has successfully been contacted (success is defined below), all retransmissions of the SIP request and the ACK for non-2xx SIP responses to INVITE MUST be sent to the same host. Furthermore, a CANCEL for a particular SIP request MUST be sent to the same SIP server that the SIP request was delivered to." - What is common practice for SIP user-agents to do in this case? /alfred [1] https://tools.ietf.org/html/rfc3263#section-4.4 [2] http://www.kamailio.org/docs/modules/3.4.x/modules/auth.html#auth.secret [3] https://github.com/alfredh/baresip/issues/39

4 4

Kamailio Updates on VUC 588 - this week on Friday
by Daniel-Constantin Mierla 08 Apr '16

08 Apr '16

Hello, The weekly VoIP Users Conference (VUC) #588 moderated by Randy Resnick will host a session for Kamailio Updates. Everyone from community is invited to join and actively participate to discussions. There are many ways to connect, from SIP to classic PSTN dial in as well as listening or watching live streams on the web. The live session will start on Friday, April 8, at 18:00 Berlin timezone (17:00 London, 12:00EDT). More details about the session are available at: - https://www.kamailio.org/w/2016/04/vuc-588-kamailio-updates/ Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com

1 2

Question about debugging using `kamcmd pkg.stats`
by Lucian Balaceanu 08 Apr '16

08 Apr '16

Hi Daniel, While debugging a 'no private memory left' issue by using `kamcmd pkg.stats` we stumbled upon data we don't understand. Could you help with some information? `kamcmd pkg.stats`returns free values of about 8M per process with an exception: kamailio with no apparent problems: `timer` process: free value: 8M used value: 100M real_used: 308M. kamailio with no private memory left: `timer` process: free value: 8/9M used value: 98M real_used: 662M (both mentioned kamailios have roughly the same load) Are these timer values normal? What is this real_used value and how can it be bigger than the allocated pkg mem? Additionally, how come we don't see the fifo process pkg stats in the output of `kamcmd pkg.stats`? We get the error message when doing reloading backlists via kamctl fifo. Thank you, Lucian Balaceanu

2 6

Compatibility between Siremis 4.2 and Kamailio 4.4
by Jay Li 08 Apr '16

08 Apr '16

Dear all, I got into the Kamailio world recently, and just wanted to get some info here about the compatibility between Siremis and Kamailio. From what I read, Siremis is wonderful web management tool for Kamailio. However the latest Siremis to date is 4.2.0 which was released more than a year ago. From the release note, it was intended to work with Kamailio 4.2, but Kamailio 4.4 has already released. I'm wondering if there's known compatibility issues between Siremis 4.2 and later released of Kamailio, like 4.3 and 4.4. Any information is appreciated. Regards,Jay

2 1

Insert callerid in Kamailio
by Nelson Migliaro 07 Apr '16

07 Apr '16

Hello, I am planning to insert the caller id in my Kamailio before sending the call to sip vendor. I am planning to send a prefix before the number and use that prefix in the SQL query to select the CLID. I was thinking in the sqlops to select the CLID and then append the result in the Remote-Party-ID: modparam("sqlops","sqlcon","cb=>mysql://kamailio:abc@10.10.1.1/testdb") modparam("sqlops","sqlcon","ca=>dbdriver://username:password@dbhost/dbname") ... ... sql_query("ca", "select clid from mitable where prefix = $var(prefix)", "ra"); xlog("number of rows in table domain: $dbr(ra=>rows)\n"); sql_result_free("ra"); Question..... ¿Is there an specific module to do this or a there a better way? Thank you Nelson.-

2 1

SIP Message Body
by Vladimer Gabunia 07 Apr '16

07 Apr '16

How can I Catch Text Message in SIP Method Message ? ________________________________ [gh.ge] ვლადიმერ გაბუნია IT სამსახურის უფროსი ტელ: (+995) 32 2505222 +8183 მობ: (995) 577 095333 შპს "ჯეო ჰოსპიტალს" სათავო ოფისი თბილისი 0160, ვაჟა-ფშაველას გამზ. № 16; http://www.gh.ge <http://gh.ge>

5 4

rtp proxy truncates sdp body
by Yasin CANER 06 Apr '16

06 Apr '16

1 0

SQLOPS select database in realtime
by Jon Bonilla (Manwe) 06 Apr '16

06 Apr '16

Hi all I have multiple auth credentials split among multiple databases (same server). I want to extract the password from the correct database for each request and pass it as arameter to the auth functions. The problem is, how can I use sqlops and select the correct database in realtime? the db connection already contains the database to connect to. Would it if I use a default connection but I put the db I want i the query? Something like: select * from database.table ... select * from $avp(s:database).table where i=1... cheers, Jon

3 2

pseudo variables in strings
by Juha Heinanen 05 Apr '16

05 Apr '16

If I have a string that contains pseudo variables, e.g,, $var(uri_pattern) = "sip:$fU@tutpro.com"; is it somehow possible to replace them with their values, something like: $var(uri) = eval("$var(uri_pattern)"); -- Juha

3 8

about rtpengine versions
by Juha Heinanen 05 Apr '16

05 Apr '16

I noticed that 15 days ago there was rtpengine commit https://github.com/sipwise/rtpengine/commit/aac8899b612dc8103b89f3f9c921f88… titled "Release new version 4.4.0.0+0~mr4.4.0.0", but I could not find the corresponding branch or tag. If someone wants to use that version, is 'git archive aac8899b612dc8103b89f3f9c921f88af3501303' the only way to do so? -- Juha

2 2

Re: [SR-Users] Ka--כרmailio asאעא ער מאיכקטח- a presence server
by Shlomi Gutman 05 Apr '16

05 Apr '16

אני אה כן בתאריך 4 באפר׳ 2016 8:45 לפנה״צ,‏ "Daniel-Constantin Mierla" < miconda(a)gmail.com> כתב: > Hello, > > can you run with debug=3 in config file and see if you get some hints from > syslog messages about what happens ? > > Cheers, > Daniel > > On 01/04/16 17:17, Igor Olhovskiy wrote: > > Hi! > I want to create custom Presence server. Main idea - phones subscribes to > it (for ex, extension 100@domain). This subscription actually lit BLF > button on a phone. > But main idea - change status of BLF can arrive from different locations > via PUBLISH, I assume. > Problem - phones has different type of SUBSCRIBE, like > dialog(application/dialog-info+xml) or presence(application/pidf+xml). Some > of phones are not fully follow RFC (Like using in SUBSCRIBE R-URI not > domain, but IP, but in From or To - domains are used) > Phones are registered elsewhere, not on Kamailio. After reading RFC > (especially event:dialog part) I’ve found about SIP-If-Match part, but > problem about sync this data across multiple PUBLISH locations. > Can be Kamailio be a presence server in this scenario? > > For ex, SUBSCRIBE > > SUBSCRIBE sip:212.232.26.232:5060;transport=udp SIP/2.0 > Via: SIP/2.0/UDP 192.168.88.60:5060;branch=z9hG4bK1656003104;rport > From: <sip:608@master.rufan.at>;tag=521811605 > To: <sip:*55@master.rufan.at>;tag=05ea6038656678bd0198b8977f3c0221.4e93 > Call-ID: 805970257-5060-2(a)BJC.BGI.II.GA > CSeq: 20202 SUBSCRIBE > Contact: <sip:608@192.168.88.60:5060> > X-Grandstream-PBX: true > Max-Forwards: 70 > User-Agent: Grandstream GXP2160 1.0.5.33 > Expires: 180 > Supported: replaces, path, timer, eventlist > Event: dialog > Accept: application/dialog-info+xml,multipart/related,application/rlmi+xml > Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, > UPDATE, MESSAGE > Content-Length: 0 > > > I’ve tried send to Kamailio this PUBLISH > > PUBLISH sip:*55@master.rufan.at:5060 SIP/2.0 > Via: SIP/2.0/UDP 10.0.20.71:5060;branch=z9hG4bK-6930-1-0 > Max-Forwards: 70 > To: <sip:*55@master.rufan.at> > From: <sip:*55@master.rufan.at:5060>;tag=1 > Call-ID: 1-6930(a)127.0.1.1 > CSeq: 1 PUBLISH > Contact: <sip:mod_sofia@127.0.1.1:5060> > Event: presence > Expires: 3600 > Content-Type: application/pidf+xml > Content-Length: 486 > > <?xml version="1.0" encoding="ISO-8859-1"?> > <presence xmlns='urn:ietf:params:xml:ns:pidf' > xmlns:dm='urn:ietf:params:xml:ns:pidf:data-model' > xmlns:rpid='urn:ietf:params:xml:ns:pidf:rpid' > xmlns:c='urn:ietf:params:xml:ns:pidf:cipid' entity='sip:* > 55(a)master.rufan.at'> > <tuple id='z9hG4bK-6930-1-0'> > <status> > <basic>open</basic> > </status> > </tuple> > <dm:person id='1'> > <rpid:activities> > <rpid:on-the-phone/> > </rpid:activities> > <dm:note>Night</dm:note> > </dm:person> > </presence> > > Kamailio answers with 200OK, but after this - no NOTIFY to subscribed > clients. > > -- > Best regards, > Igor > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > -- > Daniel-Constantin Mierlahttp://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > >

1 0

Kamailio as a presence server
by Igor Olhovskiy 05 Apr '16

05 Apr '16

Hi! I want to create custom Presence server. Main idea - phones subscribes to it (for ex, extension 100@domain). This subscription actually lit BLF button on a phone. But main idea - change status of BLF can arrive from different locations via PUBLISH, I assume. Problem - phones has different type of SUBSCRIBE, like dialog(application/dialog-info+xml) or presence(application/pidf+xml). Some of phones are not fully follow RFC (Like using in SUBSCRIBE R-URI not domain, but IP, but in From or To - domains are used) Phones are registered elsewhere, not on Kamailio. After reading RFC (especially event:dialog part) I’ve found about SIP-If-Match part, but problem about sync this data across multiple PUBLISH locations. Can be Kamailio be a presence server in this scenario? For ex, SUBSCRIBE SUBSCRIBE sip:212.232.26.232:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.88.60:5060;branch=z9hG4bK1656003104;rport From: <sip:608@master.rufan.at>;tag=521811605 To: <sip:*55@master.rufan.at>;tag=05ea6038656678bd0198b8977f3c0221.4e93 Call-ID: 805970257-5060-2(a)BJC.BGI.II.GA CSeq: 20202 SUBSCRIBE Contact: <sip:608@192.168.88.60:5060> X-Grandstream-PBX: true Max-Forwards: 70 User-Agent: Grandstream GXP2160 1.0.5.33 Expires: 180 Supported: replaces, path, timer, eventlist Event: dialog Accept: application/dialog-info+xml,multipart/related,application/rlmi+xml Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 I’ve tried send to Kamailio this PUBLISH PUBLISH sip:*55@master.rufan.at:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.20.71:5060;branch=z9hG4bK-6930-1-0 Max-Forwards: 70 To: <sip:*55@master.rufan.at> From: <sip:*55@master.rufan.at:5060>;tag=1 Call-ID: 1-6930(a)127.0.1.1 CSeq: 1 PUBLISH Contact: <sip:mod_sofia@127.0.1.1:5060> Event: presence Expires: 3600 Content-Type: application/pidf+xml Content-Length: 486 <?xml version="1.0" encoding="ISO-8859-1"?> <presence xmlns='urn:ietf:params:xml:ns:pidf' xmlns:dm='urn:ietf:params:xml:ns:pidf:data-model' xmlns:rpid='urn:ietf:params:xml:ns:pidf:rpid' xmlns:c='urn:ietf:params:xml:ns:pidf:cipid' entity='sip:*55@master.rufan.at '> <tuple id='z9hG4bK-6930-1-0'> <status> <basic>open</basic> </status> </tuple> <dm:person id='1'> <rpid:activities> <rpid:on-the-phone/> </rpid:activities> <dm:note>Night</dm:note> </dm:person> </presence> Kamailio answers with 200OK, but after this - no NOTIFY to subscribed clients. -- Best regards, Igor

3 4

dispatcher [dispatch.c:1416]: ds_load_remove(): cannot find load for (from asterisk call-id)
by TEG AMJG 04 Apr '16

04 Apr '16

Hi I am new to Kamailio and i am just trying to setup a kamailio (v4.2.3) dispatcher to distribute calls to 2 asterisk using the call load distribution algorithm (alg 10). I started to test and see how all it is working and in the logs appears that the call load record it cannot be found. - ERROR: dispatcher [dispatch.c:1416]: ds_load_remove(): cannot find load for (call id generated by asterisk) I am getting this error trying a 2 call leg conversation when asterisk is creating the BYE message destinated to the callee after receiving a BYE message from the caller, and the BYE message by Asterisk reach Kamailio to then apply the ds_load_update function >From what i understand about ds_load_update it is that it sets the load to an specific destination if Kamailio receives a 2xx reply of an INVITE message, and removes it if Kamailio receives a BYE message based from the Call-id received in that message, where it is actually failing in my implementation at the moment Here are the blocks of code of the dispatcher parameters and where the function ds_load_update() is being used: #!ifdef WITH_LOADBALANCE #Parameters using Dispatcher from DB # ---------- modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "table_name", "dispatcher") # ---------- #Enabling failover mechanism # ----------- modparam("dispatcher", "flags", 2) # ----------- #Setting dispatcher parameters in AVP # ----------- modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") modparam("dispatcher", "dstid_avp", "$avp(dsdstid)") # ----------- #Setting dispatcher result in PV variables # ----------- modparam("dispatcher", "attrs_pvname", "$var(attrs)") # ----------- #Enabling dispatcher algorithm according to call loads # ----------- modparam("dispatcher", "ds_hash_size",8) modparam("dispatcher", "ds_hash_expire", 3600) modparam("dispatcher", "ds_hash_initexpire", 60) # ----------- # Setting parameters for probing dispatcher destinations # ----------- modparam("dispatcher", "ds_ping_interval", 20) modparam("dispatcher", "ds_ping_from", "sip:kamailio1@<Kamailio IP>") modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_probing_threshhold", 2) modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=480;code=404") # ----------- ... route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing #!ifdef WITH_LOADBALANCE if(is_method("BYE|CANCEL")){ xlog("L_ERROR","UBICAR BYE ds_load_update requesturi=$ru from=$fu method=$rm callid=$ci to=$tu reply=$mt rrcode=$rr rcode=$rs\n"); ds_load_update(); } #!endif ... } ... onreply_route[MANAGE_REPLY] { #!ifdef WITH_LOADBALANCE if(is_method("INVITE")) { if(status=~"2[0-9][0-9]") { xlog("L_ERROR","UBICAR setting ds_load_update requesturi=$ru from=$fu method=$rm callid=$ci to=$tu reply=$mt rrcode=$rr rcode=$rs"); ds_load_update(); }else if(status=~"[3-7][0-9][0-9]") { ds_load_unset(); } } #!endif if(status=~"[12][0-9][0-9]"){ route(NATMANAGE); } } This is what i am getting from logs after the 2 call legs conversation completes: kam1 /usr/sbin/kamailio[4082]: ERROR: <script>: UBICAR: FROMASTERISK 70175143146f92975a5bff8876435911(a)192.168.10.194 kam1 /usr/sbin/kamailio[4082]: ERROR: <script>: UBICAR setting ds_load_update requesturi=<null> from=sip:103@192.168.10.194 method=INVITE callid=28d0f6e71625c9e87baec07f46167863(a)192.168.10.194 to= sip:104@192.168.10.194 reply=2 rrcode=OK rcode=200 kam1 /usr/sbin/kamailio[4083]: ERROR: <script>: UBICAR setting ds_load_update requesturi=<null> from=sip:103@192.168.10.194:5060 method=INVITE callid=1-3692(a)192.168.20.2 to=sip:104@192.168.10.194:5060 reply=2 rrcode=OK rcode=200 kam1 /usr/sbin/kamailio[4084]: ERROR: <script>: UBICAR BYE ds_load_update requesturi=sip:104@192.168.10.190:5060 from=sip:103@192.168.30.1:5060 method=BYE callid=1-3692(a)192.168.20.2 to=sip:104@192.168.30.1:5060 reply=1 rrcode=<null> rcode=<null> kam1 /usr/sbin/kamailio[4085]: ERROR: <script>: UBICAR BYE ds_load_update requesturi=sip:104@192.168.30.2:22523;transport=UDP from= sip:103@192.168.10.194 method=BYE callid= 28d0f6e71625c9e87baec07f46167863(a)192.168.10.194 to=sip:104@192.168.10.194 reply=1 rrcode=<null> rcode=<null> kam1 /usr/sbin/kamailio[4085]: ERROR: dispatcher [dispatch.c:1416]: ds_load_remove(): cannot find load for ( 28d0f6e71625c9e87baec07f46167863(a)192.168.10.194) Kamailio IP: 192.168.10.194 Asterisk IP: +----+-------+-------------------------+-------+----------+-------------------------------+-------------+ | id | setid | destination | flags | priority | attrs | description | +----+-------+-------------------------+-------+----------+-------------------------------+-------------+ | 1 | 0 | sip:192.168.10.190:5060 | 2 | 0 | duid=PBX1;my=pbx1;maxload=200 | pbx1 | | 2 | 0 | sip:192.168.10.191:5060 | 2 | 0 | duid=PBX2;my=pbx2;maxload=200 | pbx2 | +----+-------+-------------------------+-------+----------+-------------------------------+-------------+

3 4

get_redirects with t_next_contacts
by Marrold 04 Apr '16

04 Apr '16

Hi all, I am having some issues using get_redirects() from the UAC_Redirect module in conjunction with t_next_contacts from the TM Module. I've searched Google and the list but the only references I can find are several years old. I am attempting to configure Kamailio so it if receives a 3XX redirect message, it will store the Contacts it contains, and attempt them in 'Q' parameter order. So for the Contact below- Contact: <sip:10.0.97.21:5060>;q=0.2,<sip:5.29.13.139:5060>;q=0.5. I want Kamailio to try 5.29.13.139, and if that times out, or is rejected in someway, try 10.0.97.21. If I use t_relay rather than t_next_contacts, kamailio branches the calls and relays them in parallel regardless of Q value - this is not what I want. If I use t_next_contacts, it attempts 5.29.13.139 which is rejected with a 401, but it doesn't attempt 5.29.13.139 and responds "408", "Request Timeout MH 2" Relevant script section is below, has anyone been able to get this working? failure_route[302] { if (t_check_status("302")){ xlog("L_ALERT","Failure Route: 302\n$mb"); get_redirects("*"); if (!t_load_contacts()) { send_reply("500", "Server Internal Error - Cannot load contacts MH"); exit; }; if (!t_next_contacts()) { t_reply("408", "Request Timeout MH"); } else { t_on_failure("302"); t_relay(); exit; }; } else if (t_check_status("[456][0-9][0-9]")){ if (!t_next_contacts()) { t_reply("408", "Request Timeout MH 2"); } else { t_on_failure("302"); t_relay(); exit; } } } Thanks, Matthew

1 2

Using 302 redirects with Dispatcher Module
by Marrold 04 Apr '16

04 Apr '16

Hi All, I'd like to use the Dispatcher Module along with 30X redirects, to monitor gateway availability with SIP options and to provide redundancy / fail over. Some sources online suggest it's as easy as - if ( method=="INVITE" ) { dst_select_domain("1","4"); sl_send_reply("300","Redirect"); exit; } However when the 30X reply is sent, it contains no contact header. Please could someone advise if it's possible to use the Dispatcher module with 302 redirects, and what they syntax is? Thanks, Matthew

3 6

kamailio number translation and route
by liwei＠neulinx.com 04 Apr '16

04 Apr '16

hi all; I want to use kamailio achieve: First phase: 1) inbound caller number translation; 2) outbound caller number translation; 3) inbound callee number translation; 4) outbound callee number tanslation; Example : Call Flow backend server ------------------> kamailio --------------------------> GateWay ------------------->PSTN caller -->callee caller ---> callee 1000 --> 2000 111000--->222000 inbound outbound caller --->callee caller ---> callee 1000 ---> 2000 11000 ---> 22000 | translation | | translation | 11000 ---> 22000 111000---> 222000 Second phase: / ---->gateway1 ----->pstn / backend server ---> kamailio ----->gateway2 -----> pstn \ \ ----> gateway3 ------->pstn Could you give me some suggest or example configuration？ liwei(a)neulinx.com

2 1

Load balancing traffic based on SIP URI
by NITESH BANSAL 04 Apr '16

04 Apr '16

Hello, I want to use Kamailio to load balance traffic across multiple asterisks.The problem is that my SIP traffic is related to conference, so if a SIP callfor a particular URI is sent to an Asterisk instance, any subsequent calls for thatSIP URI should be sent to the same Asterisk instance.Is there anyway to achieve this in Kamailio? Thanks,Nitesh

3 5

SIP Authentication
by Girish 04 Apr '16

04 Apr '16

Hello Sir, I am a newbie to Kamailio and I am trying my first setup. I would like to use Kamailio to take in traffic using a SIP Trunk and Route the calls to another SIP trunk based on prefix. I have added the incoming SIP trunk IP address to the address table on the MYSQL but I am unable to get a call routed out. Is there any sample configuration for Authentication via IP address and route the incoming SIP INVITE. Regards, Phoenix

2 1

RPID and Auth DB
by Alberto Sagredo 04 Apr '16

04 Apr '16

Hi! I plan to use rpid field but when trying to get from database, subscriber table has rpid field filled, i get always NULL. Is there any more needed than: modparam("auth_db", "load_credentials", "rpid") ? Just a check of variable xlog(" Valor de RPID es : $avp(s:rpid)"); Shows me NULL all times Thanks

2 1

KAMAILIO tcp_send query, connection problem
by Ed Todd 04 Apr '16

04 Apr '16

KAMAILIO tcp_send query, connection problem I have a KAMAILIO SIP server on a static IP from Comcast, using a modem/router Cisco DPC39412B. I have a PC application server which is successfully signing on to the SIP server, via a LAN cable on the same network as the SIP server (Cisco). In my second test, the PC application server is connected to the internet via a ZTE Falcon Z-917. In a third test, the PC application server is connected to the internet via a Talk-Talk fiber broadband. The connection does not occur. >From the log files, the successful connection says: Apr 1 10:44:30 SIP-Server kamailio[7817]: DEBUG: <core> [tcp_main.c:2281]: tcpconn_send_put(): tcp_send: send from reader (7817 (14)), reusing fd Apr 1 10:44:30 SIP-Server kamailio[7817]: DEBUG: <core> [tcp_main.c:2516]: tcpconn_do_send(): tcp_send: sending... Apr 1 10:44:30 SIP-Server kamailio[7817]: DEBUG: <core> [tcp_main.c:2550]: tcpconn_do_send(): tcp_send: after real write: c= 0x7f6270b8a4a0 n=565 fd=1 On the failed connection: Apr 1 09:41:23 SIP-Server kamailio[7816]: DEBUG: <core> [tcp_main.c:1818]: tcp_send(): tcp_send: no open tcp connection found, opening new one Apr 1 09:41:23 SIP-Server kamailio[7816]: DEBUG: <core> [ip_addr.c:243]: print_ip(): tcpconn_new: new tcp connection: 192.168.0.146 Apr 1 09:41:23 SIP-Server kamailio[7816]: DEBUG: <core> [tcp_main.c:1073]: tcpconn_new(): tcpconn_new: on port 55133, type 3 Apr 1 09:41:23 SIP-Server kamailio[7816]: DEBUG: <core> [tcp_main.c:1382]: tcpconn_add(): tcpconn_add: hashes: 1776:802:0, 803 Apr 1 09:41:23 SIP-Server kamailio[7816]: DEBUG: tls [tls_server.c:184]: tls_complete_init(): Using TLS domain TLSc<default A while later, I get: Apr 1 09:41:37 SIP-Server kamailio[7804]: DEBUG: usrloc [urecord.c:357]: wb_timer(): Binding 'hvs_us_iphone','sip:hvs_us_iphone@172.56.8.182:44341;transport=TLS;line=75dc89bd64369c7' has expired Apr 1 09:41:38 SIP-Server kamailio[7820]: DEBUG: <core> [tcp_main.c:4320]: tcpconn_main_timeout(): tcp_main: entering timer for 0x7f6270b8a4a0 (ticks=862851529, timeout=862909176 (3602 s), wr_timeout=862851529 (0 s)), write queue: 197 bytes Apr 1 09:41:38 SIP-Server kamailio[7820]: ERROR: <core> [tcp_main.c:4338]: tcpconn_main_timeout(): connect 192.168.0.146:55133 failed (timeout) Apr 1 09:41:38 SIP-Server kamailio[7820]: DEBUG: <core> [tcp_main.c:4362]: tcpconn_main_timeout(): tcp_main: timeout for 0x7f6270b8a4a Any clues or tips would be greatly appreciated! Ed

2 1

Re: [SR-Users] Multiple Registrations - Overwrite oldest contact
by Bruno Emer 03 Apr '16

03 Apr '16

Sure it is! This is exactly what I was looking for! Actually I wasn't able to see that there was a way to perform this action! Thanks for your help, Olle! I will try to create this, and when I have it done, I will post my script here, so if someone else need the same problem I can help. Now, I have just one more question about my scenario: when I register a user, is there a way to create something like a "custom field" on the location? To explain better: If possible, I can add something like a custom header that says if the user is registered from a phone device or the web interface, then I can perform the loop and replace only the oldest contact using that device... *Bruno Emer* Mobile: +55 11 96540-0044 email: brunoemer(a)gmail.com 2016-04-01 17:18 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: > This is my problem... I want to have 2 contacts for AOR, and not only one. > > When the 3rd arrives, it must remove the oldest and continue with 2... > > *If I am using the 0x04 parameter, it will save just one contact, and the > user will not be able to get the Invite both, in the mobile device and web > interface.* > > *Bruno Emer* > > Mobile: +55 11 96540-0044 > email: brunoemer(a)gmail.com > <https://www.facebook.com/bruno.emer.5> > <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> > <https://twitter.com/brunoemer_> > <https://www.flickr.com/photos/122070309@N03/> > <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_> > > 2016-04-01 17:07 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: > >> Hello, *Olle!* >> >> *Thanks for your help in this case, and don't worry about the time >> (actually I wasn't even expecting to get an answer today).* >> >> >> *I know the max_contacts parameter. But the problem is that if I set the >> max_contacts to 2, when the user tries to register again, he gets an error >> message 503. The point is that I want to get the user registered, and >> replace the oldest location entry for the newest one.* >> >> >> >> *Bruno Emer* >> >> Mobile: +55 11 96540-0044 >> email: brunoemer(a)gmail.com >> >> 2016-04-01 16:31 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: >> >>> Hello all. >>> >>> I have problem here and I tried to find a solution and search over >>> internet, but without success. >>> >>> My scenario is the following: I have an application that must be >>> registered in Kamailio when a user logs in the web interface, so he can get >>> calls (something like a web softphone using webrtc). At this point, we are >>> OK, and everything is working fine. >>> >>> To get these register functions I am using the parameter >>> "save("location", "0x04")" as described in the REGISTRAR module >>> documentation, so if a user logs in another web browser or computer, only >>> the last one will continue registered and all calls will be forwarded to >>> him. >>> >>> The point is that now we are creating a phone app that will do almost >>> the same thing as the web interface, allowing users to receive calls using >>> the mobile device, and here is my problem: I want to allow my users to be >>> registered on two devices at the same time, but if a user logs into another >>> device, I don't want to reply with a 503. I want to allow the user to >>> register again, deregistering the oldest contact. >>> >>> I saw that there is a module named "ims_usrloc_scscf" and on its >>> description it says "implemented overwrite oldest contact behaviour", >>> but I couldn't find any documentation about it. >>> >>> So, is there a way to get this working today? >>> >>> >>> *Bruno Emer* >>> >>> Mobile: +55 11 96540-0044 >>> email: brunoemer(a)gmail.com >>> >> >> >

3 2

kamailio proxy
by liwei＠neulinx.com 03 Apr '16

03 Apr '16

hi ， I want to use kamailio as a proxy ; / gateway1 ----->pstn / 1、 call from backend server ---> kamailio ----->gateway2 -----> pstn \ \ gateway3 ------->pstn 2、caller and callee number translation ; Could you please give me some suggest; Wade From China liwei(a)neulinx.com

2 1

SQlite DB Lock with Presence
by Kristian Kielhofner 02 Apr '16

02 Apr '16

Hello everyone, I'm trying to use BLF with the config here (except with SQlite): http://kb.asipto.com/kamailio:presence:k43-blf BLF seems to work perfectly except after the call is ended I get "sqlite commit failed: database is locked" error messages as seen here: http://pastebin.com/xVijj98H BLF is then broken at this point. kamailio -v version: kamailio 4.3.5 (x86_64/linux) 950657 flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_PTHREAD_MUTEX, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 950657 compiled on 01:33:01 Mar 29 2016 with x86_64-openwrt-linux-gnu-gcc 4.8.3 Has anyone ever seen this before? Thanks! -- Kristian Kielhofner

3 9

MSRP question
by Hussein Said 02 Apr '16

02 Apr '16

2 4

kamailio proxy
by liwei＠neulinx.com 02 Apr '16

02 Apr '16

hi ， I want to use kamailio as a proxy ; / gateway1 ----->pstn / 1、 call from backend server ---> kamailio ----->gateway2 -----> pstn \ \ gateway3 ------->pstn 2、caller and callee number translation ; Could you please give me some suggest; Wade From China liwei(a)neulinx.com

1 0

Trouble initializing TLS on Fedora 23
by Anthony Messina 02 Apr '16

02 Apr '16

Happy New Year! I've just upgraded my Kamailio (build from master@c7e411e) instance to Fedora 23 from Fedora 22. I've built the packages specifically for Fedora 23 with the following current sources: openssl-1.0.2e-3.fc23.x86_64 openssl-libs-1.0.2e-3.fc23.x86_64 openssl version reports... OpenSSL 1.0.2e-fips 3 Dec 2015 Even so, the following error occurs. It seems like Kamailio having trouble detecting that I'm using running with the same version that I have installed, and the same version that I have compiled against. tls [tls_init.c:557]: init_tls_h(): ERROR: tls: init_tls_h: installed openssl library version is too different from the library the Kamailio tls module was compiled with: installed "OpenSSL 1.0.0-fips 29 Mar 2010" (0x10000003), compiled "OpenSSL 1.0.2d-fips 9 Jul 2015" (0x1000204f). Please make sure a compatible version is used (tls_force_run in kamailio.cfg will override this check) CRITICAL: <core> [main.c:2558]: main(): could not initialize tls, exiting... Any pointers? -A -- Anthony - https://messinet.com/ - https://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E

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Re: [SR-Users] Multiple Registrations - Overwrite oldest contact
by Bruno Emer 01 Apr '16

01 Apr '16

Hello, *Olle!* *Thanks for your help in this case, and don't worry about the time (actually I wasn't even expecting to get an answer today).* *I know the max_contacts parameter. But the problem is that if I set the max_contacts to 2, when the user tries to register again, he gets an error message 503. The point is that I want to get the user registered, and replace the oldest location entry for the newest one.* *Bruno Emer* Mobile: +55 11 96540-0044 email: brunoemer(a)gmail.com 2016-04-01 16:31 GMT-03:00 Bruno Emer <brunoemer(a)gmail.com>: > Hello all. > > I have problem here and I tried to find a solution and search over > internet, but without success. > > My scenario is the following: I have an application that must be > registered in Kamailio when a user logs in the web interface, so he can get > calls (something like a web softphone using webrtc). At this point, we are > OK, and everything is working fine. > > To get these register functions I am using the parameter "save("location", > "0x04")" as described in the REGISTRAR module documentation, so if a user > logs in another web browser or computer, only the last one will continue > registered and all calls will be forwarded to him. > > The point is that now we are creating a phone app that will do almost the > same thing as the web interface, allowing users to receive calls using the > mobile device, and here is my problem: I want to allow my users to be > registered on two devices at the same time, but if a user logs into another > device, I don't want to reply with a 503. I want to allow the user to > register again, deregistering the oldest contact. > > I saw that there is a module named "ims_usrloc_scscf" and on its > description it says "implemented overwrite oldest contact behaviour", but > I couldn't find any documentation about it. > > So, is there a way to get this working today? > > > *Bruno Emer* > > Mobile: +55 11 96540-0044 > email: brunoemer(a)gmail.com >

2 3

Multiple Registrations - Overwrite oldest contact
by Bruno Emer 01 Apr '16

01 Apr '16

Hello all. I have problem here and I tried to find a solution and search over internet, but without success. My scenario is the following: I have an application that must be registered in Kamailio when a user logs in the web interface, so he can get calls (something like a web softphone using webrtc). At this point, we are OK, and everything is working fine. To get these register functions I am using the parameter "save("location", "0x04")" as described in the REGISTRAR module documentation, so if a user logs in another web browser or computer, only the last one will continue registered and all calls will be forwarded to him. The point is that now we are creating a phone app that will do almost the same thing as the web interface, allowing users to receive calls using the mobile device, and here is my problem: I want to allow my users to be registered on two devices at the same time, but if a user logs into another device, I don't want to reply with a 503. I want to allow the user to register again, deregistering the oldest contact. I saw that there is a module named "ims_usrloc_scscf" and on its description it says "implemented overwrite oldest contact behaviour", but I couldn't find any documentation about it. So, is there a way to get this working today? *Bruno Emer* Mobile: +55 11 96540-0044 email: brunoemer(a)gmail.com

2 2

Make Kamailio Great Again!
by Alex Balashov 01 Apr '16

01 Apr '16

For immediate release: ATLANTA, GA (1 April 2016)--Alex J. Balashov, a self-styled businessman based in Atlanta, Georgia, USA, has a plan to "Make Kamailio Great Again". "Evariste Systems is huge. My name is on the building," said Balashov of his iconic VoIP consulting brand. "And you know what, I have been very successful. Everybody loves me." Balashov has capitalised on a contentious election cycle marked by deep political polarisation, growing income inequality and geopolitical challenges such as global terrorism. And his sharp message of alarm about the declining influence of the Kamailio SIP server project has resonated with increasing numbers in the CxO suite, vaulting him to the lead in the race for the IETF SIP Working Group nomination, according to recent polls of primary voters. He has been quick to tout his competitive credentials in a tough global open-source ecosystem. At a recent colloqium on unified communications, he asked: "When was the last time anybody saw us beating, let's say, OpenSIPS in Git commits? They kill us. I beat OpenSIPS all the time. All the time." As Balashov sees it, a major cause of the beleaguered Kamailio project's woes lies in its liberal patch acceptance policy and lax scrutiny of third-party contributions: "When GitHub sends its people, they're not sending their best. They're not sending you. They're not sending you. They're sending people that have lots of problems, and they're bringing those problems. They're bringing drugs. They're bringing crime. They're rapists. And some, I assume, are good people." He has proposed a controversial solution that has drawn ire from liberal ranks in the open-source community, but has also attracted applause and standing ovations at his speaking engagements: "We have to have a firewall around the Kamailio source code. We have to have an access control list. And in that firewall, we're going to have a big fat door where commits and pull requests can come into the master branch, but they have to come in legally. The firewall will go up, and GitHub will start behaving." Balashov's firewall proposal has been met with scorn from critics who deride it as impractical and quixotic. In particular, commentators have raised questions about funding and resources as well as GitHub's willingness to entertain a boundary around a project in its vicinity. Balashov isn't concerned, however: "I will build a great firewall--and nobody builds firewalls better than me, believe me--and I'll build them very inexpensively. I will build a great, great stateful packet inspection wall on our border with GitHub, and I will make GitHub pay for that wall. Mark my words." He has also been rebuked by rival IETF leadership candidates for his often acerbic Twitter remarks directed at Lennart Poettering and the developers of "firewalld". As he sees it, however, the network effects of social media are a strength: "My Twitter has become so powerful that I can actually make my enemies tell the truth." He scoffed at the suggestion that his characterisations of industry actors behind the RedHat-led "systemd" movement are misleading: "RedHat was the worst Steward of Linux in the history of the kernel. There has never been a Steward so bad as RedHat. The source code blew up around us. We lost everything, including all synergies. There wasn't one good thing that came out of that administration or them being Stewards of Linux." Balashov's idiosyncratic campaign is not standing still. He has proven to be a capable populist, adapting rapidly to an evolving sense of the kinds of pronouncements that activate his swelling crowds of devotees. Along the way, he has deftly deflected calls to subject his policy proposals to expert review. "I know what I'm doing, and I listen to a lot of people, I talk to a lot of people, and at the appropriate time I'll tell you who the people are. But I speak to a lot of people, but my primary consultant is myself, and I have a good instinct for this stuff." At a recent gathering of SIP stack interoperability specialists, Balashov the latest pillar of his platform to "Make Kamailio Great Again", in view of growing security vulnerabilities in the latest Kamailio modules: "Alex J. Balashov is calling for a total and complete shutdown of commits entering the master branch from the territory of the European Union until our project's representatives can figure out what's going on. According to Netcraft, among others, there are a lot of buffer overflows in Kamailio by large segments of the EU population."

2 1

Acc INVITE dialog start timing difference
by Dirk Teurlings - Signet B.V. 01 Apr '16

01 Apr '16

Hi, I'm seeing a difference in timestamp for acc modules recorderd INVITES and the dialog start event. This is the log entry: Apr 1 12:55:59 host /usr/sbin/kamailio[15642]: INFO: <script>: 5d3ff3407b9761b24c4f75195c6ccb1d@192.168.10.245:5060 - dialog:start And this has been added by ACC in the database: INVITE as2880319e 56FE53B9-49E3774-3E86ED1B 5d3ff3407b9761b24c4f75195c6ccb1d@192.168.10.245:50... 200 OK 2016-04-01 12:56:01 Why are these two seconds apart? Shouldn't the acc module record the same dialog start event time as the start of a dialog? When I look at the value of DLG_lifetime this confirms my findings, as it results in: dialog_time = TS - dlg_start When looking at the database BYE-time - INVITE time, the call is two seconds shorter. Any insight would be appreciated! Cheers, Dirk

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