- sr-users - kamailio.org

I think I found issue to Kamailio not forwarding ACK sent to it by Asterisk
by Wilkins, Steve 22 May '18

22 May '18

Hello All, When Kamailio sends a '200 OK' to Asterisk, it is putting its Public IP into the Record-Routes. In Asterisk 14.6, it would send the 'ACK' back to the Private IP Address of Kamailio, but Asterisk 15.x is using the Public IP Address that Kamailio placed in the Record-Routes so... Is there a way to force Kamailio to set the Record-Route to its Private IP address of the first Record-Route in messages forwarded to Asterisk? Thank you?

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rtpp_test() in kamailio log
by Giovanni Tommasini - evosip 22 May '18

22 May '18

Hi guys, I've a question on using rtpengine with kamailio *scenario*: kamailio istance with rtpengine enabled and configured in localhost (but with a remote rtpengine is the same) modparam("rtpengine", "rtpengine_sock","udp:localhost:22222") kamailio version 4.4.6 + rtpengine version 6.3.0.0 running on Ubuntu 16.04 when I start kamailio I see a lot of INFO log rows like that May 22 12:13:12 my-dev /usr/sbin/kamailio[3388]: INFO: rtpengine [rtpengine.c:2546]: rtpp_test(): rtp proxy <udp:localhost:22222> found, support for it enabled May 22 12:13:12 my-dev /usr/sbin/kamailio[3397]: INFO: rtpengine [rtpengine.c:2546]: rtpp_test(): rtp proxy <udp:localhost:22222> found, support for it enabled ... more to the point. I see always 24 rows for each rtpengine_sock configured i n rtpengine.c I find the rtpp_tes function and that's fine, but why 24 times? thanks a lot -- Giovanni Tommasini | evosip web: evosip.cloud

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Kamailio sends SIP messages to old websocket after reconnection during call
by Idris AVCI 22 May '18

22 May '18

Hi, We use kamailio as a backend for JSSIP based WebRTC UAs. We use SIP over Weboscket as transport and everythings work fine generally. Some websocket connections to kamailio drop during an active call and JSSIP reconnects immediately. New websocket connection has a different port than the dropped one but kamailio tries to send to old websocket port which is not active anymore. We use GRUU and configured all the related parameters in registrar, dialog, tm, websocket, tls, xhttp and userloc modules both for versions 4.4 and 5.1 but couldn't find a solution yet. For a typical call below are the ordered events: 1. UA register to kamailio over wss (Code: WS1) 2. Kamailio send an INVITE to the UA (Code: Call1) 3. UA answer and media starts 4. UA drops WS1 to kamailio and reconnect (Code: WS2) 5. UA has an active websocket (WS2) to kamailio 6. Kamailio tries to send BYE message of Call1 to WS1 which is no longer active and complains that websocket could not be found 7. Kamailio send an INVTE to the UA (Code: Call2) over WS2 8. UA answer and media starts 9. Kamailio send BYE message of Call2 to UA over WS2 and everything work as expected (No websocket drop) Error log: WARNING: <core> [msg_translator.c:2761]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found I think we need a mechanism that will update the dialog information of Call1 to use WS2 after reconnection. Any help greatly appreciated. Thanks.

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htable - unable to update entry when event_route[htable:expired:<table>] is executed
by David Heath 22 May '18

22 May '18

Hi, Looking for some help/ advice on updating the expiration timer/ values in a hash table when when [htable:expired:<table>] occurs. On startup my script successfully reads in a set of entries from a DB; I also know from testing/ tracing that I can add new entries to the hash table in request_route. My issue occurs when the expiration timer kicks in - my goal was to re-read from the DB to ensure either the entry still exists or hasn’t changed and either delete, update or refresh the value until the next time the expiration event occurs. This way entries in the DB could be modified overtime. At the moment all that’s occurring is that in the event_route[htable:expired:cli] block the entries are being drained one-by-one. I re-read the row from the from the DB and I reset the value using $sht(cli=>$shtrecord(key)) = $xavp(cli=>MAPPED_CLI); I’ve also tried updating the expiration timer against the key. I’ve tried both $shtex(cli=>$shtrecord(key)) = 60 and $shtex(cli=>$shtrecord(key)) = $TS+60 [the latter based on the values shown from sht_print()] but the entries still get deleted. I’ve tried this against both 5.1.0 & 5.1.2. Any help appreciated. Thanks David Heath

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Is it possible to change a positive (2xx) response to a negative one (4xx) during kamailio processing of a reply (g711 fallback drama)?
by George Diamantopoulos 21 May '18

21 May '18

Hello all, I've been hitting a wall in getting T.38 fallback to G.711 to work in a specific case, and I was wondering if I could use kamailio to the rescue, since all my other endeavours have failed miserably. In short, this is this the scenario where I need to have this manipulation happen: - During an established dialog between a SIP endpoint and asterisk (with kamailio in the middle acting as proxy), the latter (asterisk) sends a re-INVITE with a single offer for m=image with T.38 in the SDP. - The other (eccentric) endpoint (yes, it's a Cisco) replies with 200 OK, but will reject the T.38 offer in the SDP body instead by including a port number of '0' - Instead of falling back to G.711 as configured, asterisk in that case will do nothing, and the dialog will eventually end with a BYE (don't remember which of the two endpoints will send that, but it doesn't matter much) The thing here is that asterisk *WILL* fallback to G711 with a second re-INVITE, if the other endpoint replies to the first T.38 re-INVITE with a negative response (488) instead of a positive one (200) declining the media in the SDP body. I've looked at using freeswitch for a fax gateway instead, but apparently FS doesn't support fallback at all even in the 488 reply case, much less in the inactive media stream in 200 OK corner case I'm investigating. So, I thought I'd do that in kamailio and convert those 200 OKs to 488s with change_reply_status() from the textopsx module. Of course, the universe hates me, and this will fail spectacularly: > ERROR: textopsx [textopsx.c:301]: ki_change_reply_status(): the class of provisional or positive final replies cannot be changed So you can't change a reply's "class" with change_reply_status(), only the response code within that class can be manipulated. I guess my question is now whether there is any way to do this sort of transformation with kamailio, perhaps with some other function or method? If anyone else has had to deal with this problem and come up with a different solution I'd be glad to hear it, as neither my head nor the surrounding walls can take any more mutual banging at this point. Thanks! BR, George. P.S. I hate fax

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Charging support/protocol information in Kamailio
by amit Kumar 21 May '18

21 May '18

Hi All, Could you please provide the charging related Information in Kamailio. ie list of Charging protocol/standard Kamailio Support ??. As per the kamailio modules(ims modules) , Kamailio support Diameter.Is there any other standard/interface Kamailio support.?? Thanks, amit

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Cannot get rid of following errors, and wondering if this is causing some of my issues
by Wilkins, Steve 21 May '18

21 May '18

ERROR: <core> [core/pvapi.c:1452]: pv_printf(): no more space for spec value ERROR: <core> [core/pvapi.c:1461]: pv_printf(): buffer overflow -- increase the buffer size... I have pv_buffer_size=16384; tcp_rd_buf_size=16384; Any ideas? Thank you,

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Supper confused on Kamailio with Asterisk Call
by Wilkins, Steve 21 May '18

21 May '18

Hi All, I am using Kamailio 5.2 and Asterisk (14.6 & 15.x) and I am having a very strange issue. If I use Asterisk 14.6, the call (WebRTC<=>WebRTC) works perfectly. However, if I use Asterisk 15.x, the call drops in 30 seconds. In comparing tcpdump files, the first place I see a difference is in the '200 OK' response to Kamailio' s 'INVITE to Asterisk'; in Asterisk 15.3 the first record Record-Route is Kamailio Public IP Address, however, in the Asterisk 14.6 (the working call) it is Kamailio' s Private IP Address. Asterisk Developers told me it is Kamailio doing this...and I am confused. Is this true, and if so why would it be different based on the Asterisk version? Thank you!

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Sudden database drop
by Amar Tinawi 20 May '18

20 May '18

Hello list recently i faced strange behaviour when dealing with ng-voice version of kamailio ims specially in P-CSCF node, the database "pcscf" dropped suddenly (deleted) without any command or action from user this happened many times and unfortunately there is nothing in the syslog as error massages Any advice please ? Thanks Cheers, Ammar

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WebRTC to WebRTC call Kamailio ACK Problem
by Wilkins, Steve 20 May '18

20 May '18

Hello, I am using Kamailio 5.2 and Asterisk 15.3 and while doing a WebRTC (client1) to WebRTC (client2) call, Asterisks sends a re-invite and Kamailio responds with a '200 OK', but when Asterisk sends back the "ACK" to Kamailio, it cannot be relayed. Errors => WARNING: <core> [core/msg_translator.c:2767]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found ERROR: <core> [core/msg_translator.c:1982]: build_req_buf_from_sip_req(): could not create Via header ERROR: <core> [core/forward.c:549]: forward_request(): building failed ERROR: sl [sl_funcs.c:362]: sl_reply_error(): stateless error reply used: I'm terribly sorry, server error occurred (1/SL) As a side note, I still occasionally see the Errors => ERROR: <core> [core/pvapi.c:1452]: pv_printf(): no more space for spec value ERROR: <core> [core/pvapi.c:1461]: pv_printf(): buffer overflow -- increase the buffer size... No matter how large I make pv_buffer_size, and tcp_rd_buf_size; pv_buffer_size=16384, and tcp_rd_buf_size=16384; Thank you all!

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deb.kamailio.org bionic
by Kelvin Chua 19 May '18

19 May '18

any plans of adding support for bionic? Kelvin Chua

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Transport issue thought
by Wilkins, Steve 18 May '18

18 May '18

It appears that the bottom line of my TCP transport not working is that Kamailio is randomly assigning large port numbers to send the TCP traffic out on. I am not able to randomly open high ports for this purpose. Is there a way to tell Kamailio to only use specific ports for this. I have tried using force_send_socket() with Kamailio' s IP, and the port I want to send out on, but this did not work either. Does this sound like I am on the right track? Thank you, -

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Transport question
by Wilkins, Steve 18 May '18

18 May '18

Hi All, I was attempting to switch from UDP to TCP for relaying. I used t_relay_to_tcp() instead of t_relay() => if (!t_relay_to_tcp()) { sl_reply_error(); } The problem is that I am still seeing the SIP messages as UDP on the Server that the messages are relayed to. Any Ideas? Thank you, Steve

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Hunting - Dynamic logistics
by KamDev Essa 18 May '18

18 May '18

So I finally got to the docs on hunting under the TM Modules and there was an awesome write up and it is making a lot more sense now. Basic logic is to append_branch with your extensions and and manage serial and/or parallel forking by setting the q values. The script is static. Is there a module that automatically loads registered contacts from a grouped list of extensions or is the best way to dip into the db and DIY? I would like to know how some of you have implemented hunting, Also is there a way to set the fr_inv_timer of each branch. Thanks in advance. KD

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Alcatel-Lucent Enterprise acquires majority stake in Sipwise
by Andreas Granig 18 May '18

18 May '18

Dear community, I'd like to inform you about Alcatel-Lucent Enterprise (ALE) acquiring a majority stake in Sipwise. Over the last 10 years, we at Sipwise were committed to contributing significant efforts to the open source telecom environment, resulting amongst others in the free and open source Sipwise CE soft-switch appliance, the rtpengine media relay, as well as major contributions to the Kamailio SIP proxy. Under the ALE roof, Sipwise is continuing to operate as its own brand, and our commitment to open source and our community components will be as strong as ever. Please find the official announcement here: https://www.sipwise.com/ale-sipwise/ Best regards, Andreas

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Re: [SR-Users] rtpproxy not forwarding rtp packets.
by Giovanni Tommasini - evosip 18 May '18

18 May '18

Hi Younas, when you have a call if you make a trace in your sip server with sngrep or tshark can you see the RTP packet ? you says "I have kamailio server behind nat with rptproxy.", so is it possible that the router in front your SIP server blocks the traffic? could you have a trace there? I mean the packets exit the rtpproxy but don't arrive to callee or just don't come out of the rtpproxy? Giovanni Tommasini | *evosip* <http://evosip.cloud> 2018-05-18 1:36 GMT+02:00 Aqs Younas <aqsyounas(a)gmail.com>: > Thanks for replying. Callee is actually a sip provider. I see no firewall > or iptables rules on server to prevent rtp following towards callee. > > Is there anything i can do to see which thing is blocking rtps. > > Best > > On Fri, 18 May 2018 at 3:59 AM, Mack Hendricks <ap(a)goflyball.com> wrote: > >> Is there something blocking RTP traffic from reaching the callee? Is the >> callee a carrier or the actually endpoint endpoint (aka SIP Phone)? >> >> *Mack Hendricks / Head of Support / dOpenSource* >> web: http://dopensource.com >> support: +888-907-2085 >> dSIPRouter <http://dsiprouter.org> - GUI focused on implementing >> Kamailio to provide SIP Trunking and PBX Hosting Services >> >> On May 18, 2018, at 12:50 AM, Aqs Younas <aqsyounas(a)gmail.com> wrote: >> >> Greetings list, >> >> >> I have kamailio server behind nat with rptproxy. But i am getting no >> voice on the call. After taking trace i could see that rtpproxy was getting >> rtp packets but not packets was being forwarded towards callee side. >> >> Though i see in rtpproxy logs, packets being relayed from caller side. >> >> Does rtpproxy wait to receive a single rtp frame from callee before >> fowarding that to callee? >> >> Since, i am getting no rtp from callee. So, might be rptrpoxy is waiting >> to learn source address of callee side rtp, that is why it is not >> forwarding rtp packets from caller towards callee. >> >> >> Any pointer/suggestion is much appreciated. >> >> Best Regards, >> >> Aqs Younas >> >> _______________________________________________ >> Kamailio (SER) - Users Mailing List >> sr-users(a)lists.kamailio.org >> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> >> >> _______________________________________________ >> Kamailio (SER) - Users Mailing List >> sr-users(a)lists.kamailio.org >> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> > > _______________________________________________ > Kamailio (SER) - Users Mailing List > sr-users(a)lists.kamailio.org > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > >

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Help on how to debug cfg_get variable having unexpected value?
by Malcolm O'Hare 18 May '18

18 May '18

I'm trying to debug an issue I've encountered where a variable that I've set using kamcmd cfg.set has an unexpected value when used in kamailio. The variable should be 0 or 1, but it looks like its getting set accidently somehow. 2018-05-11T22:02:58.344260+00:00 ip-172-31-129-45 kamailio[5253]: DEBUG: <core> [select.c:263]: resolve_select(): 'with_dynamodb_user_data' 2018-05-11T22:02:58.344264+00:00 ip-172-31-129-45 kamailio[5253]: DEBUG: <core> [cfg/cfg_select.c:174]: select_cfg_var(): DEBUG: select_cfg_var(): select fixup is postponed: features.with_dynamodb_user_data 2018-05-11T22:05:03.467110+00:00 ip-172-31-129-45 kamailio[5551]: INFO: <core> [cfg/cfg_ctx.c:608]: cfg_set_now(): INFO: cfg_set_now(): features.with_dynamodb_user_data has been changed to 1 2018-05-11T22:15:51.695118+00:00 ip-172-31-129-45 kamailio[5574]: [2c03c333-1705-41b1-aa26-6f10492a5a6f] DEBUG: with_dynamodb_user_data is 1157627905 I've tried to debug using gdb using breaks and set a watch on the memory address of the var, but it looks like it keeps using different memory locations. The value in the address pointed at by group.handle seems to change for each invocation of set_cfg_now, so when I try and put a watch on the address of *group.handle + var.offset it never gets hit. (gdb) print *group $36 = {num = 25, mapping = 0x7f56ee4474f0, vars = 0x0, add_var = 0x0, size = 164, meta_offset = 968, var_offset = 984, handle = 0x7f56edf92fd0, orig_handle = 0x0, dynamic = 1 '\001', next = 0x0, name_len = 8, name = "f"} (gdb) print *var $62 = {def = 0x7f56ee471668, name_len = 23, pos = 24, offset = 164, flag = 0} Any ideas? Malcolm

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Auth module with shared "secret"
by Asgaroth 18 May '18

18 May '18

Hi All, I am testing a scenario where we have 3 proxies and I'm trying to ensure that if a messages comes in to 1 proxy, which challenges the ua, is able to be auth'd successfully on one of the remaining 2 proxies if, for some reason the ua decides to send the response to one of the remaining two proxies. I am testing this using a seperate kamailio proxy which loadbalances messages across the 3 proxies that I am testing with. I have 1 ua (zoiper) which I tell to subscribe for mwi, the initial subscribe is sent to proxy 1, which sends a 407 back, the new subscribe with auth, from the ua, is sent to proxy 2, which deems the nonce as expired and rechallenges, this continues in a loop across the 3 proxies and never authenticates. I think I'm missing something in my settings that I just cannot put my finger on, and/or, I'm mis-understanding the documentation and how these settings interact with each other. My auth module settings are as follows: # # ----- auth params ----- # modparam("auth", "qop", "auth") modparam("auth", "nonce_count", 1) modparam("auth", "one_time_nonce", 1) modparam("auth", "nid_pool_no", 4) modparam("auth", "nc_array_size", 4194304) modparam("auth", "otn_in_flight_no", 8388608) modparam("auth", "auth_checks_no_dlg", 9) modparam("auth", "auth_checks_in_dlg", 15) modparam("auth", "auth_checks_register", 11) modparam("auth", "secret", "secret_32_char_str") I tried with one_time_nonce enabled and disabled but the result is the same, continous rechallenge. The documentation mentions that if nonce_count and one_time_nonce are enable, and a ua sends a response with nonce and qop in the auth header then one_time_nonce will not be used, one_time nonce will be used as a fallback in the case the ua does not support qop. My auth block in the routing logic looks like so: route[AUTH_CHECK]{ xlog("L_DBG", "route[AUTH_CHECK] : $rm : Performing authentication checks for '$var(creds_key)'"); $var(challenge_flags) = 1; if ( ! pv_auth_check("$fd", "$sht(creds=>$var(creds_key)::Password)", "20", "1") ) { switch($retcode) { case -1: xlog("L_DBG", "route[AUTH_CHECK] : $rm : Generic error occurred, no reply sent out."); break; case -2: xlog("L_DBG", "route[AUTH_CHECK] : $rm : Invalid password supplied, re-challenging client and removing from hash table."); sht_rm_name_re("creds=>$var(creds_key)"); break; case -4: xlog("L_DBG", "route[AUTH_CHECK] : $rm : Nonce has expired, re-challenging client."); $var(challenge_flags) = $var(challenge_flags) + 16; break; case -5: xlog("L_DBG", "route[AUTH_CHECK] : $rm : No credentials supplied, challenging client."); break; case -6: xlog("L_DBG", "route[AUTH_CHECK] : $rm : Nonce has already been used to auth a previous request, challenging client."); break; case -8: xlog("L_DBG", "route[AUTH_CHECK] : $rm : Auth user is different to From/To user, challenging client."); break; } xlog("L_INFO", "route[AUTH_CHECK] : $rm : auth_challenge will be called with '$fd' and '$var(challenge_flags)'"); auth_challenge("$fd", "$var(challenge_flags)"); exit; } consume_credentials(); } When I look at the logs I always see the proxies seeing the nonce as expired: oUBGp0n6HRWu_mvPPxJQqg..: INFO: <script>: route[AUTH_CHECK] : SUBSCRIBE : auth_challenge will be called with 'example.com' and '17' oUBGp0n6HRWu_mvPPxJQqg..: INFO: <script>: route[AUTH_CHECK] : SUBSCRIBE : auth_challenge will be called with 'example.com' and '17' oUBGp0n6HRWu_mvPPxJQqg..: INFO: <script>: route[AUTH_CHECK] : SUBSCRIBE : auth_challenge will be called with 'example.com' and '17' oUBGp0n6HRWu_mvPPxJQqg..: INFO: <script>: route[AUTH_CHECK] : SUBSCRIBE : auth_challenge will be called with 'example.com' and '17' oUBGp0n6HRWu_mvPPxJQqg..: INFO: <script>: route[AUTH_CHECK] : SUBSCRIBE : auth_challenge will be called with 'example.com' and '17' oUBGp0n6HRWu_mvPPxJQqg..: INFO: <script>: route[AUTH_CHECK] : SUBSCRIBE : auth_challenge will be called with 'example.com' and '17' I have checked: [1] that the proxies are all syncing time, and they are not out of sync, [2] The source address of the client doesnt change (in case auth_checks_no_dlg is failing), [3] The r-uri of the subscription dialog is the same (in case auth_checks_no_dlg is failing) I am using kamailio version 5.0.6: kamailio -V version: kamailio 5.0.6 (x86_64/linux) 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: unknown compiled on 15:57:47 Mar 5 2018 with gcc 4.8.5 Any thoughts on what I can look at or if I'm missing something fundamental here? All suggestions/criticisms are appreciated. Thanks for reading.

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rtpproxy not forwarding rtp packets.
by Aqs Younas 17 May '18

17 May '18

Greetings list, I have kamailio server behind nat with rptproxy. But i am getting no voice on the call. After taking trace i could see that rtpproxy was getting rtp packets but not packets was being forwarded towards callee side. Though i see in rtpproxy logs, packets being relayed from caller side. Does rtpproxy wait to receive a single rtp frame from callee before fowarding that to callee? Since, i am getting no rtp from callee. So, might be rptrpoxy is waiting to learn source address of callee side rtp, that is why it is not forwarding rtp packets from caller towards callee. Any pointer/suggestion is much appreciated. Best Regards, Aqs Younas

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Further question on transport issue.
by Wilkins, Steve 17 May '18

17 May '18

Hello, I am still struggling with getting TCP relay to work. Kamailio is now trying to send TCP but it is using the incorrect port. Kamailio is relaying to Asterisk on xx.xx.xx.xx:some-large-port but Asterisk is listening on xx.xx.xx.xx:5060. I'm not sure why this is happening. I would appreciate any help. Thank you,

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cfg script 101
by KamDev Essa 17 May '18

17 May '18

I have decided to do Hunting DIY. I needed some help with the cfg script part and using a database dip. Say I do the db dip sql_xquery("ca", "call usp_GetHuntBranches( '$rU', '$rd');", "ra");that will return the uri/q pairs and I will add using append_branch dynamically. How do I do the looping in the script? I have 2 ways to deliver the list via the sp. - Rows with uri and q values. easiest - Single row with variable paired fields like say ur1,q1,ur2,q2 etc etc Questions : - Can the script handle sql_xquery with multiple rows. If so, can I get a code snippet or example that show me how to iterate through. Is there a "while" loop capability in the script? - Is there a "substring" or "split" function in the script that could allow me to get the data as a single text with uri and q separated by a \ or |. - Or is there a document that explains the various generic functions I can use the cfg script area. KD

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NatHelper's Keep-alive timeout behind proxy
by Asgaroth 17 May '18

17 May '18

Hi All, We are looking to enable nathelpers keepalive timeout parameter where it will remove an aor's contact if it doesn't receive a reply from the ua. The problem we have is that our registrar's are seperated from our proxies, so this means nathelper is sending options messages via the proxy to the ua. In this case, if the us has lost connectivity, the kamailio proxy sends a 408 Request Timeout back to the registrar and the nathelper module sees that as a response to the initial options message it sends, so it never actually removes the contact from the location table. Is there a way I can stop the proxy from sending any responses for these specific nat ping/options messages, or, is there a way I can tell the nathelper modules that a 408 response should be treated as the ua being down and then after the timeout interval remove the contact from the location table? I've had a read of the nathelper, tm and sl modules but I didnt see anything that jumps out at me that I could achieve one or the other possible solutions I mentioned above. Does anyone have any suggestions or pointers on how we can achieve this with the registrars being seperate from the edge proxies. Any pointrs/tips etc would be greatly appreciated. Thanks

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Updated NAT traversal article
by Alex Balashov 17 May '18

17 May '18

Hi, I have updated this article with some other topics: http://blog.csrpswitch.com/server-side-nat-traversal-with-kamailio-the-defi… "The definitive guide" is of course an ambitious and moving target, so there may be more evolution in the future. -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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t_load_contacts() and t_next_contacts() guts
by KamDev Essa 16 May '18

16 May '18

I am tackling Hunting and see that there are two functions recommended to accomplish this. Before I jump into using them I wanted to know a bit about each regard to the call flow. >From where does t_load_contacts() load contacts and how does it order then in case of a serial fork flow, Is there a generic way of doing it say with a loop in the script. Docs on these two functions are a bit cryptic for me H

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Contact Header
by KamDev Essa 16 May '18

16 May '18

My scenario is UA <====> Kamailio <====> Freeswitch UA calls a number. Kam delivers to FS and FS plays a wav file. My problem is that Kam relays the 200 OK from FS back to UA without change and UA ACKs back to FS because the FS IP is in the Contact Header. - Why does Kam not change the contact header since the ACK should come back to Kam - If this is normal what do I need to do to change the contact header for the 200K to the UA to make this work. KD

3 4

Whats is wrong in this call
by Sergey Safarov 15 May '18

15 May '18

Hello I have a call with not passed ACK by Kamailio. http://prntscr.com/jhtobo Same behaviour i looked before when SIP endpoint contact string is wrong In this call look as contact string correct is correct (packets 7 and 8) and i not understand why ACK (packet 9) not passed to callee. Could you help me understand this. PCAP file is create using homer tracer and TCP transport packets is stored as UDP. Sergey

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Last negative 5XX reply not getting relayed
by Sergiu Pojoga 15 May '18

15 May '18

Not sure if it's a default Kamailio behavior or something I'm doing wrong... When receiving a negative reply for an INVITE (more like in the 5xx range), proxy doesn't relay the message to the originator but instead issues it's own "500 Service Unavailable". That doesn't happen, from observations so far, to "603 Declined" or "486 Busy Here" or "404 Not Found" or "200 OK" replies. *For example, here's the negative reply proxy receives from the callee:* 2018/05/14 10:52:37.716627 65.XX.XX.166:5060 -> 65.XX.XX.167:5060 SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 65.xx.xx.167;branch=z9hG4bK9223.00b4f3143d245a895c4bb6a107add258.0;received=65.xx.xx.167;rport=5060 Via: SIP/2.0/UDP 205.xx.xx.221;received=205.xx.xx.221;rport=5060;branch=z9hG4bKQ4XpXga0vgZ8B From: "514XXXXXXX" <sip: 514XXXXXXX @205.xx.xx.221>;tag=tXj0NHrje9jNS To: <sip:450XXXXXXX@65.xx.xx.167>;tag=as18e452b7 Call-ID: 48eac296-d229-1236-a685-005056a149a4 CSeq: 122818714 INVITE Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas X-Asterisk-HangupCause: Subscriber absent X-Asterisk-HangupCauseCode: 20 Content-Length: 0 *Here's what the originator gets in the final message:* 2018/05/14 10:52:37.718143 65.xx.xx.167:5060 -> 205.xx.xx.221:5060 SIP/2.0 500 Service Unavailable Via: SIP/2.0/UDP 205.xx.xx.221;rport=5060;branch=z9hG4bKQ4XpXga0vgZ8B;received=205.xx.xx.221 From: "514XXXXXXX" <sip:514XXXXXXX@205.xx.xx.221>;tag=tXj0NHrje9jNS To: <sip:450XXXXXXX@65.xx.xx.167>;tag=31a2c69b5d434ef4a60796e26f568bf0-9343 Call-ID: 48eac296-d229-1236-a685-005056a149a4 CSeq: 122818714 INVITE Server: Kamailio 5.1 Content-Length: 0 I'd like any reply to be relayed back as is, unless I do some t_check_status() => t_reply() for specific reasons (which I have none at this point). Thanks Sergiu

3 5

ACC_CDRS table with src and dst data are inverted
by Emanuele Gambaro 15 May '18

15 May '18

Hi to all, I have some troubles with acc_db module and acc_cdrs table. First scenario: Alice calls Bob and after a while, when a call is established, Alice hangs up, sending a BYE. The acc_cdrs table is filled up with the right data: * src_user, src_domain, src_ip are from Alice account (the caller) * dst_user etc etc are from Bob account (the callee) In this scenario all works as expected: the src data are from who starts the call, and the dst data are from who receives the call. In the following scenario src and dst data are inverted. Alice calls Bob and after a while, when a call is established, Bob hangs up, sending a BYE. The acc_cdrs table now is: * src_user, src_domain, src_ip are from Bob account (the callee that sends the BYE message) * dst_user etc etc are from Alice account (the caller that receives the BYE) This is the intended behavior or I’m missing something? How I can record the CDR data with the right CALLER and CALLEE data ? Emanuele

2 1

UAC Module Registration Table
by Greg Adams 15 May '18

15 May '18

Hey there, I am currently using the UAC module to maintain remote registrations. I am just curious if it is possible to register new users immediately rather than waiting for reg_timer_interval? I am not seeing any functionality like that but just wondering if I am missing something or if it would be possible to add in. For example an RPC command that takes a key and instantly registers that user. Thanks, Greg

2 1

Parallel Forking - 1 Ring on extension to notify that Call Forward is in place
by KamDev Essa 14 May '18

14 May '18

I am trying to simulate a Broadsoft feature that blips the extension with a single ring that basically heralds the user that their extension has CF turned on. I know parallel forking can do that. But how do I set the fr timer on the call to the extension @ just 1 ring (5 seconds) and let the other one the default timeout set in fr_inv_timer. KD

2 1

Tagging a call with custom pseudo variables - cfg 101
by KamDev Essa 14 May '18

14 May '18

Being spoiled by Freeswitch profiles and dialplans, the stock 1 piece Kamailio cfg file is cool @ first to get certain call flows knocked out but quite a maze when you get in deeper. Right now, using the 1 piece stock cfg file, I am faced with the fact that my carrier outbound calls started failing after I enabled VM. Basically WITH_VOICEMAIL in route LOCATION started sending the carrier calls to TOVOICEMAIL and it basically dies there. I can see the route(TOVOICEMAIL) line and can condition it out with an if but I really don't now how to differentiate between a ext to ext call or a outbound call in the cfg file. However given SQL I could. Is there a way for me to place custom pseudo variables in the call right @ the start and use them in the script as needed. For example I could mark the call as ext to ext or carrier bound right @ the start and dodge VM in the LOCATION route. KD

2 5

Question about using Kamailio and Asterisk and flow of an "INVITE"
by Wilkins, Steve 13 May '18

13 May '18

Hello All, I am trying to resolve, in my mind, the flow of a WebRTC<=>WebRTC call using Kamailio and Asterisk. Each WebRTC client is registered in Kamailio and when I call WebTRC Client1 from WebRTC Client2 what I see is -> The Invite is sent from Kamailio to Asterisk and then Asterisk is sending the Invite back to Kamailio. Also depending on The version of Asterisk, the INVITE will then get forwarded to the AOR that is registered in Kamailio for the called number. Does this seem correct? It seems like there is an extra hop in there. The reason I am now very curious now is because everything works fine if using Kamailio 5.0 and Asterisk 14.6, but I switch to Asterisk 15.3 I get the extra hop and call is dropped after 30 seconds. I would appreciate any thoughts on this. Thank you in advance.

3 6

"Sleeping Giant - No Snore" Kam?
by KamDev Essa 11 May '18

11 May '18

We have noticed pesky SIP hackers already knocking on our POC servers. Kam is holding the fort by sending out the proverbial 401/407 (snore). I can SQLOPS to do a dip to check destination domain or Source IP and nix the snore for alien requests, but wanted to know if there was a native way to keep silent when we have a request from foes. KD

3 6

Understanding port changing
by Social Boh 11 May '18

11 May '18

Hello list, I'm looking for to understand a problem have with Kamailio. The configuration is: user (behind NAT) -> Kamilio (Public IP) -> user (behind NAT) When I make a call from user A to user B (both registered) , User B receive the INVITE correctly but in the ringing SIP message change the port present on the Contact Header and same think on 200OK so when Kamailio try to send it ACK SIP message to this port fail because the Softphone listen on the port where has received the INVITE from Kamailio. This happens with 3 different Softphone: BRIA, Xlite, Linphone on the same computer. Thank you Regards -- --- I'm SoCIaL, MayBe

2 1

WebRTC to SIP gateway
by Pan Christensen 11 May '18

11 May '18

Hello! It's been several years since I've used Kamailio. My current employer wants to implement WebRTC, which is currently not supported in our SIP backend, and asked if I could set up a Kamailio server as a gateway. I've been able to make calls in all directions between SIP and WebRTC clients registered locally on Kamailio. When I tried to connect the server to the SIP backend, I ran into an issue. I'm able to register SIP clients in the backend via the gateway and make calls everywhere. However, the WebRTC client fails to register. Here are the messages between the Kamailio gateway and the SIP backend: U 2018/05/09 10:12:58.316643 GATEWAY:15060 -> DOMAIN:5060 REGISTER sip:DOMAIN SIP/2.0. Via: SIP/2.0/UDP GATEWAY:15060;branch=z9hG4bK4fc6.04d1730be5b78d595c69a3aa137987c1.0. Via: SIP/2.0/AUTO lr2l9s72ehhc.invalid;rport=61353;received=CLIENT;branch=z9hG4bK5927151. Max-Forwards: 68. To: <sip:4777519304@DOMAIN>. From: <sip:4777519304@DOMAIN>;tag=9qhqhrnj3s. Call-ID: j5h7830ivr5dfc2mn5sov1. CSeq: 4 REGISTER. Contact: <sip:c3qkm4fv@lr2l9s72ehhc.invalid;transport=ws>;+sip.ice;reg-id=1;+sip.instance="<urn:uuid:0ef2deac-2d56-465a-840b-543b9fd01af8>";expires=600. Expires: 600. Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO. Supported: path,gruu,outbound. User-Agent: JsSIP 3.2.9. Content-Length: 0. Path: <sip:GATEWAY:15060;lr>. . U 2018/05/09 10:12:58.368409 DOMAIN:5060 -> GATEWAY:15060 SIP/2.0 400 Wrong transport. Provided transport either invalid or not supported.. Via: SIP/2.0/UDP GATEWAY:15060;branch=z9hG4bK4fc6.04d1730be5b78d595c69a3aa137987c1.0. Via: SIP/2.0/AUTO lr2l9s72ehhc.invalid;rport=61353;received=CLIENT;branch=z9hG4bK5927151. To: <sip:4777519304@DOMAIN>;tag=91334f57. From: <sip:4777519304@DOMAIN>;tag=9qhqhrnj3s. Call-ID: j5h7830ivr5dfc2mn5sov1. CSeq: 4 REGISTER. Content-Length: 0. I believe that this error message is caused by ';transport=ws' in the Contact header. I'm not allowed to modify this header. In the backend database, I found that some other clients have ';transport=UDP' in their path headers, so I tried to add that. (Why can I not add parameters in path module without adding username?) I still got the same error. How do I best proceed? For your information: We have outsourced the development of the WebRTC client, so we are able to change it. We also have the option of paying the supplier of the backend for development there. With kind regards Pan B. Christensen Developer Phonect AS Brugata 19, PB 9156 Grønland, N-0133 Oslo, Norway E-mail: pan.christensen(a)phonect.no<mailto:pan.christensen@phonect.no> Mobile: +47 41 88 88 00 [cid:image007.png@01D3A0E8.376921D0] <http://www.phonect.no/> [facebook_2]<https://www.facebook.com/phonectno> [LinkedIn_logo_initials (1)] <https://www.linkedin.com/company/44983?trk=tyah&trkInfo=clickedVertical%3Ac…>

2 2

"No Answer" time out after number of rings
by KamDev Essa 11 May '18

11 May '18

Does Kamailio have any way to control how long it tries to connect per call? KD

2 5

Is Media Release (signaling only) possible in Kamailio?
by KamDev Essa 10 May '18

10 May '18

What cfg files changes do I need to make to get Kamailio to be a signally only server, yet manipulate the SDP part of the INVITE message to allow remote parties to send media to each other? In Freeswitch terms "bypassmedia". KD

2 14

Build Kamailio with MariaDB failed.
by Ye, Charlie (NSB - CN/Qingdao) 10 May '18

10 May '18

Hi all, Does everyone use MariaDB in Kamailio? When I built Kamailio with MariaDB, the below error happened. Would anyone give a help? Thanks in advance! [root@clcmos /usr/local/src/kamailio-devel/kamailio]$ # make all make -C src/ all make[1]: Entering directory `/usr/local/src/kamailio-devel/kamailio/src' make[3]: `libsrdb2.so.1.0' is up to date. make[3]: `libsrdb1.so.1.0' is up to date. LD (gcc) [M db_mysql.so] db_mysql.so /usr/bin/ld: cannot find -lmariadb collect2: error: ld returned 1 exit status make[2]: *** [db_mysql.so] Error 1 make[1]: *** [modules] Error 1 make[1]: Leaving directory `/usr/local/src/kamailio-devel/kamailio/src' make: *** [all] Error 2 Thanks and Best Regards, Charlie

4 6

Parse presece xml and put stats into asterisk database
by Aqs Younas 09 May '18

09 May '18

Greetings list, I need to put presence info of a device into asterisk database through sqlops module. Currently, i opt to use a lua script to parse xml body of PUBLISH packet issued by pua_dialoginfo and insert trying/ringing stats into asterisk database. Is there any better way to accomplish this? Best Regards, Aqs Younas

1 0

P-CSCF error: pv_parse_spec2(): error searching pvar "uac_req"
by John M 09 May '18

09 May '18

Hello, I upgraded Kamailio 4.4 to 5.2.0-dev5 and when I tried to start the service(p-cscf), I am getting below error. However I am able to start the service with 4.4 p-cscf example cfg. Also after upgrading I-CSCF and S-CSCF nodes are running without any issue. Any help would be greatly appreciated! [root@8bd3221bd31e kamailio]# kamailio -f /usr/local/src/kamailio-5.1/kamailio/misc/examples/ims/pcscf/kamailio.cfg loading modules under config path: /usr/local/lib64/kamailio/modules/ 0(2464) ERROR: <core> [core/pvapi.c:903]: pv_parse_spec2(): error searching pvar "uac_req" 0(2464) ERROR: <core> [core/pvapi.c:1107]: pv_parse_spec2(): wrong char [m/109] in [$uac_req(method)] at [9 (5)] 0(2464) CRITICAL: <core> [core/cfg.y:3489]: yyerror_at(): parse error in config file /usr/local/src/kamailio-5.1/kamailio/misc/examples/ims/pcscf/kamailio.cfg, line 831, column 3-18: Can't get from cache: $uac_req(method) ERROR: bad config file (1 errors) 0(2464) INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Regards, Uts

2 1

Use GStreamer instead of FreeSwitch or Asterisk?
by Chandramouli P 09 May '18

09 May '18

Hello, I am using Kamailio as registration server and FreeSwitch for signalling (RTP packet handling). Can I use GStreamer instead of Freeswitch or Asterisk? Thank you. Regards, CM

2 1

Connecting UAs behind Firewall/CgNat with Kamailio without using a Stun Server
by Pinter, Gerd. 09 May '18

09 May '18

Hi there, unfortunately I am personally not a complete IT geek, I am more in to Pro Audio. Anyway, I have to maintain a SIP Server for high quality Audio transmission with special UA's. Also we use Software UA's running as Apps (Luci Live) on mobile devices such as I Phones and I pads which are by nature situated in mobile networks. Due to "lack of IP address issues" most providers use private ranges separated by CGNAT from the real Internet. This causes a lot of trouble. In some cases stun server entries on the clients helps, in other cases don't! But these days I read in announcement of a german company (Mayah Communications), which is in the ProAudio biz since decades, that they offer a SIP Service where no stun is recommended whether the clients are behind NAT or not. After I acknowledged that IT business is not witchcraft (almost :-) ) , I thought that there might exist a solution for my Kamailio SIP server to make it work like the "Mayah" Sip Server. What settings have to be done that Kamailio uses IP Header Information for SIP signalling instead of the content of SIP packages themselves. Without stun you find the Local Address in the SIP packages, and with Stun the external Address but mostly with the wrong portnumber, but in the IP header you'll find the correct IP:port. I already read a lot of the Kamailio documentation but due to the fact that I am not too deep in IT I am not sure if I got it all. If someone could give me a useful hint on that, that would make my day. Hope this description is unterstandable. Nice Regards Gerhard Pinter

4 8

Kamailio + RTPEngine + RTPEngine Recording daemon forward-to option
by Balázs Radák 08 May '18

08 May '18

Dear all, I would need some help setting up Kamailio with RTP engine. I'm trying to make use of RTPEngine recording daemon's forward-to option, to have the calls streams sent to 3rd party application. Have someone managed to have this forward-to option work? If I understood well, when RTPEngine's forward-to option is turned on, the call metainfo + streams can be sent to a socket server? Thanks in advance! Regards, Balazs

2 1

Copy $xavp
by Duarte Rocha 08 May '18

08 May '18

Greetings, I'm getting a query result in a $xavp. I'm trying to copy the entire structure to a new $xavp like this : $xavp(Valid_Acd_Results) = $xavp(ACD_Query_Result[$var(iterator)]); However, i can't retrieve the value with : xnotice(" Mode = $xavp(Valid_Acd_Results=>TBK_AcdMode)"); since it returns NULL. Am i doing something wrong? Thanks

1 0

Record-route and proxy_challenge
by Евгений Голей 08 May '18

08 May '18

Hi! Tell me why when using the save ('') function in the response generated by the server, there are the Record-route headers that were in the request. And when using the following functions: - proxy_challenge (), - www_challenge (), - send_reply (), - sl_send_reply there is no Record-route header. How can you add them? -- Evgeniy

2 2

Garbage in to field
by Kjeld Flarup 08 May '18

08 May '18

I'm having problems with my To field. I'm doing an async_route, and afterwards sends the call with auth to a SIP server. My problem is that in the first invite I see the expected To field sip:004520202020@isp.com But when the Auto is send the To field is changed to the, To field for the PBX sip:1234@pbx.local I tried to fix this with uac_replace_to("$ru"), but that ends up in this To field: sip:004520202020@isp.comsip:004520202020@isp.com It seems like uac_replace_to sometimes appends rather that replaces. If I run it twice, then it for sure appends. Is this expected behaviour. -- -------------------- Med Liberalistiske Hilsner ---------------------- Civilingeniør, Kjeld Flarup - Mit sind er mere åbent end min tegnebog Sofienlundvej 6B, 7560 Hjerm, Tlf: 40 29 41 49 Den ikke akademiske hjemmeside for liberalismen - www.liberalismen.dk

2 1

How to change Calling Line ID in cr_route
by KamDev Essa 08 May '18

08 May '18

I have cr_route working well. Only problem is that the username is sent off to the carrier and the 3 or 4 digit username appears as the callerid. Is there a module or scripts that can replace the username with a db table defined callerid. Say 202 maps to a 10 digit TN. KD

1 0

STUN
by Wilkins, Steve 08 May '18

08 May '18

Hello all, Does anyone know the current status of how to enable STUN. I am using Kamailio Ver 5.x. Thank you, -Steve

5 6

Kamailio v5.1.3 Released
by Daniel-Constantin Mierla 07 May '18

07 May '18

Hello, Kamailio SIP Server v5.1.3 stable release is out. This is a maintenance release of the latest stable branch, 5.1, that includes fixes since the release of v5.1.2. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.1.x. Deployments running previous v5.1.x versions are strongly recommended to be upgraded to v5.1.3. For more details about version 5.1.3 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2018/04/kamailio-v5-1-3-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

3 3

Determining next hop for any SIP message
by George Diamantopoulos 07 May '18

07 May '18

Hello all, I've been trying to figure out a cleaner way to determine the next hop for a SIP message, mainly for use within the NATMANAGE route for multi-homed kamailio instances (with three or more network interfaces on the kamailio host). So far I have achieved this with a series of nested ifs, depending on whether the message is a request or a response and by calculating the next hop based on the various headers (R-URI, Route) and variables ($T_req($Ri), $dd, ) involved in SIP routing. A simpler way to do it, of course, would be to use the onsend_route, but that would most likely introduce an unnecessary overhead for all routed messages. I recently noticed there a pseudovariable called $nh(key), and I believe I can use $nh(d) to the same effect. I understand however, that this works for requests only. Also, the description of this PV in the documentation reads as follows: Return attributes of next hop for the SIP request. Address is taken from > dst_uri, if set, if not from new r-uri or original r-uri. > What is not clear to me is if this covers in-dialog requests with the Route header set as well. Does the inclusion of a route header set the dst_uri PV? And if yes, is it safe to rely on dst_uri during request processing or is it set only after completion of script processing? Lastly, is there an analogue for SIP responses? If not, is it safe to rely on first Via header to determine next hop for responses, or are there any other corner cases I need to heed? Thanks in advance, BR, George

3 6

Server-side NAT traversal with Kamailio
by Alex Balashov 07 May '18

07 May '18

Recent repeated questions on this topic have inspired a blog post: http://blog.csrpswitch.com/server-side-nat-traversal-with-kamailio-the-defi… This is a rough cut, and may be edited or expanded later. -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

3 5

Route header in local reply
by Евгений Голей 07 May '18

07 May '18

Hi. Colleagues write how to configure the local responses to requests so that the response added the header of the Route in accordance with the record-route headers that were in the request ? Example SBC --> Kamailio REGISTER sip:domain.com SIP/2.0 Record-Route: <sip:AAA.AAA.AAA.AAA:6060;lr> Via: SIP/2.0/UDP CCC.CCC.CCC.CCC;branch=z9hG4bKae7f.474bd538e793b8bc912ce7d0c8e0bae8.0 Via: SIP/2.0/UDP S.s.s.s;branch=z9hG4bKae7f.d81efe54000000000000000000000000.0 To: <sip:000143949@domain.com> From: <sip:000143949@domain.com>;tag=d55866b28e60059575f1fd1facc53fdd-516b CSeq: 10 REGISTER Call-ID: f74322050fd19c19f5d0ce763cc7dbdb Max-Forwards: 70 Content-Length: 0 User-Agent: SBC Expires: 0 Kamailio --> SBC SIP/2.0 200 OK Record-Route: <sip:AAA.AAA.AAA.AAA:6060;lr> Via: SIP/2.0/UDP CCC.CCC.CCC.CCC;branch=z9hG4bKae7f.474bd538e793b8bc912ce7d0c8e0bae8.0 Via: SIP/2.0/UDP S.s.s.s;branch=z9hG4bKae7f.d81efe54000000000000000000000000.0 To: <sip:000143949@domain.com>;tag=550bc1454fc5070bdadb084c686d99a2.ac18 From: <sip:000143949@domain.com>;tag=d55866b28e60059575f1fd1facc53fdd-516b CSeq: 10 REGISTER Call-ID: f74322050fd19c19f5d0ce763cc7dbdb Server: kamailio (5.1.0 (x86_64/linux)) Content-Length: 0 -- Evgeniy

1 1

How long time does a push take?
by Kjeld Flarup 07 May '18

07 May '18

Hello I have a question to those of You who uses push via Google or Apple to initiate calls. How long time does it usually take, before the phone gets the push and wakes up? Our experience is, that it can take from 2 to 10 seconds, and that not impressing when we are talking about telephony. -- -------------------- Med Liberalistiske Hilsner ---------------------- Civilingeniør, Kjeld Flarup - Mit sind er mere åbent end min tegnebog Sofienlundvej 6B, 7560 Hjerm, Tlf: 40 29 41 49 Den ikke akademiske hjemmeside for liberalismen - www.liberalismen.dk

4 5

Pleasing all kinds of UACs for RTP/SRTP with rtpengine
by Sergiu Pojoga 07 May '18

07 May '18

In continuation of previous <https://marc.info/?l=sr-users&m=150730146302404&w=2>thread about none/optional/compulsory SRTP handling... I used the suggested approach of *t_reuse_branch()* in a *tm:branch-failure* detected by t_check_status("(415)|(488)") replies, with the only difference of me doing it vice-versa from the suggested mantra of offer SRTP first then fallback to RTP. It works... well, for some UACs, not so with others. The 'good' UACs happily accept an updated branch, like for e.g. Yealink, will reply with "*488 Not Acceptable Here*" at first followed by a *200 OK*. The 'bad' UACs, like for e.g. Bria || Zoiper, will issue a "*415 Unsupported Media Type*" at first, and then a "*482 Merged Request*" after receiving and updated branch with a different RTP profile. Rightfully so, they decided that the updated branch is a retransmission, since both branches have the same CSeq, Call-ID and From tag. The branch ID however differs, so does the RTP profile obviously. *Any suggestion on how to convince all UACs that the 2nd attempt is different than the 1st one?* According to RFC 3261, a request is a merged request if ... "8.2.2.2 Merged Requests If the request has no tag in the To header field, the UAS core MUST check the request against ongoing transactions. If the From tag, Call-ID, and CSeq exactly match those associated with an ongoing transaction, but the request does not match that transaction (based on the matching rules in Section 17.2.3), the UAS core SHOULD generate a 482 (Loop Detected) response and pass it to the server transaction."

2 2

TOPOS (5.1.[23]) breaks PRACK
by Daniel Tryba 07 May '18

07 May '18

Enabling topos breaks PRACK in my simple (near default) proxy setup (attached kamailio.conf). asterisk 13 (10.0.3.147) <-> kamailio (10.0.3.87) <-> asterisk 13 (10.0.3.154) With the event route topos:msg-outgoing dropping all. The flow of messages is as expected. INVITE -> <- Ringing PRACK -> <- 200 OK With topos enabled, the PRACK doesn't get past route[WITHINGDLG] and is answered with the sl_send_reply("404","Not here"); INVITE -> <- Ringing PRACK -> <- 404 Not here These specific tests were done on kamailio 5.1.2, but with 5.1.3 I see the exact same messages.

2 1

Need help with NAT Traversal - Changing INVITE OK contact to IP port to Contact in Location
by KamDev Essa 06 May '18

06 May '18

The stock cfg states definng WITH_NAT and alls good. I did exactly that. However boing with stock cfg and WITH_NAT, the Registration info in location table is perfect with the UAs external IP address and port. This is not the case with the OK that in response to an INVITE. In the Contact of the OK the private IP shows up and Kamailio just sends it off to the Carrier who obviously cannot send a ack to a private IP. Is there a step by step guide for NAT Traversal? H

1 0

log CDRs to text file
by eyas barhouk 06 May '18

06 May '18

Hello Dears I'm trying to get Kamailio to log its CDRs (collected By ACC module ) to text file instead of a database so i used modparam("acc", "db_url", "flatstore:/var/log/acc") and in each time kamailio trying to write on the text file i get the error : " db_flatstore flat_new_connection() : not a directory" so could any one guide me about how i have to create the database text file "acc"? and what is the cause of the error i got ?

2 2

Carrier IP on softphone call history !!!
by KamDev Essa 04 May '18

04 May '18

I have a simple call flow for Inbound call going to an extendion carrier call ===> WITH_ALIASDB ===> extension (on Xlite) Works perfectly getting the call. Issue : - On Xlite I see the call as <TN>@<carrierIP>. I dont want my customers to know what carrier I use and also I want to be in the loop for CDR. - Hanging on either phone will not send a BYE to my server. How do I get Kamailio to be in the SIP ladder for this call? KD

2 7

Private IP in Contact Header of 200 OK back to Carrier, so no ACK - Call drop
by KamDev Essa 04 May '18

04 May '18

Scenario Carrier ===> Kamailio ===> Ext on domainINVITE ===> ====> 200 OK <== <=== Repeated because carrier pcap shows private 192.168 IP in Contact header. cfg has WITH_NAT setup. I assumed that all I need to do. What more is required to get Kamilio IP and port in 200K to Carrier? K

2 2

Where do I get rpc documentation for Kamailio
by KamDev Essa 04 May '18

04 May '18

I would like to perform subscriber and domain adds and reloads programmatically. Where can I find info on kamailio APIs I could use? KD

2 2

Kamilio Hunting 101
by KamDev Essa 04 May '18

04 May '18

With alias behind me I want to tackle a simple sequential and simulring (parallel). I see docs on using lookup() for simul and t_load_contacts but I don't see the basics on what table needs to be populated to get this working. Test set is basic : 1. 1234567890 from carrier2. Hunt to 201(a)mytestdomain.com, 202(a)mytestdomain.com. These are both rgistered. What tables do I populate or what CLI tools do I use to get this working. Thanks KD

1 0

How does one populate the dbaliases table and how do we initiate a reload.
by KamDev Essa 04 May '18

04 May '18

How do I add entries into the dbaliases table and how do I reload without restating Kamailio. KD

2 3

MSILO: Message limit on SIP stored offline MESSAGE delivery
by Abdul Basit 04 May '18

04 May '18

Hi Team, I have setup IM server with kamailio using MSILO module. Its working fine now. I have a scenario where lot of offline messages are being stored in mysql database for a user B. As soon as destination (B) REGISTER with kamailio, these messages will deliver to him. Now, kamailio will deliver all of these messages that will cause user (B) device (softphone on mobile) stuck or freezs for the moment messages are being delivered like bombardment from kamailio. Is there any way, where I can define batch size of messages to be delivered after provided interval? like batch_size=10 messages and batch_interval= 5sec, etc. I want to avoid applying static limit to mysql query in module source code. Please advise. -- regards, abdul basit | p: +92 32 1416 4196 | o: +92 30 0841 1445

1 1

kamailio 5.1 keepalive module + usrloc
by Aidar Kamalov 04 May '18

04 May '18

Hi, can I with keepalive module ping with is_alive function endpoints connected to kamailio? Or this module can use only static destionations via modparam("keepalive", "destination", "sip.provider.com")? -- Aydar A. Kamalov

1 0

Relay Latency
by Daniel Greenwald 03 May '18

03 May '18

We are using kamailio 5.0 as an inbound dispatcher. Sometimes in testing we see it take over half a second to relay a 200 OK. The logs show the new dialog being creating for the 200 OK within .0001 seconds of receiving the inbound 200 but it takes another .5 seconds for the relayed 200 to actually hit the wire. Here is a screenshot showing the timing of the packets. The box is an EC2 m3.medium under almost zero load (1-2 calls). Is this normal? What should I be looking at? Screenshot below: 10.0.23.34 is the kamailio box. Carrier IP obfuscated. You can see the carrier resends the 200 OK because of the delay.

3 6

Kamailio with dispatcher not send BYE to Asterisk.
by Jeferson Oliveira 03 May '18

03 May '18

Hello everyone, I have an error that I have not yet been able to solve and would like the help of colleagues to indicate a correct path. The problem that is occurring is that when the client disconnects the call kamailio is not sending the BYE forward until arriving at the asterisk. Both in the test scenario and in the production scenario the problem is the same and the message I see in the capture is 404 Not here, msg this coming from kamailio. Production scenario. PSTN <----------> Dialer --------->kamailio -----------> asterisk1 -----------> asterisk2 Test scenario. sipp generated calls ------> kamailio -------> asterisk1 -------> asterisk2 When this occurs, the calls that are disconnected by the client are in a "zombie" state in asterisk, and end up being terminated by timeout with the following message in the asterisk CLI: /[Apr 25 17:49:59] WARNING[2121]: chan_sip.c:4072 retrans_pkt: Retransmission timeout reached on transmission 22-6073(a)10.110.7.242 for seqno 1 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions// //Packet timed out after 31999ms with no response/ In the sipp panel I see in the Retransmission column several incrementing counters, as per the attachment. If I take the kamailio from the move and point the sipp to only one of the asterisk, the retransmissions do not happen and BYE follows normally. My kamailio.cfg configuration file can be downloaded from this url: https://drive.google.com/file/d/1bBj4GEZSPrp1iJXSLWQ4Z6g59tEFjnNT/view?usp=… Thank you very much. --

4 8

Record-Route and received/rport parameters
by Asgaroth 03 May '18

03 May '18

Hi All, I've come accros a scenario where a proxy is sitting on a private address range and inserts a record-route specifying the the private address. This causes issues whereby the BYE to an INVITE attempts to relay to the private address defined in record-route. I was wondering, if we were inject a received and rport parameter into the record-route header of the original invite, whould kamailio relay the response (BYE) to the receive/rport destination instead of the uri defined in the record-route header. As an example, initial invite comes in with a record-route as follows: Record-Route: <sip:172.17.0.2:5062;lr;ftag=b4551d29> If we injected received and rport as follows: Record-Route: <sip:172.17.0.2:5062;lr;ftag=b4551d29;rport=33429;received=212.172.2.212> and relayed the message to the B2B. Then, I assume, when the B2B creates its BYE message, the Route header should look like this: Route: <sip:172.17.0.2:5062;lr;ftag=b4551d29;rport=33429;received=212.172.2.212> Once this hits the kamailio instance to relay to the last route header as mentioned above, would it set $du to received:rport like it does with Via headers? Thanks

2 4

DMQ and/or Database for dialogs, registrations, etc..
by Joel Serrano 03 May '18

03 May '18

Hi all, Just wanted to know what your opinions were on using DMQ modules over database for things like dialog replication, registrations, etc... Is DMQ the "new way to go"? I know that there lots of ways of doing things with each having pros/cons... But I was wondering... What does the community think on this topic? Are you guys taking advantage of the DMQ modules or are you still relying on database as much as possible? Maybe a combination of both? Cheers, Joel.

8 19

Seeing Large Waits
by Mack Hendricks 03 May '18

03 May '18

Hey All, We are load testing Kamailio 4.2.3 and we are seeing the waiting within the transaction manager increasing. I’ve increased the children from 8 to 16 to 24, but we are still seeing a large amount of waiting. I don’t see an errors in the logs. Can anyone give me a clue on where to look. 05-02-18_03:07:38 PM tm.stats { current: 390 waiting: 226 total: 2361 total_local: 0 replied_locally: 1406 6xx: 0 5xx: 0 4xx: 88 3xx: 0 2xx: 3180 created: 2361 freed: 1971 delayed_free: 0 } Mack Hendricks / Head of Support / dOpenSource web: http://dopensource.com <http://dopensource.com/> support: +888-907-2085 dSIPRouter <http://dsiprouter.org/> - GUI focused on implementing Kamailio to provide SIP Trunking and PBX Hosting Services

4 6

BLF+Asterisk+NAT
by John Bob 03 May '18

03 May '18

Phone sends Subscribe Kamailio Responds 202 OK Kamailio Responds with NOTIFY Via: SIP/2.0/TCP 66.66.66.66:5051;branch=z9hG4bKfe3e.e4fedb84000000000000000000000000.0 To: <sip:30041@66.66.66.66 <mailto:30041@66.66.66.66>>;tag=48ae253ab7 From: <sip:47701@66.66.66.66 <mailto:47701@66.66.66.66>>;tag=00c39aabf419ddeab4250cd65af8ef90-6fb2 CSeq: 2 NOTIFY Call-ID: 1493ef52c41b6c1a Content-Length: 0 User-Agent: kamailio (5.1.2 (x86_64/linux)) Max-Forwards: 70 Event: dialog Contact: <sip:192.168.1.30;transport=udp> Subscription-State: terminated;reason=timeout In that packet I see Subscription-state: terminated;reason=timeout Im not sure what I am missing or misconfigured. Any help would be great. I’ve attached my config. Thanks.

2 7

Seeing Large Waits
by Mack Hendricks 02 May '18

02 May '18

Hey All, We are load testing Kamailio 4.2.3 and we are seeing the waiting within the transaction manager increasing. I’ve increased the children from 8 to 16 to 24, but we are still seeing a large amount of waiting. I don’t see an errors in the logs. Can anyone give me a clue on where to look. 05-02-18_03:07:38 PM tm.stats { current: 390 waiting: 226 total: 2361 total_local: 0 replied_locally: 1406 6xx: 0 5xx: 0 4xx: 88 3xx: 0 2xx: 3180 created: 2361 freed: 1971 delayed_free: 0 } Mack Hendricks / Head of Support / dOpenSource web: http://dopensource.com <http://dopensource.com/> support: +888-907-2085 dSIPRouter <http://dsiprouter.org/> - GUI focused on implementing Kamailio to provide SIP Trunking and PBX Hosting Services

1 0

how to disable sip authentication
by BIPIN KUMAR Das 02 May '18

02 May '18

Hi, Is there any way in Kamailio where I can suppress sip registration authentication. I want, Kamailio can send 200 OK for sip registration without going for 401 Unauthorized. Please tell me if you have such configuration in Kamailio. thanks Bipin

2 1

Kamailio config file String Concatenation
by George Diamantopoulos 02 May '18

02 May '18

Hello, I'm a a bit confused as to when string concatenation will work in kamailio config file. So far I have identified the following cases: Working: As part of the right hand side of an assignment: $var(x) = "string1" + "string2"; String argument of some functions: xlog("L_ERR", "string1" + "string2" + DEFINED_STRING + '\n'); Conditional Statements: if ( ru =~ "$" + "sip:" + $var(user) ) { Not Working: In string transformations (the fact that quotes is not part of the argument should give a hint though: $var(x) = $(var(y){s.replace,match,"repl1" + "repl2"}); String argument of some other functions: append_hf_value("P-Asserted-Identity", "<" + "$var(new_pai)" + ">"); lookup("location", "sip:$var(username)@" + SIP_REALM_DEFINITION)); In module parameter definitions: modparam("nathelper", "sipping_from", "sip:keepalive@" + SIP_REALM_DEFINITION) So far, I have found this to be fairly consistent, except for use within functions. For example, string concatenation works for the second argument of xlog, but not for the string arguments of append_hf_value() or lookup(). Is there anyway to know this for each function beforehand, or is my only choice to go through trial and error here? Thanks! BR, George

3 4

Routing 10 Digit Inbound call to Registered User or VM server
by KamDev Essa 02 May '18

02 May '18

I have Kamailio 5.1 installed and running with 2 extensions, 201 and 202 registered. Both extensions can call each other perfectly. Next I am workin on inbound calls. I have an inbound call from a carrier that arrives perfectly but ends up in this area of the cfg case -3: send_reply("404", "Not Found"); exit; and sends a 404 not found. How do I : - Associate the 10 digit DID to the 201 or 202? Custom table I make or is there something native to Kamailio. - Relay the call to the 201 or 202. Thank in advance KamDev

1 0

Routing inbound call from carrier to user or voice mail
by KamDev Essa 02 May '18

02 May '18

I have Kamailio 5.1 installed and running with 2 extensions, 201 and 202 registered. Both extensions can call each other perfectly. I have an inbound call from a carrier that arrives perfectly but ends up in this area of the cfg case -3: send_reply("404", "Not Found"); exit; and sends a 404 not found. How do I : - Associate the 10 digit DID to the 201 or 202? - In the cfg, check if user is registered if - yes, relay call to user - no relay call to VM server. KamDev

2 1

Auto 100 Reply
by Duarte Rocha 02 May '18

02 May '18

Greetings, I have set the auto_inv_100() parameter from the TM module so i can get automatic "100" replies to the INVITE. However, in some cases i want to reply to the INVITE with provisional responses (181 for example) and i would like that "100" could be sent before the "181". In order to achieve this, on transactions that are going to be replied with 181 i'm turning off the parameter with t_set_auto_inv(0) and the sl_send_reply("100","Trying"). Is this the best way to do this? Thanks

2 1

I am not able to make a call from kam1 SIP Registered user to kam2 SIP Registered user.
by Suresh Talasaniya 02 May '18

02 May '18

Hello, Is it possible to configure two Kamailio servers with shared registration database? Any SIP UA can be registered on any server but it must transparently call another SIP UA which can be registered on the same or on another Kamailio server. *I configured usrloc module with below params.* ## ---------- usrloc params ------------------- #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) *modparam("usrloc", "db_mode", 3)* modparam("usrloc", "use_domain", MULTIDOMAIN) modparam("usrloc", "timer_interval", 30) modparam("usrloc", "matching_mode", 1) modparam("usrloc", "cseq_delay", 10) modparam("usrloc", "fetch_rows", 3000) modparam("usrloc", "hash_size", 12) modparam("usrloc", "timer_procs", 4) modparam("usrloc", "handle_lost_tcp", 1) modparam("usrloc", "close_expired_tcp", 1) modparam("usrloc", "db_timer_clean", 1) modparam("usrloc", "skip_remote_socket", 0) modparam("usrloc", "server_id_filter", 0) modparam("usrloc", "preload", "location") #!endif I am not able to make a call from kam1 SIP Registered user to kam2 SIP Registered user. it would be appreciable if anyone helps me to resolve the issue. Thank you! -- Regards, Suresh Talasaniya. Contact : +91-9724264776 Skype : suresh.talsaniya

2 1

Nat_UAC
by Nicolas Breuer 02 May '18

02 May '18

Hello The list, What do you think of the purpose of this function Nat_Uac_test('16') 16 - Test if the source port is different from the port in the "Via" header Correct me if I'm wrong but compare the source port of the message with the source port of the via is only valid on the initial invite ? All subsequent messages or replies don't list the UAC anywhere in the via. Thanks for your replies,

2 1

Working with users using external server
by Arik Halperin 02 May '18

02 May '18

Hello, I have a project for which I need to get user data(username and password) from an external server with REST APIs. What is the best way to achieve this? Thanks, Arik Halperin

2 1

Send UPDATE to inform changes in SDP
by Alfonso Pinto 30 Apr '18

30 Apr '18

Hi guys, I'm trying to setup a Call Forward No Reply with Kamailio and I've run into this problem (that was expected): When A INVITEs B, B replies with 183 and some SDP. B doesn't answer so Kamailio triggers the timeout and sends the INVITE TO C. C replies with 200 OK and some SDP, but A side drops the call as the media session doesn't match with the expected one. To solve it, I'm trying to send to A an UPDATE before relaying the 200 OK from C. I tried to use UAC module (uac_req_send + $uac_req) but I don't see how I can UPDATE the existing dialog. uac_req_send seems to try to setup a new dialog always. Is there a way to indicate to uac_req_send that I wan't to use the existing dialog? Or maybe there is another mechanism I can use to send the UPDATE inside the dialog? Regards, Alfonso.

4 5

Issue while using jsonrpc-s module in kamailio 4.4.
by Deepen Patel 30 Apr '18

30 Apr '18

Hi All, I am configuring Kamailio version 4.4 with jsonrpc-s and xhttp module. My aim is to use Kamailio commands using RPC from a remote server. I need help to solve error. After send rpc command, I got execution error in response. I am attaching code in kamailio.cfg file for jsonrp. event_route[xhttp:request] { if(src_ip!=127.0.0.1){ xhttp_reply("403", "Forbidden", "text/html", "<html><body>Not allowed from $si</body></html>" ); exit; } if ($hu =~ "^/RPC"){ xlog("json rpc dispatch"); # xlog($jsonrpl(method)); jsonrpc_exec('{"jsonrpc": "2.0", "method": "dispatcher.reload", "id": 1}'); jsonrpc_dispatch(); # xlog($jsonrpl(code)); # xlog($jsonrpl(text)); # xlog($jsonrpl(body)); xhttp_reply("200", "OK", "text/html", "<html><body> $jsonrpl(text), $jsonrpl(body)</body></html>"); } else { xhttp_reply("200", "OK", "text/html", "<html><body>Wrong URL $hu</body></html>"); } return; } Also, I am attaching logs. which could help you. If you can dream it, we will deliver it. <https://www.facebook.com/linteltechnologies/> <https://www.linkedin.com/company/7045679/> <http://blog.lintel.in/> Deepen Patel Python Developer A 507, Ashwamegh Elegance-2, Beside Central Mall, Ahmedabad - 380015 Gujarat, India P +1(281) 205-9247 M +91-898-077-3553 E deepen(a)lintelindia.com Skype deepenpatel.lintel W <http://lintel.in>http://lintel.in The science of today is Technology of tomorrow.

1 0

nathelper issues
by Konstantin Polyakov 29 Apr '18

29 Apr '18

Hello All, Unfortunately we found two more bugs in the nathelper module in my favorite Kamailio! Oh my god! 1. nathelper doesn't send pings when db_mode=3 and db is mongodb. I will provide fix soon. The root cause is incorrect date format in get_all_db_ucontacts function in usrloc. We need to use datatime type instead string. 2. nathelper doesn't send pings if ping_nated_only=0 (default value). It looks like incorrect condition in nh_timer in nathelper. Definitely lack of testing. It is interesting how testing is organized in the Kamailio community? Best regards. Konstantin ping_nated_only

1 0

User agent is registered but location table doesn't contain its contact.
by Konstantin Polyakov 28 Apr '18

28 Apr '18

Hello All, Hello Daniel! We have met a new issue. We use db_mode=1 ( Write-Through scheme. All changes to usrloc are immediately reflected in database too ). timer_interval has default value 60 seconds. Problem description. Some user agent is registered successfully in Kamailio registrar. Its contact exists inside internal cache (ul.lookup command shows it). But location table doesn't contain this contact. After several minutes (5-6 min) it becomes available in the location table too. The issue is reproduced periodically. Most of the time everything is ok. Is that expected behavior when internal cache is used? Did somebody else observed such situation? Is there some workaround? Best regards Konstantin

2 2

SIP Registartion Sharing between two kamailio
by Suresh Talasaniya 27 Apr '18

27 Apr '18

Hello All, I have two Kamailio servers and shared the database between two Kamailio. Below are my usrloc configuration in kamailio.cfg # ----- usrloc params ----- modparam("usrloc", "db_url", DBURL) *modparam("usrloc", "db_mode", 3)* modparam("usrloc", "use_domain", MULTIDOMAIN) modparam("usrloc", "timer_interval", 30) modparam("usrloc", "matching_mode", 1) modparam("usrloc", "cseq_delay", 10) modparam("usrloc", "fetch_rows", 3000) modparam("usrloc", "hash_size", 12) modparam("usrloc", "timer_procs", 4) modparam("usrloc", "handle_lost_tcp", 1) modparam("usrloc", "close_expired_tcp", 1) modparam("usrloc", "db_timer_clean", 1) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) modparam("registrar", "gruu_enabled", 0) modparam("registrar", "received_avp", "$avp(RECEIVED)") modparam("registrar", "default_expires", 3600) modparam("registrar", "min_expires", 1200) modparam("registrar", "max_expires", 3600) modparam("registrar", "default_q", 1000) modparam("registrar", "max_contacts", 10) modparam("registrar", "retry_after", 10) modparam("registrar", "xavp_cfg", "reg") modparam("registrar", "use_path", 1) modparam("registrar", "path_mode", 0) modparam("registrar", "path_use_received", 1) modparam("registrar", "contact_max_size", 1024) modparam("registrar", "path_check_local", 1) modparam("registrar", "sock_flag", 18) modparam("registrar", "sock_hdr_name", "Sock-Info") #------------------------------------------------------------ I have two users in same domain. 1. 1000(a)dev.domain.com --> Registered on kamailio 1 2. 1001(a)dev.domain.com --> Registered on kamailio 2 I am not able to make a call from 1000(a)dev.domain.com to 1001(a)dev.domain.com and vice versa. It will be appreciable if anyone help to resolve the issue. Thank you in advance! -- Regards, Suresh Talasaniya. Contact : +91-9724264776 Skype : suresh.talsaniya

2 1

Sending Notify With Custom Event Header
by strain17 27 Apr '18

27 Apr '18

Hi All, I'm trying to send a custom NOTIFY message to a list of users that are registered with Kamailio. The procedure would be invoked via HTTP (jsonrpc/xmlrpc over HTTP). 1. Should I be looking at the t_uac_start/t_uac_dlg MI commands to accomplish this? Is this the best way to do it if the NOTIFY sent to the phones are unsolicited? 2. I tried to use the xhttp_rpc module for testing out tm.t_uac_start but kept running into 400 invalid headers. I'm not sure if there is a syntax issue. The NOTIFY example is NOTIFY sip:123@192.168.5.5 . . From:333@1.1..1\r\nTo:test@2.2.2.2 - Request URL: http://x.x.x.x/http_rpc/tm/tm.t_uac_start?arg=NOTIFY+sip%3A123%40192.168.5.… - Request Method: GET - Status Code: 400 Invalid headers Thanks for any feedback!

2 1

I need some documentation :)
by Facchini Antoine 26 Apr '18

26 Apr '18

hello here ! I'm very newbie ... I search documentation (because I guess u can't explain that here like this) for create proxy. I want redirect all SIP traffic to an Asterisk instance. I search again and again but i don't find anything...

2 1

rare crash at kamailio 5.0.1
by David Escartin 26 Apr '18

26 Apr '18

Hello all last March 26th we experienced a crash on a 5.0.1 kamailio server and a core was generated. Seems that the call involved in the core generated was doing several retransmisions of a request (expected) and when it finally forwarded the request upstream, seems fter receiving a 481 from a-side, it crashed. here you have the backtrace <154>Mar 26 19:54:03 mad-proxy-inout-1 /usr/local/kamailio/sbin/kamailio[24992]: : <core> [core/mem/q_malloc.c:469]: qm_free(): BUG: qm_free: bad pointer 0x7f247acb5878 (out of memory block!) called from tm: h_table.c: free_cell_helper(187) - aborting (gdb) bt #0 0x00000036e8a32925 in raise () from /lib64/libc.so.6 #1 0x00000036e8a34105 in abort () from /lib64/libc.so.6 #2 0x0000000000654c24 in qm_free (qmp=0x7f23b9cba000, p=0x7f247acb5878, file=0x7f2479e3cd86 "tm: h_table.c", func=0x7f2479e3d070 "free_cell_helper", line=187, mname=0x7f2479e3c780 "tm") at core/mem/q_malloc.c:471 #3 0x00007f2479d6bf16 in free_cell_helper (dead_cell=0x7f23c54fea50, silent=0, fname=0x7f2479e4da80 "timer.c", fline=651) at h_table.c:187 #4 0x00007f2479db8148 in wait_handler (ti=1503818844, wait_tl=0x7f23c54fead0, data=0x7f23c54fea50) at timer.c:651 #5 0x0000000000629578 in timer_list_expire (t=1503818844, h=0x7f23b9d38dc0, slow_l=0x7f23b9d3b0c8, slow_mark=43533) at core/timer.c:874 #6 0x00000000006299d4 in timer_handler () at core/timer.c:939 #7 0x0000000000629e73 in timer_main () at core/timer.c:978 #8 0x0000000000422d4e in main_loop () at main.c:1690 #9 0x0000000000429732 in main (argc=13, argv=0x7fff2b9f8088) at main.c:2639 (gdb) frame 4 #4 0x00007f2479db8148 in wait_handler (ti=1503818844, wait_tl=0x7f23c54fead0, data=0x7f23c54fea50) at timer.c:651 651 UNREF_FREE(p_cell); (gdb) info locals p_cell = 0x7f23c54fea50 ret = 1 (gdb) p *p_cell $1 = {next_c = 0x0, prev_c = 0x0, hash_index = 28196, label = 341555146, flags = 4132, nr_of_outgoings = 2, ref_count = {val = 0}, from = { s = 0x7f23c87c79ed "From: <sip:932687220@100.100.100.177;user=phone;noa=national>;tag=50aa4a46-co11146-INS002\r\nCall-ID: b8997821-2814-46d3-b8f7-cf690f50f640\r\nCSeq: 1114601 BYE\r\nUser-Agent: ENSR3.0.67.10-IS2-RMRG1119-RG2040"..., len = 89}, callid = { s = 0x7f23c87c7a46 "Call-ID: b8997821-2814-46d3-b8f7-cf690f50f640\r\nCSeq: 1114601 BYE\r\nUser-Agent: ENSR3.0.67.10-IS2-RMRG1119-RG2040-CPI1-CPO2612\r\nContent-Length: 0\r\n\r\n", len = 47}, cseq_n = { s = 0x7f23c87c7a75 "CSeq: 1114601 BYE\r\nUser-Agent: ENSR3.0.67.10-IS2-RMRG1119-RG2040-CPI1-CPO2612\r\nContent-Length: 0\r\n\r\n", len = 13}, to = { s = 0x7f23c87c799e "To: \"34676893111\"<sip:34676893111@100.100.100.134;user=phone>;tag=64jNUpSZK8mSp\r\nFrom: <sip:932687220@100.100.100.177;user=phone;noa=national>;tag=50aa4a46-co11146-INS002\r\nCall-ID: b8997821-2814-46d3-b8f7"..., len = 79}, method = { s = 0x7f23c87c78c8 "BYE sip:34676893111@100.100.100.134:5080 SIP/2.0\r\nVia: SIP/2.0/UDP 200.200.200.78:5060;branch=z9hG4bK2d5c0cb4e11bb1503BYE50aa466c2b8a\r\nMax-Forwards: 34\r\nRoute: <sip:100.100.100.177;lr;ftag=64jNUpSZK8mSp;d"..., len = 3}, tmcb_hl = {first = 0x7f23bd9f7818, reg_types = 131714}, wait_timer = { next = 0x0, prev = 0x0, expire = 1503818844, initial_timeout = 80, data = 0x7f23c54fea50, f = 0x7f2479db8097 <wait_handler>, flags = 513, slow_idx = 0}, uas = {request = 0x7f23c87c71d8, end_request = 0x7f23c87c8288 "\300\300\300\300", response = {activ_type = 481, flags = 128, t_active = 0 '\000', branch = 0, buffer_len = 526, buffer = 0x7f23bef07b18 "SIP/2.0 481 Call Does Not Exist\r\nVia: SIP/2.0/UDP 200.200.200.78:5060;branch=z9hG4bK2d5c0cb4e11bb1503BYE50aa466c2b8a\r\nFrom: <sip:932687220@100.100.100.177;user=phone;noa=national>;tag=50aa4a46-co11146-I"..., my_T = 0x7f23c54fea50, timer = {next = 0x0, prev = 0x0, expire = 0, initial_timeout = 0, data = 0x0, f = 0x7f2479db7b4b <retr_buf_handler>, flags = 0, slow_idx = 0}, dst = {send_sock = 0x7f247a95be78, to = {s = { sa_family = 2, sa_data = "\023\304\325\300\313N\000\000\000\000\000\000\000"}, sin = {sin_family = 2, sin_port = 50195, sin_addr = { s_addr = 1321976021}, sin_zero = "\000\000\000\000\000\000\000"}, sin6 = {sin6_family = 2, sin6_port = 50195, sin6_flowinfo = 1321976021, sin6_addr = {__in6_u = {__u6_addr8 = '\000' <repeats 15 times>, __u6_addr16 = {0, 0, 0, 0, 0, 0, 0, 0}, __u6_addr32 = {0, 0, 0, 0}}}, sin6_scope_id = 0}}, id = 0, proto = 1 '\001', send_flags = {f = 0 '\000', blst_imask = 0 '\000'}}, retr_expire = 0, fr_expire = 0}, local_totag = { s = 0x0, len = 0}, cancel_reas = 0x0, status = 481}, uac = 0x7f23c54fec58, async_backup = {backup_route = 1, backup_branch = 4294967295, blind_uac = 0, ruri_new = 1}, fwded_totags = 0x0, uri_avps_from = 0x0, uri_avps_to = 0x0, user_avps_from = 0x0, user_avps_to = 0x0, domain_avps_from = 0x0, domain_avps_to = 0x0, xavps_list = 0x0, reply_mutex = {val = 0}, reply_locker_pid = {val = 0}, reply_rec_lock_level = 0, fr_timeout = 480, fr_inv_timeout = 1920, rt_t1_timeout_ms = 650, rt_t2_timeout_ms = 4000, end_of_life = 1503819200, relayed_reply_branch = 1, on_failure = 0, on_branch_failure = 0, on_reply = 0, on_branch = 0, on_branch_delayed = 0, md5 = 0x7f23c54fea50 ""} (gdb) frame 3 #3 0x00007f2479d6bf16 in free_cell_helper (dead_cell=0x7f23c54fea50, silent=0, fname=0x7f2479e4da80 "timer.c", fline=651) at h_table.c:187 187 sip_msg_free_unsafe(rpl); (gdb) info locals b = 0x0 i = 1 rpl = 0x7f247acb4fc0 tt = 0x0 foo = 0x7f23b9e2bd90 cbs = 0x0 cbs_tmp = 0x7f23c60c16b0 __FUNCTION__ = "free_cell_helper" (gdb) p *rpl $3 = {id = 2060145040, pid = 32548, tval = {tv_sec = 72, tv_usec = 139794652378712}, fwd_send_flags = {f = 1 '\001', blst_imask = 0 '\000'}, rpl_send_flags = { f = 0 '\000', blst_imask = 0 '\000'}, first_line = {type = -347, flags = 30263, len = 32548, u = {request = {method = {s = 0x2 <Address 0x2 out of bounds>, len = 1983381159}, uri = {s = 0x5 <Address 0x5 out of bounds>, len = 2}, version = {s = 0x2b <Address 0x2b out of bounds>, len = 2057435688}, method_value = 0}, reply = {version = {s = 0x2 <Address 0x2 out of bounds>, len = 1983381159}, status = {s = 0x5 <Address 0x5 out of bounds>, len = 2}, reason = {s = 0x2b <Address 0x2b out of bounds>, len = 2057435688}, statuscode = 0}}}, via1 = 0x0, via2 = 0x7f247acb5038, headers = 0x7fff2b9f3e50, last_header = 0x4, parsed_flag = 139794576763810, h_via1 = 0x1, h_via2 = 0x7f247616eba0, callid = 0x1, to = 0x270000000b, cseq = 0x1, from = 0x7f247aa1fa28, contact = 0x7f247acb5258, maxforwards = 0x7f247acb54d8, route = 0x0, record_route = 0x7fff2b9f3e50, content_type = 0x7f247acb5318, content_length = 0x0, authorization = 0x0, expires = 0x0, proxy_auth = 0x0, supported = 0x0, require = 0x0, proxy_require = 0x0, unsupported = 0x7f247acb5258, allow = 0x0, event = 0x7f247acb53d8, accept = 0x0, accept_language = 0x0, organization = 0x0, priority = 0x0, subject = 0x0, user_agent = 0x0, server = 0x0, content_disposition = 0x0, diversion = 0x0, rpid = 0x0, refer_to = 0x0, session_expires = 0x0, min_se = 0x0, sipifmatch = 0x0, subscription_state = 0x0, date = 0x0, identity = 0x0, identity_info = 0x0, pai = 0x0, ppi = 0x0, path = 0x7f247a98b6d0, privacy = 0x0, min_expires = 0x0, body = 0x7f247acb5258, eoh = 0x7fff2b9f3e50 "@f\313z$\177", unparsed = 0x0, rcv = {src_ip = {af = 0, len = 0, u = {addrl = {0, 0}, addr32 = {0, 0, 0, 0}, addr16 = {0, 0, 0, 0, 0, 0, 0, 0}, addr = '\000' <repeats 15 times>}}, dst_ip = {af = 3233857728, len = 0, u = {addrl = {2882400237, 0}, addr32 = {2882400237, 0, 0, 0}, addr16 = {61421, 43981, 0, 0, 0, 0, 0, 0}, addr = "\355\357ͫ", '\000' <repeats 11 times>}}, src_port = 0, dst_port = 0, proto_reserved1 = 0, proto_reserved2 = 536, src_su = {s = {sa_family = 0, sa_data = "\000\000 \266\230z$\177\000\000\030\002\000"}, sin = {sin_family = 0, sin_port = 0, sin_addr = {s_addr = 2056828448}, sin_zero = "$\177\000\000\030\002\000"}, sin6 = {sin6_family = 0, sin6_port = 0, sin6_flowinfo = 2056828448, sin6_addr = {__in6_u = {__u6_addr8 = "$\177\000\000\030\002\000\000\000\000\000\000\210T\313z", __u6_addr16 = {32548, 0, 536, 0, 0, 0, 21640, 31435}, __u6_addr32 = {32548, 536, 0, 2060145800}}}, sin6_scope_id = 32548}}, bind_address = 0x7f247616ebb0, proto = 32 ' '}, buf = 0x7f247616eba6 "rtpengine", len = 139, new_uri = {s = 0xf0f0f0f0 <Address 0xf0f0f0f0 out of bounds>, len = 2060145720}, dst_uri = { s = 0x20 <Address 0x20 out of bounds>, len = 2060144576}, parsed_uri_ok = 1, parsed_uri = {user = { s = 0x7f247637fead "7:packetsi32e5:bytesi1896e6:errorsi0eeee5:flagsl11:initialized4:send4:recveeeee6:totalsd3:RTPd7:packetsi4951e5:bytesi158432e6:errorsi0ee4:RTCPd7:packetsi32e5:bytesi1896e6:errorsi0eee6:result2:oke", len = 2}, passwd = { s = 0x7f247637feaf "packetsi32e5:bytesi1896e6:errorsi0eeee5:flagsl11:initialized4:send4:recveeeee6:totalsd3:RTPd7:packetsi4951e5:bytesi158432e6:errorsi0ee4:RTCPd7:packetsi32e5:bytesi1896e6:errorsi0eee6:result2:oke", len = 7}, host = {s = 0x900000002 <Address 0x900000002 out of bounds>, len = 1}, port = { s = 0x7f247acb5038 "\004", len = 0}, params = {s = 0x0, len = 2060145336}, sip_params = {s = 0x7fff2b9f3e50 "@f\313z$\177", len = 2}, headers = { s = 0x7f247637feb6 "i32e5:bytesi1896e6:errorsi0eeee5:flagsl11:initialized4:send4:recveeeee6:totalsd3:RTPd7:packetsi4951e5:bytesi158432e6:errorsi0ee4:RTCPd7:packetsi32e5:bytesi1896e6:errorsi0eee6:result2:oke", len = 4}, port_no = 0, proto = 0, type = ERROR_URI_T, flags = 0, transport = { s = 0x400000001 <Address 0x400000001 out of bounds>, len = 32}, ttl = {s = 0x7f247acb5038 "\004", len = 0}, user_param = {s = 0x0, len = 2060145432}, maddr = {s = 0x7fff2b9f3e50 "@f\313z$\177", len = 1}, method = { s = 0x7f247637feba "5:bytesi1896e6:errorsi0eeee5:flagsl11:initialized4:send4:recveeeee6:totalsd3:RTPd7:packetsi4951e5:bytesi158432e6:errorsi0ee4:RTCPd7:packetsi32e5:bytesi1896e6:errorsi0eee6:result2:oke", len = 2}, lr = { s = 0x7f247637febc "bytesi1896e6:errorsi0eeee5:flagsl11:initialized4:send4:recveeeee6:totalsd3:RTPd7:packetsi4951e5:bytesi158432e6:errorsi0ee4:RTCPd7:packetsi32e5:bytesi1896e6:errorsi0eee6:result2:oke", len = 5}, r2 = {s = 0x700000002 <Address 0x700000002 out of bounds>, len = 1522079939}, gr = { s = 0x7f247acb5038 "\004", len = 0}, transport_val = {s = 0x0, len = 2060145528}, ttl_val = {s = 0x7fff2b9f3e50 "@f\313z$\177", len = 2}, user_param_val = { s = 0x7f247637fec1 "i1896e6:errorsi0eeee5:flagsl11:initialized4:send4:recveeeee6:totalsd3:RTPd7:packetsi4951e5:bytesi158432e6:errorsi0ee4:RTCPd7:packetsi32e5:bytesi1896e6:errorsi0eee6:result2:oke", len = 6}, maddr_val = {s = 0x0, len = 0}, method_val = {s = 0x600000001 <Address 0x600000001 out of bounds>, len = 1896}, lr_val = {s = 0x7f247acb5038 "\004", len = 0}, r2_val = {s = 0x0, len = 2060145624}, gr_val = {s = 0x7fff2b9f3e50 "@f\313z$\177", len = 1}}, parsed_orig_ruri_ok = 1983381191, parsed_orig_ruri = {user = {s = 0x2 <Address 0x2 out of bounds>, len = 1983381193}, passwd = { s = 0x6 <Address 0x6 out of bounds>, len = 2}, host = {s = 0x0, len = 2060144696}, port = {s = 0x0, len = 0}, params = {s = 0x7f247acb54d8 "\002", ---Type <return> to continue, or q <return> to quit--- len = 731856464}, sip_params = {s = 0x0, len = 0}, headers = {s = 0x0, len = 0}, port_no = 49344, proto = 49344, type = ERROR_URI_T, flags = 2882400237, transport = {s = 0x0, len = 0}, ttl = {s = 0x218 <Address 0x218 out of bounds>, len = 2060144520}, user_param = {s = 0x218 <Address 0x218 out of bounds>, len = 2060146440}, maddr = {s = 0x7f247616ebb0 "rtpengine: bencode.c", len = 1981214752}, method = {s = 0x7f247616eba6 "rtpengine", len = 139}, lr = { s = 0xf0f0f0f0 <Address 0xf0f0f0f0 out of bounds>, len = 2060146360}, r2 = {s = 0x20 <Address 0x20 out of bounds>, len = 2060145216}, gr = { s = 0x2 <Address 0x2 out of bounds>, len = 1983381199}, transport_val = {s = 0x3 <Address 0x3 out of bounds>, len = 0}, ttl_val = {s = 0x0, len = 1}, user_param_val = {s = 0x0, len = 2060144696}, maddr_val = {s = 0x0, len = 0}, method_val = {s = 0x0, len = 731856464}, lr_val = { s = 0x1 <Address 0x1 out of bounds>, len = 1983381205}, r2_val = {s = 0x2 <Address 0x2 out of bounds>, len = 1983381207}, gr_val = { s = 0x5 <Address 0x5 out of bounds>, len = 2}}, add_rm = 0x7f247acb5878, body_lumps = 0x7f247aa43300, reply_lump = 0x0, add_to_branch_s = "\000\000\000\000\000\000\000\000\230U\313z$\177\000\000P>\237+\377\177\000\000\003\000\000\000\000\000\000\000\244\353\026v$\177\000\000\001\000\000\000\000\000\000\000\240\353\026v$\177\000\000\001", add_to_branch_len = 0, hash_index = 8, msg_flags = 28, flags = 2060162240, set_global_address = { s = 0x7f247aa43300 "\004", len = 2060146168}, set_global_port = {s = 0x7f247acb5758 "\001", len = 0}, force_send_socket = 0x7fff2b9f3e50, path_vec = { s = 0x1 <Address 0x1 out of bounds>, len = 1983381213}, instance = {s = 0x3 <Address 0x3 out of bounds>, len = 1983381216}, reg_id = 11, ruid = { s = 0xe00000002 <Address 0xe00000002 out of bounds>, len = 0}, location_ua = {s = 0x7f247acb5598 "\003", len = 0}, ldv = {flow = {decoded = 0, rcv = { src_ip = {af = 2060146264, len = 32548, u = {addrl = {140733925244496, 30064771073}, addr32 = {731856464, 32767, 1, 7}, addr16 = {15952, 11167, 32767, 0, 1, 0, 7, 0}, addr = "P>\237+\377\177\000\000\001\000\000\000\a\000\000"}}, dst_ip = {af = 1983381227, len = 32548, u = {addrl = {2, 139794578931437}, addr32 = {2, 0, 1983381229, 32548}, addr16 = {2, 0, 0, 0, 65261, 30263, 32548, 0}, addr = "\002\000\000\000\000\000\000\000\355\376\067v$\177\000"}}, src_port = 4, dst_port = 0, proto_reserved1 = 0, proto_reserved2 = 2, src_su = { s = {sa_family = 6, sa_data = "\000\000P>\237+\377\177\000\000\230U\313z"}, sin = {sin_family = 6, sin_port = 0, sin_addr = {s_addr = 731856464}, sin_zero = "\377\177\000\000\230U\313z"}, sin6 = {sin6_family = 6, sin6_port = 0, sin6_flowinfo = 731856464, sin6_addr = {__in6_u = { __u6_addr8 = "\377\177\000\000\230U\313z$\177\000\000\000\000\000", __u6_addr16 = {32767, 0, 21912, 31435, 32548, 0, 0, 0}, __u6_addr32 = { 32767, 2060146072, 32548, 0}}}, sin6_scope_id = 0}}, bind_address = 0x0, proto = 88 'X'}}}} (gdb) (gdb) bt full #0 0x00000036e8a32925 in raise () from /lib64/libc.so.6 No symbol table info available. #1 0x00000036e8a34105 in abort () from /lib64/libc.so.6 No symbol table info available. #2 0x0000000000654c24 in qm_free (qmp=0x7f23b9cba000, p=0x7f247acb5878, file=0x7f2479e3cd86 "tm: h_table.c", func=0x7f2479e3d070 "free_cell_helper", line=187, mname=0x7f2479e3c780 "tm") at core/mem/q_malloc.c:471 qm = 0x7f23b9cba000 f = 0x7f23bc87a2c8 size = 528 next = 0x400 prev = 0x7fff2b9f7920 __FUNCTION__ = "qm_free" #3 0x00007f2479d6bf16 in free_cell_helper (dead_cell=0x7f23c54fea50, silent=0, fname=0x7f2479e4da80 "timer.c", fline=651) at h_table.c:187 b = 0x0 i = 1 rpl = 0x7f247acb4fc0 tt = 0x0 foo = 0x7f23b9e2bd90 cbs = 0x0 cbs_tmp = 0x7f23c60c16b0 __FUNCTION__ = "free_cell_helper" #4 0x00007f2479db8148 in wait_handler (ti=1503818844, wait_tl=0x7f23c54fead0, data=0x7f23c54fea50) at timer.c:651 p_cell = 0x7f23c54fea50 ret = 1 #5 0x0000000000629578 in timer_list_expire (t=1503818844, h=0x7f23b9d38dc0, slow_l=0x7f23b9d3b0c8, slow_mark=43533) at core/timer.c:874 tl = 0x7f23c54fead0 ret = 0 #6 0x00000000006299d4 in timer_handler () at core/timer.c:939 saved_ticks = 1503818844 run_slow_timer = 0 i = 525 __FUNCTION__ = "timer_handler" #7 0x0000000000629e73 in timer_main () at core/timer.c:978 No locals. #8 0x0000000000422d4e in main_loop () at main.c:1690 i = 1 pid = 0 si = 0x0 si_desc = "udp receiver child=0 sock=100.100.100.169:5060", '\000' <repeats 16 times>, "\001\000\000\000\340|\237+\377\177\000\000W\327e\000\000\000\000\000\200\335u\000\000\000\000\000\250\065\313zo\000\000\000\316\357u\000\000\000\000\000\205\335u\000\000\000\000\000\004\000\000\000\000\000\000\000\000\240˹#\177---Type <return> to continue, or q <return> to quit--- \000" nrprocs = 1 woneinit = 1 __FUNCTION__ = "main_loop" #9 0x0000000000429732 in main (argc=13, argv=0x7fff2b9f8088) at main.c:2639 cfg_stream = 0x1f8f010 c = -1 r = 0 tmp = 0x7fff2b9f8742 "" tmp_len = 54 port = -398323320 proto = 54 options = 0x732e78 ":f:cm:M:dVIhEeb:l:L:n:vKrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:x:X:Y:" ret = -1 seed = 3083103015 rfd = 4 debug_save = 0 debug_flag = 0 dont_fork_cnt = 0 n_lst = 0x36e8a90687 p = 0x7fff2b9f7f88 "\200\200\237+\377\177" st = {st_dev = 2307, st_ino = 9183277, st_nlink = 3, st_mode = 16832, st_uid = 522, st_gid = 501, __pad0 = 0, st_rdev = 0, st_size = 4096, st_blksize = 4096, st_blocks = 8, st_atim = {tv_sec = 1496231473, tv_nsec = 0}, st_mtim = {tv_sec = 1520405673, tv_nsec = 0}, st_ctim = { tv_sec = 1520405673, tv_nsec = 0}, __unused = {0, 0, 0}} __FUNCTION__ = "main" (gdb) i dont know if you can help us to know the reason of the crash, because the call trace seemed quite normal, and we don't know if the cause was from the rtpengine responses of the bye requests, or was caused by the request retransmisions, which would be rare because we sometimes have those type of calls. thanks a lot and regards david

2 5

ASN.1 HI2- IRI-Parameters
by Mojtaba 26 Apr '18

26 Apr '18

Hello. I need to use HI2Operations for transfering IRI-Parameters in kamailio.Dose anybody have experience in this regards? Thanks. Mojtaba

2 1

Possibility of kemi lua to keep lua global vars
by Yuriy Gorlichenko 26 Apr '18

26 Apr '18

Hi For now as I see: every request that calls lua script reevables my global lua variables For example I trying to connect to DB via luasql driver and checking if connection already existis to avoid new connection I have function init() that calls in ksr_request_route() Inside i have ome check: if not CON then CON = assert(MYSQL:connect(DBdata.dbname,DBdata.username,DBdata.password,DBdata.host,DBdata.port)) if not CON then KSR.log("info","|init|log| DB NOT connected\n") KSR.sl.sl_send_reply(503,"Internal server error") KSR.x.exit() else KSR.log("info","|init|log| DB connected\n") end end So in this case if CON was already created it should not try to connect But for now it tries for every request. So means kemi not stores global lua variables between sessions (in one kamailio process) Also as I know INIT routes initiates before lua machine will be started so it not allowed to init some lua global variables in the init routes So my question is there is a way to keep global vars per kamailio process instead per kamailio request?

2 1

Sha2 self-signed certs config in Kamilio
by Kiran Gaddam 25 Apr '18

25 Apr '18

Hello All, I want to enable certificate validation on the server. I am only using self-signed certs. I have the same cert/key in the client and server and want to only allow connection from clients with this cert/key. I have turned on the following in tls.cfg and done all the steps required in kamailio.cfg file. But it's failed to verify certs and allowing the clients which doesn't have same certs. Please help to configure the cert/key in right way. [server:default] method = TLSv1 verify_certificate = yes require_certificate = yes private_key = /usr/local/etc/kamailio/selfsigned.key certificate = /usr/local/etc/kamailio/selfsigned.pem ca_list = /usr/local/etc/sip-router/cacert.pem [client:default] verify_certificate = yes require_certificate = yes In advance Thank you. Thanks, Kiran

2 2

TLS support and certificate validation
by Jignesh Gandhi 24 Apr '18

24 Apr '18

I want to enable certificate validation on the server. I am only using self-signed certs. I have the same certificate/key in the client and server and want to only allow connection from clients with this cert/key. I have turned on the following in tls.cfg and done all the steps required in kamailio.cfg file. [server:default] method = TLSv1 verify_certificate = yes require_certificate = yes private_key = /usr/local/etc/kamailio/selfsigned.key certificate = /usr/local/etc/kamailio/selfsigned.pem ca_list = /usr/local/etc/sip-router/cacert.pem [client:default] verify_certificate = yes require_certificate = yes Any help is appreciated. Thanks, --Jignesh

2 1

TCP packet relay delayed (Kamailio 4.1.5)
by Andrew Chen 24 Apr '18

24 Apr '18

Hi Kamailio community, I need your urgent help regarding a situation where the Kamailio TCP packet delayed in relaying SIP packets. They delay between the time Kamailio logs the relay to the time it left the system was ~20 seconds difference, which is huge in SIP world. Any ideas, thoughts or suggestions on how I can narrow down to the problem? We have about 30 cases of this today Thanks in advance. -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

3 7

Diversion Headers
by Duarte Rocha 24 Apr '18

24 Apr '18

Greetings, Does the function "add_diversion() present in the "Diversion" module allows me to introduce other parameters (privacy, counter, etc) or is it only for the "diversion-reason"? If it doesn't let me do that, how can I add the Diversion with insert_hf() and make sure the headers are added in the right order? Cheers

2 1

Any ideas on how to implement prepaid calling function which is persistent after restart
by Kit Lim 24 Apr '18

24 Apr '18

Hi Everyone, My project involves keeping track of the allowable duration for the caller to talk, and keeping track of the status of the call with the back end database. Currently i have implemented using the following modules (aside from the default modules in the sample config script). * http_async_client.so - to communicate with the backend DB * dialog.so - mainly use to detect the 200 ok answer in relation to invite, and the end of the call * cnxcc.so - to release the call when the allowable talk time is reached The flow is as below:- Invite -- > http to backend DB to ask for talk time <-- DB returns value 200 ok (answer) ---> http backend DB to update that the call has answered <-- DB returns ok dialog end or Cnxcc terminate --> http backend DB to update that the call has hangup My problem is that if there is a need to restart the kamailio config there is no 'graceful' way to hangup all the calls and inform the backend, or deny any new calls from coming in. If kamailio is restarted then cnxcc would have forgotten that there is a maximum talk time allowed to be called or recover from the restart. How to I make the system resilience to failures/restarts. Appreciate any good suggestions. regards CK

2 1

Re: [SR-Users] Kamailio doesn't start with db_mongodb and usrloc.
by Konstantin Polyakov 24 Apr '18

24 Apr '18

Dear All, It looks like there is a serious error inside db_mongodb module in the latest versions of Kamailio 4.4.7 and 5.1.2. During query operations string fields inside result structure db1_res_t could be corrupted because of incorrect usage of bson memory. The issue is located in the function db_mongodb_convert_bson() around line 614. This issue is reproduced when we need to read more than 150 documents from the database. So that is why this issue still exists. Firstly bson_iter_utf8 function returns pointer to memory which is allocated inside bson memory and which is used multiple times. So the data must be copied into Kamailio memory instead storing that pointer. The pointer becomes invalid after some time. Secondly, this pointer must not be freed by the caller of bson_iter_utf8 . At the current version bson allocated pointer is freed by Kamailio function pkg_free like it is was allocated by pkg_malloc . So I am going to create ticket in the github and propose the fix. It looks like nobody uses mongodb with Kamailio in db_mode=1 :) Best regards Konstantin >---------------------------------------------------------------------- > >Message: 1 >Date: Sat, 21 Apr 2018 23:37:26 +0300 >From: Konstantin Polyakov < piligrim_pk(a)mail.ru > >To: sr-users(a)lists.kamailio.org >Subject: [SR-Users] Kamailio doesn't start with db_mongodb and usrloc. >Message-ID: < 1524343046.238272820(a)f356.i.mail.ru > >Content-Type: text/plain; charset="utf-8" > > >Hello! > >Similar issue but with different symptoms. > >kamailio 4.4.7 (tried to use db_mongodb module from 5.1.2) >usrloc is configured to use db_mode=1 and db_insert_null=1. > >MongoDB 3.6.3 >mongo-c-driver was tested 1.9.2 and 1.9.4. > >location table contains 300 records. > >The issue was not reproduced since February when we started a new feature. Now it is reproduced constantly, but on some machines. > >Logs: > >[0;39;49m[0;39;49m 1(13958) DEBUG: usrloc [udomain.c:313]: dbrow2info(): non-local socket <all.com>...ignoring >[0;39;49m[0;33;49m 1(13958) CRITICAL: usrloc [udomain.c:476]: preload_udomain(): empty username record in table location...skipping >[0;39;49m[0;33;49m 1(13958) CRITICAL: usrloc [udomain.c:256]: dbrow2info(): bad callid >[0;39;49m[0;31;49m 1(13958) ERROR: usrloc [udomain.c:484]: preload_udomain(): skipping record for agent in table location >[0;39;49m[0;33;49m 1(13958) CRITICAL: usrloc [udomain.c:476]: preload_udomain(): empty username record in table location...skipping >[0;39;49m[0;33;49m 1(13958) CRITICAL: usrloc [udomain.c:231]: dbrow2info(): bad contact >[0;39;49m[0;31;49m 1(13958) ERROR: usrloc [udomain.c:484]: preload_udomain(): skipping record for cflags in table location >[0;39;49m[0;39;49m 1(13958) DEBUG: usrloc [udomain.c:313]: dbrow2info(): non-local socket <��keepalive>...ignoring >[0;39;49m[0;35;49m 1(13958) WARNING: usrloc [../../ut.h:717]: shm_str_dup(): shm_str_dup fallback; dup called for src->s == NULL or src->len < 0 >[0;39;49m[0;35;49m 1(13958) WARNING: usrloc [../../ut.h:731]: shm_str_dup(): shm_str_dup fallback; skip memcpy for src->s == NULL >[0;39;49m[0;33;49m 1(13958) CRITICAL: usrloc [udomain.c:476]: preload_udomain(): empty username record in table location...skipping >[0;39;49m[0;33;49m 1(13958) CRITICAL: usrloc [udomain.c:231]: dbrow2info(): bad contact >[0;39;49m[0;31;49m 1(13958) ERROR: usrloc [udomain.c:484]: preload_udomain(): skipping record for cseq in table location >[0;39;49m[0;33;49m 1(13958) CRITICAL: usrloc [udomain.c:231]: dbrow2info(): bad contact >[0;39;49m[0;31;49m 1(13958) ERROR: usrloc [udomain.c:484]: preload_udomain(): skipping record for 5.17.xxx in table location >[0;39;49m[0;33;49m 1(13958) CRITICAL: usrloc [udomain.c:231]: dbrow2info(): bad contact >[0;39;49m[0;31;49m 1(13958) ERROR: usrloc [udomain.c:484]: preload_udomain(): skipping record for .17.x in table location >[0;39;49m[0;33;49m 1(13958) CRITICAL: usrloc [udomain.c:476]: preload_udomain(): empty username record in table location...skipping >[0;39;49m[0;39;49m 1(13958) DEBUG: usrloc [udomain.c:313]: dbrow2info(): non-local socket <nce>...ignoring >[0;39;49m[0;35;49m 1(13958) WARNING: usrloc [../../ut.h:717]: shm_str_dup(): shm_str_dup fallback; dup called for src->s == NULL or src->len < 0 >[0;39;49m[0;35;49m 1(13958) WARNING: usrloc [../../ut.h:731]: shm_str_dup(): shm_str_dup fallback; skip memcpy for src->s == NULL >[0;39;49m[0;33;49m 1(13958) CRITICAL: usrloc [udomain.c:476]: preload_udomain(): empty username record in table location...skipping >[0;39;49m[0;33;49m 1(13958) CRITICAL: usrloc [udomain.c:476]: preload_udomain(): empty username record in table location...skipping >[0;39;49m[0;33;49m 1(13958) CRITICAL: usrloc [udomain.c:476]: preload_udomain(): empty username record in table location...skipping >[0;39;49m[0;39;49m 1(13958) DEBUG: usrloc [udomain.c:313]: dbrow2info(): non-local socket <c-89-5ada654b-35a2-42>...ignoring >[0;39;49m[0;39;49m 1(13958) DEBUG: usrloc [udomain.c:313]: dbrow2info(): non-local socket <>...ignoring >[0;39;49m[0;33;49m 1(13958) CRITICAL: usrloc [udomain.c:231]: dbrow2info(): bad contact >[0;39;49m[0;31;49m 1(13958) ERROR: usrloc [udomain.c:484]: preload_udomain(): skipping record for 38.146:5360 in table location >[0;39;49m[0;39;49m 1(13958) DEBUG: usrloc [udomain.c:313]: dbrow2info(): non-local socket <ified>...ignoring >[0;39;49m[0;31;49m 1(13958) ERROR: usrloc [ucontact.c:94]: new_ucontact(): no ruid for aor: lQ.com@5.17.60@8/Ղusername >[0;39;49m[0;31;49m 1(13958) ERROR: usrloc [urecord.c:149]: mem_insert_ucontact(): failed to create new contact >[0;39;49m[0;31;49m 1(13958) ERROR: usrloc [udomain.c:517]: preload_udomain(): inserting contact failed >[0;39;49m[0;31;49m 1(13958) ERROR: usrloc [ul_mod.c:457]: child_init(): child(1): failed to preload domain 'location' >[0;39;49m[0;31;49m 1(13958) ERROR: <core> [sr_module.c:923]: init_mod_child(): Error while initializing module usrloc (/usr/ecsp/lib64/kamailio/modules/usrloc.so) >[0;39;49m[0;31;49m 1(13958) ERROR: <core> [pt.c:329]: fork_process(): init_child failed for process 1, pid 13958, "udp receiver child=0 sock=5.17.138.137:5060 (5.17.138.137:5060)" >[0;39;49m[0;33;49m 1(13958) CRITICAL: <core> [main.c:1615]: main_loop(): Cannot fork >[0;39;49m[1;31;49m 0(13956) ALERT: <core> [main.c:740]: handle_sigs(): child process 13958 exited normally, status=255 >[0;39;49m[0;36;49m 0(13956) INFO: <core> [main.c:768]: handle_sigs(): terminating due to SIGCHLD >[0;39;49m[0;39;49m 0(13956) DEBUG: <core> [main.c:770]: handle_sigs(): terminating due to SIGCHLD > > >Could you please help us to resolve the issue? >Thank you in advance. >Best regards >Konstantin >

1 0

UAC with HA1
by Ben Hood 24 Apr '18

24 Apr '18

Hi, Does the UAC module support ha1 passwords for the remote registrar? The DDL for the uacreg table indicates the might be the case, but it’s unclear which one the UAC module would use by default. Is there a parameter to set this? Cheers, Ben

2 1

Planning the release of Kamailio v5.1.3
by Daniel-Constantin Mierla 24 Apr '18

24 Apr '18

Hello, I am considering to package a new release from branch 5.1, respectively v5.1.3 sometime next week, likely on Tuesday or Wednesday (April 24 or 25). I any issue is not yet reported on github tracker, do it as soon as possible to have a chance to be investigated. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

1 1

Use of MariaDB to replace MySQL
by Ginhoux, Patrick 23 Apr '18

23 Apr '18

Hi, Currently, my Kamailio proxy uses a MySql DB to get configuration data. We have a plan to migrate MySQL to MariaDB. Is there MariaDB modules with Kamailio ? Thanks in advance for your answer. P. Ginhoux

2 2

Kamailio not listening to proper ipv6 address
by selak Dev 23 Apr '18

23 Apr '18

Hi, I need to enable Kamailio to work with ipv6 also, and I enabled auto_bind_ipv6=1 and when restarted I can see that it is listening to ipv6 ip addresses also. The problem is that it stripped the ip address at the end and it is listening to the wrong one. When I try to register I get timeout, since it is not listening the correct address. ip address of the server is 2604:a880:400:d0::yyyy:xxxx and when I execute "sudo netstat -plnt" too see which app is listening to which address/port I can see: .... tcp6 0 0 2604:a880:400:d0:::5085 :::* LISTEN 17748/kamailio tcp6 0 0 2604:a880:400:d0:::5061 :::* LISTEN 17748/kamailio It somehow stripped the last part of the address (yyyy:xxxx). I dont know if it is related to :: which would be all zeros, but it doesn't work even if i explicitly specify which address to listen with listen=[2604:a880:400:d0:0000:0000:yyyy:xxxx] or listen=tcp:[2604:a880:400:d0:0000:0000:yyyy:xxxx]:5061 or listen=tcp:[2604:a880:400:d0::yyyy:xxxx]:5061 I tried all combinations and the results are always the same... Have anybody had this problem before? Do you know how to fix this? Best regards, Marko Selakovic

3 3

select outgoing interface
by Laurent Schweizer 23 Apr '18

23 Apr '18

Dear all, I have a config with multiple interface( multiple IP in same subnet) and want to know how I can select the interface used to send a request (UDP). The best is the possibility to choose for each request the outgoing interface but if it's not possible I think I can also work with a solution where the outgoing interface is defined globally ? I need this because one provider request one IP for inbound call and one IP for outbound call . Thanks Laurent

2 2

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