sr-users May 2017

sr-users@lists.kamailio.org

91 participants
119 discussions

Dispatcher - no destination sets
by Jean Cérien 24 May '17

24 May '17

Hello I am working with Kamaiio 5.0 on the dispatcher, with the default dispatcher.cfg file ( https://github.com/kamailio/kamailio/blob/master/src/modules/dispatcher/doc… ) I've added via mysql a gateway: mysql> select * from dispatcher \G; *************************** 1. row *************************** id: 1 setid: 1 destination: 192.168.2.20 flags: 0 priority: 0 attrs: description: local gw 1 row in set (0.00 sec) Kamctl sees it correctly: kamctl dispatcher show dispatcher gateways +----+-------+--------------+-------+----------+-------+-------------+ | id | setid | destination | flags | priority | attrs | description | +----+-------+--------------+-------+----------+-------+-------------+ | 1 | 1 | 192.168.2.20 | 0 | 0 | | local gw | +----+-------+--------------+-------+----------+-------+-------------+ Here is the piece of cfg: if(ds_list_exist(1)) { xlog("L_INFO", "----------- data in set 1"); } else { xlog("L_INFO", "----------- NO data in set 1"); } if(!ds_select_dst("1", "4")) { send_reply("404", "No destination"); exit; } Yet, when an INVITE arrives, I get the following errors: DEBUG: dispatcher [dispatcher.c:876]: w_ds_list_exist(): --- Looking for dispatcher set 1 DEBUG: dispatcher [dispatch.c:1314]: ds_list_exist(): -- Looking for set 1 INFO: dispatcher [dispatch.c:1320]: ds_list_exist(): destination set [1] not found INFO: <script>: ----------- NO data in set 1 ERROR: dispatcher [dispatch.c:1679]: ds_select_dst_limit(): no destination sets same result wether I state the dataset as 1 or "1" - Mysql table column type is int(11). What could be wrong ? Thanks for the help J.

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Kamailio 4.4 crashing
by Pete Kelly 24 May '17

24 May '17

Hi I have Kamailio 4.4 proxying a "normal" call and am finding that it is crashing consistently when it tries to proxy the 200OK from a BYE request when dealing with a certain endpoint. The SIP looks very normal to me and I've managed to obtain a trace, full debug logs and a core dump of the issue. I can provide full copies of the files but the interesting parts from the logfile are here: https://pastebin.com/QkwVMxJj bt output from the coredump is here: https://pastebin.com/auNRaeaU Can anyone please advise?

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can kamailio+rtpproxy traversal any kind of nat
by 赵国杰 24 May '17

24 May '17

Hi everyone, Right now I'm running kamailio and rtpproxy on a server behind nat with a public ip. I enabled the advertise address. If i registered two clients via my home wifi or 3g/4g network, then everything's fine. However, if i start the two clients via my office wifi, two clients can calll each other but with no audio. Obviously, the rtp media session has trouble traversal the office LAN nat. I have no idea what went wrong. Anyone can point me the right direction? Sincerely, jesse

1 0

tsilo module enhancement
by Jayesh Nambiar 24 May '17

24 May '17

Hello, I see the tsilo module works only with registered contacts using location table. Is there a possibility that the ts_append function can actually take the destination URI as an argument and append branch to respective destination? Basically I want to avoid maintaining location table. In a typical mobile use case, where the incoming call is suspended and a push notification is generated to the called party; the called party generates a REGISTER request with expires header as 0; kamailio just extracts the contact header and sends the call. Problem comes in when the same user receives a PUSH on another device; I cannot figure a neat way to append the same call on the new device. Is there a better alternative to achieve this without looking up the location table. Thanks for any help. - Jayesh

2 2

R-URI for ACK
by David Wilson 24 May '17

24 May '17

Hello All, I’m trying to interop with a provider whose ACK during an inbound call seems to be misrouted. During the INVITE, OK, ACK handshake, my understanding (which matches the examples in RFC 3665) is that the R-URI in the ACK should be the Contact specified in the 200 OK. I have also found multiple discussions that imply that this is necessary. However, I can’t find a convincing normative reference for this... Is there a reference (ideally in RFC 3261) that makes this clear? Cheers, Dave.

1 0

Kamailio and XMLRPC commands with POST
by Olli Attila 24 May '17

24 May '17

Hi, I'm using kamailio 5.0.0 and trying to use core commands through xmlrpc interface but the requests are failing towards Kamailio. I have compiled and enabled the xmlrpc module in config and I have this on my kamailio.cfg: # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif I have tried to test this by using curl and calling "core.uptime" method in xml body (core.uptime works when executed through kamcmd commandline tool). The curl is called from the same machine that I have Kamailio installed on: "curl -X POST -H 'Host: 127.0.0.1:5060' -H 'User-Agent: xmlrpcclient' -H 'Content-Type: text/xml' -d '<?xml version='1.0'?><methodCall><methodName>core.uptime</methodName><params></params></methodCall>' http://127.0.0.1:5060" This fails and I only get: "curl: (7) Failed to connect to 127.0.0.1 port 5060: Connection refused" network traffic shows me this.. firewall is disabled completely when doing the test: 1 0.000000 127.0.0.1 -> 127.0.0.1 TCP 74 55630→5060 [SYN] Seq=0 Win=43690 Len=0 MSS=65495 SACK_PERM=1 TSval=1347930477 TSecr=0 WS=64 2 0.000032 127.0.0.1 -> 127.0.0.1 TCP 54 5060→55630 [RST, ACK] Seq=1 Ack=1 Win=0 Len=0 Can anyone help me to get to the right direction with this? Cheers, --Olli

5 10

rtpengine opening second RTP port
by John Petrini 23 May '17

23 May '17

Hello List, I'm having an issue with rtpengine changing the source port of the RTP stream after receiving a 183 from the carrier. Though RTP engine continues to send RTP to the proper carrier IP and port the carrier drops the inbound audio once the source port changes. Can anyone tell me what might trigger this type of behavior from rtpengine and if it's expected? Thank You, ___ John Petrini

3 3

Missing accounting events in ACC table
by Muhammad Shahzad 23 May '17

23 May '17

Hi, About 3 weeks ago i upgraded one of my production server with latest stable kamailio version 4.2.1-fad00a. Now i am getting a lot of complaints about missing CDR events in ACC table. I observe following problems, 1. There are only BYE records in acc table, no record for INVITE or ACK. 2. In kamailio logs when ACK is received against 200 OK response for INVITE, i see following errors, -- ERROR: <core> [parser/parse_from.c:113]: parse_from_uri(): failed to parse >From uri ERROR: pv [pv_core.c:434]: pv_get_xto_attr(): cannot parse From URI NOTICE: <script>: [udp:<null>@1.0.0.127:5060]: Call from 'you(a)kamailio.org' to 'you(a)kamailio.org' has been hanged up by '<null>' at '1419364717.255484' -- Of course all these errors are bogus, I have checked all headers in ACK (not just FROM header), they all seem perfectly fine and valid. 3. Then the dialog times out, -- WARNING: dialog [dlg_handlers.c:1440]: dlg_ontimeout(): timeout for dlg with CallID '6D8BD23CAC65AE3C1DE1D0B531F87B8CFEAA9CB9' and tags '1D3ECD34F5731AB845BA3064AC95BB2D' '7f55e81e0630-100007f-13c4-6009-2440a4-5fa31570-2440a4' -- 4. Any further sequential requests complain about "unable to find dialog", e.g. -- NOTICE: <script>: Sequencial 'BYE' request received from caller ERROR: uac [replace.c:591]: restore_uri(): new URI [] shorter than old URI [ sip:00xxxxxxxxxx@sip.domain.com] WARNING: dialog [dlg_handlers.c:1174]: dlg_onroute(): unable to find dialog for BYE with route param '5ae1.d595' [7845:22877] -- 5. However the acc record for BYE is written to db and log file, -- NOTICE: acc [acc.c:318]: acc_log_request(): ACC: transaction answered: timestamp=1419364760;method=BYE;from_tag=7f55e81e0630-100007f-13c4-6009-2440a4-5fa31570-2440a4;to_tag=1D3ECD34F5731AB845BA3064AC95BB2D;call_id=6D8BD23CAC65AE3C1DE1D0B531F87B8CFEAA9CB9;code=200;reason=OK;src_user=00xxxxxxxxxx;src_domain= sip.domain.com ;src_ip=xx.xx.xx.xx;dst_ouser=+1xxxxxxxxxx;dst_user=1xxxxxxxxxx;dst_domain=yy.yy.yy.yy -- The same config was working fine with older version 4.2.0-97cab8. The kamailio config i am using is pretty much standard, -- #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_DLG 4 ... modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) modparam("acc", "detect_direction", 1) modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) # log to db modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", "DBURL") ... request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # handle requests within SIP dialogs route(WITHINDLG); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { t_relay(); }; exit; }; #### only initial requests (no To tag) #### t_check_trans(); .... # account only INVITEs if (is_method("INVITE")) { setflag(FLT_DLG); # create dialog setflag(FLT_ACC); # do accounting setflag(FLT_ACCFAILED); # ... even if the transaction fails $avp(dlg_timeout) = 60; dlg_manage(); .... } -- Any ideas why its happening? Since it is 3 weeks old so may be problem has already been spotted and fixed by someone else. Otherwise let me know how can i provide more info to help fix this issue. Thank you.

3 11

How to make OutBound calls with kamailio
by sharad tyagi 23 May '17

23 May '17

Hi I am novice to kamailio , I have installed kamailio-4.2.0 and SIREMIS. I have add and registered two IP phones with kamailio these phone able to call each other. How can i make outbound call with that I am using a asterisk server as PSTN Gateway. i am not famalier with kamailio script. Regards Sharad Tyagi SparkTG info Pvt Ltd

4 3

Siremis v4.4.0 Released
by Daniel-Constantin Mierla 23 May '17

23 May '17

Hello, a short note to announce that Siremis v4.4.0 has been released - the web management application for Kamailio. It is supposed to mark the compatibility with Kamailio v4.x series, its future version will target Kamailio v5.0 compatibility. - http://siremis.asipto.com/2017/05/18/siremis-v4-4-0-released/ Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - May 22-24 (USA) - www.asipto.com Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

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sr-users May 2017