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37519 discussions

mod.stats rpc call crashes
by Andy 26 May '17

26 May '17

Hi, I see kamailio crashing on a call to possibly (mod.stats rpc call), @function __strcmp_sse42 probably is referencing null, which is passed mname and pointer to mname. backtrace: thanks a lot. -- View this message in context: http://sip-router.1086192.n5.nabble.com/mod-stats-rpc-call-crashes-tp158967… Sent from the Users mailing list archive at Nabble.com.

2 4

question about cseq handling
by 张顺通 26 May '17

26 May '17

In urecord.c:get_ucontact_by_instance, REGISTER without instance-id will check cseq number(must be higher), REGISTER with instance-id will not check cseq number. what is the difference?why not check cseq in REGISTER with instance-id? Thanks

2 1

Some Errors
by Douglas Girard 25 May '17

25 May '17

Hello, I am using a fresh install of centos 7, homer (using install script) and kamailio (installed with homer script). After modifying kamailio.cfg file with correct DB username, password and Server IP, i am able to start and stop kamailio service. Now my issue is that I have begun to duplicate SIP messages from my PBX, but am receiving these and other errors from kamailio: May 25 12:44:20 localhost.localdomain homer[14456]: ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on query: Data too long for column 'msg' at row 1 (1406) May 25 12:44:20 localhost.localdomain homer[14456]: ERROR: <core> [db_query.c:235]: db_do_insert_cmd(): error while submitting query and: May 25 12:44:42 localhost.localdomain homer[14456]: ERROR: <core> [receive.c:173]: receive_msg(): core parsing of SIP message failed ( 10.3.3.24:51553/1) May 25 12:45:12 localhost.localdomain homer[14460]: ERROR: <core> [parser/parse_fline.c:257]: parse_first_line(): parse_first_line: bad message (offset: 0) Any suggestions? Thanks!

1 0

Dialog module timeout behaviours
by Alex Balashov 25 May '17

25 May '17

Hello, I'm running 4.3.5:1b0c0a and have run into two distinct, bizarre issues with the dialog module that I am unable to explain. In all cases, I am using $dlg_ctx(send_bye) = 1, and passing in a dialog timeout of 7200 sec via AVP (timeout_avp modparam). 1) Local BYEs are sent after either ~60 or ~120 sec -- This happens on a very small number of answered calls for no apparent reason. I have spent days poring through the traces trying to figure out if there's anything eccentric about how the messages, but can't find anything out of the ordinary except (a) 1 retransmission of 200 OK to INVITE due to not receiving ACK fast enough due to end-to-end latency, and (b) consequently, retransmission of ACK. My understanding is that the dialog module does not prophylactically end calls as a response to strange or improper messaging, but only (a) when the dialog timeout is reached or (b) keepalive OPTION pings (if enabled) fail. Keepalive OPTIONS pings are not enabled in these scenarios, and are not sent in the traces. (2) Local BYEs sent in response to race conditions that result in strange dialog state transitions -- There are some calls that look something like: A. Caller INVITEs callee; B. Callee sends progress; C. Callee sends 200 OK answer and immediately sends BYE (e.g. 1 ms later); D. Kamailio receives these nearly simultaneously and forwards the BYE first; E. Caller sends 487 Request Terminated response (processing BYE as if it were CANCEL); F. Kamailio forwards 200 OK back to caller, who ignores it. G. Callee retransmits 200 OK through Kamailio. A few minutes later, Kamailio sends BYE in both directions even though dialog should have been removed from state tracking long ago because of the dispositive 487 reply. ... In particular, I'm curious as to whether there have been any fixes to deal with scenarios like #2 since 4.3.5. I wasn't able to spot any obvious ones in the commit history for the module. I'm also at a complete loss to explain #1. I have traces for all of these scenarios, but would prefer to send privately. Many thanks! -- 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/

2 4

in-dialog contact update before dialog being confirmed
by Jayesh Nambiar 25 May '17

25 May '17

Hello, In a user-user based calling on mobile apps, when the call is connected and the user moves from a Wifi to Mobile data (network change); the device sends a Re-Invite with new contact information and subsequent messages go to appropriate changed contact address. But when the dialog is in say ringing state; and the caller moves from one network to another; what would be an appropriate SIP method to trigger the contact update. My problem is generating a Re-INVITE or an UPDATE before the call is connected is very difficult. What I tried is sending an in-dialog INFO message with new contact address; which gets relayed to the called party and the response for INFO is also properly received on the new contact address. But the subsequent response of the INVITE like a 486 or 200 goes to the original contact only. What would be the best way to update the contact address of a client if I'm maintaining all the dialogs in kamailio. Thanks, - Jayesh

3 4

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.

3 5

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?

2 1

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.

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