sr-users September 2018

sr-users@lists.kamailio.org

72 participants
124 discussions

remove_hf("Bla") for all requests
by Sergiu Pojoga 12 Sep '18

12 Sep '18

Hi there, Say I need to remove_hf("User-Agent") for all requests, back and forth. So I add it at the top of *request_route *section. However, replies don't seem to be affected by it. Do I really need to arm a *t_on_reply route* for this simple purpose? "Problem" with that is that *route[RELAY]* already has some *onreply_route* block doing things like *NATMANAGE *for some methods and as far as I know - only one* onreply_route* can be armed for a request? route[RELAY] { ... if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } t_relay(); } onreply_route[MANAGE_REPLY] { ... route(NATMANAGE); } Any suggestions? Thanks.

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Re: [SR-Users] kamailio + rtpengine + custom filename
by Жан Базаров 12 Sep '18

12 Sep '18

Message-ID: <CABVi_EzZRF8nUqoBGcUtUeAV2pknn5S0urcmJFgHSSz6N1kJ5g(a)mail.gmail.com> Content-Type: text/plain; charset="UTF-8" Hello, You could use metadata to save your favourite filename on it. After recording is finished, replace it. The metadata file is created based on <CALL_ID> for each call. With Regards. Mojtaba On Mon, Sep 10, 2018 at 8:53 PM Жан Базаров <chiefkeeft(a)gmail.com> wrote: > > Hello! Filenames have the format <call_ID>-<random>-<SSRC>.<extension> > How i can change filename in start_recording() ??? Thx you. Can you help me with it task? Show me, please, an example. I want filename of the following format, for example: 'output-dir/yyyy/mm/dd/fromnumber/callid-tonumber-uniqueid.wav'.

2 1

Missing ACC record in case of canceled call forking
by Marco Capetta 12 Sep '18

12 Sep '18

Hi All, I'm facing a strange problem of missing ACC record in case of parallel call forking. The scenario is the following: - A subscriber with 1 device registered - B subscriber with 2 device registered (B1 and B2) CASE 1: - A calls B - B1 and B2 start ringing - A hangups the call before B1 or B2 answers Kamailio generates an ACC record. CASE 2: - A calls B - B1 and B2 start ringing - B1 rejects the call sending back a 486 - B2 is still ringing - A hangups the call before B2 answers Kamailio DOESN'T generate an ACC record. We have Kamailio v5.1.4 with TM module enabled. ACC configuration is the following: modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 1) modparam("acc", "detect_direction", 1) modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) modparam("acc", "failed_transaction_flag", 3) I increased debug level of TM and ACC modules and I added some debug lines as well. The difference between the two calls is that, after the CANCEL sent by A is processed by Kamailio, in CASE 1 I have the following lines: Aug 28 14:21:11 spce proxy[13188]: DEBUG: tm [t_hooks.c:258]: run_trans_callbacks_internal(): DBG: trans=0x7f5a88049198, callback type 512, id 0 entered Aug 28 14:21:11 spce proxy[13188]: DEBUG: acc [acc_logic.c:670]: tmcb_func(): acc callback called for t(0x7f5a88049198) event type 512, reply code 487 Aug 28 14:21:11 spce proxy[13188]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 8, index i 0 Aug 28 14:21:11 spce proxy[13188]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 8, index i 1 Aug 28 14:21:11 spce proxy[13188]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 8, index i 2 Aug 28 14:21:11 spce proxy[13188]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 8, index i 3 Aug 28 14:21:11 spce proxy[13188]: DEBUG: acc [acc_extra.h:63]: free_strar_mem(): Freeing memory, type is 2, message_index 8, index i 4 Those debug lines are not printed for CASE 2. Is there any configuration I'm missing or is it a bug? Thank you very much for your support. Marco

2 14

Call termination via dialog RPC dlg.end_dlg for the SIP-UA behind NAT
by Denys Pozniak 11 Sep '18

11 Sep '18

Hello! When calling RPC command dlg.end_dlg, BYE for leg behind NAT sends according to the Contact header not to the real ip:port. RPC ul.dump shows correct address in Received: sip:2.2.2.2:7488, but looks dialog does not know real address. How can I force sending to the real address? [root@proxy1]# kamcmd dlg.list { h_entry: 3753 h_id: 6741 call-id: 0_618427980(a)10.10.10.10 from_uri: sip:1234567890@1.1.1.1:5060 to_uri: sip:3333333333@1.1.1.1:5060 state: 2 start_ts: 0 init_ts: 1536671492 timeout: 0 lifetime: 7200 dflags: 512 sflags: 4096 iflags: 32 caller: { tag: 2750096738 contact: sip:1234567890@10.10.10.10:5060 cseq: 2 route_set: socket: udp:1.1.1.1:5060 } callee: { tag: contact: cseq: route_set: socket: } profiles: { } variables: { } } [root@proxy1]# kamcmd ul.dump Info: { AoR: 1234567890 HashID: 262653578 Contacts: { Contact: { Address: sip:1234567890@10.10.10.10:5060 Expires: 1269 Q: -1.000000 Call-ID: 0_213504837(a)10.10.10.10 CSeq: 8 User-Agent: Yealink SIP-T21P Received: sip: 2.2.2.2:7488 Path: [not set] State: CS_NEW Flags: 0 CFlags: 65600 Socket: udp: 1.1.1.1:5060 Methods: 16383 Ruid: uloc-5b97c358-6052-1 Instance: [not set] Reg-Id: 0 Server-Id: 0 Tcpconn-Id: -1 Keepalive: 1 Last-Keepalive: 1536674159 Last-Modified: 1536674159 } } } -- BR, Denys Pozniak

1 0

Remove branch from branch list
by Yuriy Gorlichenko 11 Sep '18

11 Sep '18

Hi I'm trying to manipulate with branches at the $branch variable. 8 actually need to remove unneeded branch if not passed some checks. For now I Just rewriting uri and dst_uri with some face address but it looks little bit dirty way. Is here any an other more clear way to do this?

2 2

Record Routes
by Wilkins, Steve 11 Sep '18

11 Sep '18

Hello All, Can a Route: be removed?, if so, how? Thank you?

2 4

Logging of messages generated by Kamailio itself
by Konstantin Polyakov 11 Sep '18

11 Sep '18

Hello, Could somebody help with one issue? In some cases Kamailio can send messages itself. For example, when fr_inv_timer is expired Kamailio sends 408 to the INVITE sender and CANCEL to the INVITE receiver. These two messages (408 and CANCEL) are not printed into the log and I cannot find a way to do that. There are also other scenarios when some Kamailio modules send SIP messages by its own. So is there way to print out into log all such messages? My goal is to get full message flow in the log per certain call-id. Thank you in advance. Best regards. Konstantin

2 9

kamailio + rtpengine + custom filename
by Жан Базаров 11 Sep '18

11 Sep '18

Hello! Filenames have the format <call_ID>-<random>-<SSRC>.<extension> How i can change filename in start_recording() ???

2 1

tpops module
by volga629＠networklab.ca 11 Sep '18

11 Sep '18

Hello Everyone, I am using tcpops module for keepalive and on BYE when tcp keeplaive close is called produce this error in log. Sep 8 11:21:26 canlvprx01 /usr/sbin/kamailio[18933]: ERROR: <core> [core/sr_module.c:1781]: get_int_fparam(): Could not convert PV to int Sep 8 11:21:26 canlvprx01 /usr/sbin/kamailio[18933]: ERROR: tcpops [tcpops_mod.c:293]: w_tcp_keepalive_disable1(): invalid parameter 'con' (must be a number) kamailio-5.1.3-1.gitf0dce0c99.fc27.x86_64 if (is_method("BYE|CANCEL")) { $avp(bye_conid) = $conid; rtpengine_delete(); t_on_reply("4"); } onreply_route[4] { if(is_method("BYE") && t_check_status("200")) { tcp_keepalive_disable("$avp(bye_conid)"); } } volga629

3 3

Simultaneous usage of t_on_reply() and t_on_failure()
by Володимир Іванець 10 Sep '18

10 Sep '18

Hello everyone, I wanted to ask those who know if Kamailio's behavior I'm facing is expected or I should make some improvements to the configuration. Kamailio version is 5.1.0. I have a route where RTPEngine parameters are being collected and *rtpengine_offer()* is called. After that *t_on_reply("REPLY_SIP_TO_SIP");* followed by the *t_on_failure("FAILURE_SIP_TO_SIP");* are used. The idea is to process all responces except 415 or 488 from UAC as usual in *onreply_route[REPLY_SIP_TO_SIP]* and use *failure_route[FAILURE_SIP_TO_SIP]* to update SDP with *rtpengine_offer()* if necessary. *onreply_route[REPLY_SIP_TO_SIP]* just goes to *exit;* if *$rs* equals 415 or 488. This works fine with Htek phone which sends 100, 180 and then 488. But I can not see *failure_route[FAILURE_SIP_TO_SIP]* execution for calls to Zoiper which replies with 100 and immediately 415. t_on_failure(failure_route) documentation says: "Sets failure routing block, to which control is passed after a transaction completed with a negative result but before sending a final reply." and to be honest I don't really understand how lacking of responce prevents failure_route from executing. Thanks a lot!

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sr-users September 2018