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37519 discussions

sercmd dispatcher.list command giving an error: 500 - No Destination Sets
by Deepak Jaiswal 22 Aug '17

22 Aug '17

Hi, I am new to Kamailio and trying to debug an issue. We have two servers. On one server, I am getting destination like below. sercmd dispatcher.list { SET_NO: 1 SET: { SET_ID: 1 DEST: { URI: sip:xxx.xxx.xxx.139:5060 FLAGS: AX PRIORITY: 0 ATTRS: reg-time=1503397149 } DEST: { URI: sip:xxx.xxx.xxx.:5060 FLAGS: AX PRIORITY: 0 ATTRS: reg-time=1503397150 } } } However, on another server, I am getting below error. sercmd dispatcher.list error: 500 - No Destination Sets sercmd -s unixs:/tmp/kamailio_ctl dispatcher.list error: 500 - No Destination Sets [root@uk-vap-002 ~]# sercmd system.listMethods cfg.add_group_inst cfg.commit cfg.del cfg.del_delayed cfg.del_group_inst cfg.diff cfg.get cfg.help cfg.list cfg.rollback cfg.set cfg.set_delayed cfg.set_delayed_int cfg.set_delayed_string cfg.set_now_int cfg.set_now_string cfg.seti cfg.sets core.arg core.echo core.flags core.info core.kill core.printi core.prints core.ps core.psx core.pwd core.sctp_info core.sctp_options core.shmmem core.tcp_info core.tcp_options core.udp4_raw_info core.uptime core.version ctl.connections ctl.listen ctl.who dispatcher.list List going on.................................... I tried to find dispatcher.list file on both server. However, it is not there on both the server. Could you please help me on this? Thanks and Regards Deepak K. Jaiswal Tech support Engineer My Phone : +91- 95351-23352, +91- 88675-01385 My Movius work line: +1 (470) 298-7210 [ovius]<http://www.moviuscorp.com/> www.moviuscorp.com<http://www.moviuscorp.com/>

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append_branch and tls
by James Cloos 21 Aug '17

21 Aug '17

[I see I posted this to -dev instead of -users; fixing that -JimC] When using rewrite_ruri() and append_branch() to set a list of next hops to try, what is the proper syntax to specify that a given route should use tcp or tls/tcp? I've mostly used a single next hop via t_relay_to_tcp() or t_relay_to_tls(), but need route stacks now. -JimC -- James Cloos <cloos(a)jhcloos.com> OpenPGP: 0x997A9F17ED7DAEA6

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Kamailio+SEMs conference call setting
by leslie mekiyess 21 Aug '17

21 Aug '17

Hi, I have a Kamailio server , a SEM one, an SQL server and Siremis installed and working. I am able to perform p2p calls with a softmobile. But I didn't succeed to perform conference / group / speed dial calls. Besides defining some records in the database, I don't know how to complete the necessary (Kamailio/SEM ...) configuration in order to perform conference call, group and speed dial . Any assitance would very appreciated. Leslie :-)

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siptrace: how to duplicate the complete dialog?
by Daniel Tryba 21 Aug '17

21 Aug '17

I'm trying to trace a complete dialog on a loadbalancer/proxy: UAC <-> Proxy <-> backend 109.235.34.a <-> 109.235.32.b <-> 185.61.68.c Proxy runs 4.3.x (but with 4.4.x I appear to have the same results). siptrace config: ============================================= modparam("siptrace", "duplicate_uri", "sip:10.11.12.13:9060") modparam("siptrace", "hep_mode_on", 1) modparam("siptrace", "trace_to_database", 0) modparam("siptrace", "trace_flag", 22) modparam("siptrace", "trace_on", 1) ============================================= After REQINIT (which does pike and handles OPTIONS), I setflag(22) the message. ============================================= route(REQINIT); setflag(22); route(NATDETECT); route(WITHINDLG); ============================================= Next depending in where the message comes from: ============================================= if(!is_in_subnet($si, backends)) { ds_select_dst(1, 0); t_on_failure("RTF_DISPATCH"); } else { remove_hf("Route"); loose_route(); } route(RELAY); ============================================= sip_trace() is being called in the routes: WITHINDLG MANAGE_BRANCH MANAGE_REPLY MANAGE_FAILURE RTF_DISPATCH The resulting duplicate packets are: 11:20:21.037 INVITE 109.235.34.a:39621 109.235.32.b:5060 tcp 11:20:21.040 INVITE 109.235.34.a:39621 109.235.32.b:5060 tcp 11:20:21.047 401 185.61.68.c:5060 109.235.32.b:5060 udp 11:20:21.073 ACK 109.235.34.a:39621 109.235.32.b:5060 tcp 11:20:21.117 INVITE 109.235.34.a:39621 109.235.32.b:5060 tcp 11:20:21.135 100 185.61.68.c:5060 109.235.32.b:5060 udp 11:20:21.450 180 185.61.68.c:5060 109.235.32.b:5060 udp 11:20:27.929 200 185.61.68.c:5060 109.235.32.b:5060 udp 11:20:28.106 ACK 109.235.34.a:39621 109.235.32.b:5060 tcp 11:20:30.625 BYE 185.61.68.c:5060 109.235.32.b:5060 udp 11:20:30.659 BYE 185.61.68.c:5060 109.235.32.b:5060 udp 11:20:30.663 BYE 109.235.32.b:5060 109.235.34.a:39621 tcp 11:20:30.664 200 109.235.34.a 39621 109.235.32.b:5060 tcp 11:20:30.664 200 109.235.32.b:5060 185.61.68.c:5060 udp What is missing are the forwarded replies, there is an ACK that matches the 401, but I have no clue what message was send to the UAC. Adding a sip_trace after REQINIT shows nothing extra. How (if possible) do I get all messages being send via the proxy? I could use sngrep/sipgrep, but that is an other process to monitor for failure.

2 4

allow inbound calls from any ip
by Logeshwaran G 21 Aug '17

21 Aug '17

Hi, I need to allow the inbound calls from any ip i kamailio. Input please Thanks & Kind Regards, Logeshwaran G

2 1

Take call counts from kamailio
by Logeshwaran G 21 Aug '17

21 Aug '17

Hi, How to take call counts from the kamailio? Thanks & Kind Regards, Logeshwaran G

2 12

sdp extra \r\n in multipart body
by Diego Nadares 21 Aug '17

21 Aug '17

Hi guys, I have the following multipart body Content-Type: multipart/mixed;boundary=uniqueBoundary Remote-Party-ID: <tel:1112222;phone-context=+54> Content-Length: 364 --uniqueBoundary Content-Type: application/sdp v=0 o=user1 53655765 2353687637 IN IP4 111.111.111.111 s=- c=IN IP4 111.111.111.111 t=0 0 m=audio 6000 RTP/AVP 8 *a=rtpmap:8 PCMA/8000* *--uniqueBoundary--* Content-Type: application/gtd Content-Disposition: signal;handling=optional IAM, CPC,20 GCI,885sdfasdf1123 --uniqueBoundary-- As you can see *--uniqueBoundary *before *a=rtpmap:8 PCMA/8000* has not an extra '\r\n'. Is this correct? Or should have it? The thing is that rtpengine is receiving sdp with no final '\r'n' from kamailio: Dump for 'offer' from 127.0.0.1:37388: { "sdp": "v=0#015#012o=user1 53655765 2353687637 IN IP4 111.111.111.111#015#012s=-#015#012c=IN IP4 111.111.111.111#015#012t=0 0#015#012m=audio 6000 RTP/AVP 8#015#012*a=rtpmap:8 PCMA/8000"*, "ICE": "remove", "direction": [ "pub", "priv" ], "replace": [ "origin", "session-connection" ], "transport-protocol": "RTP/AVP", "call-id": "1-17755(a)111.111.111.111", "received-from": [ "IP4", "111.111.111.111" ], "from-tag": "1", "command": "offer" } And the resulting sdp from rtpengine is this: --uniqueBoundary Content-Type: application/sdp v=0 o=user1 53655765 2353687637 IN IP4 172.16.213.16 s=- c=IN IP4 172.16.213.16 t=0 0 m=audio 35066 RTP/AVP 8 *a=rtpmap:8 PCMA/8000a=sendrecv* a=rtcp:35067 The problem is related with that '\r\n' but Is this a malformed message or a possible bug? Thanks in advance! Diego

2 1

Documenation of functions exported for KEMI
by Sebastian Damm 19 Aug '17

19 Aug '17

Hi, I was wondering whether there is some kind of documentation of all functions currently exported for KEMI by Kamailio modules. I know I can find them somehow via grep, but I could not find a page in the Kamailio wiki listing all of them. Is there any? Best Regards, Sebastian

2 1

Converting to kemi
by James Cloos 19 Aug '17

19 Aug '17

[I see I posted this to -dev instead of -users; fixing that -JimC] I'm working on converting some configs to kemi lua and have a couple of questions: First: Does KSR.pv.get() support multiple pseudos in a single call? The examples such as misc/examples/kemi/kamailio-basic-kemi-lua.lua make several calls, but also do things like KSR.pv.get("$T_branch_idx]") to avoid having the append "]" to the string. That makes me wonder why that file has code like this: KSR.dbg("new branch [".. KSR.pv.get("$T_branch_idx]") .. " to ".. KSR.pv.get("$ru") .. "\n"); instead of like: KSR.pv.get("[$T_branch_idx] to $ru\n") Second: Is it still the case that, eg, event_route[xhttp:request] cannot be in the kemi lua code? There was another. Perhaps I'll recall what it was... :( -JimC -- James Cloos <cloos(a)jhcloos.com> OpenPGP: 0x997A9F17ED7DAEA6

2 1

Kamailio_428 memory usage
by Evgeniy G. 18 Aug '17

18 Aug '17

Good afternoon! Can you please tell us how the Kamailio works with shared memory. The fact is that there was an accident in which Kamailio reported an error - little common memory. This happened a day later as I added the dialplan module. After that, I began to monitor the values: used_size real_used_size max_used_size free And I saw that I can not properly assess. I note that the server does not process calls, it is idle. After the server is started, the indicators are as follows: used_size: 63 real_used_size: 146 max_used_size: 146 free 106 After rebooting the module (for example, drouting, in which the data has not changed): used_size: 63 real_used_size: 193 max_used_size: 231 free 62 I found, a description, the reasons for the increase in memory after rebooting the modules, I quote "As suggested by Daniel, for carrierroute reload, we need the same amount of memory as it used to be. Block when reloading the table. " But why: 1. after increasing the memory the server frees only real_used_size 2. That that is specified as real_used_size on my observations can not be released in current of several days (probably weeks). 3.If in this state again to reboot the module (the same one that led to an increase in memory), the memory does not increase. Question: during what time should kamailio release the memory specified as real_used_size? Also what is considered as real_used_size? -- View this message in context: http://sip-router.1086192.n5.nabble.com/Kamailio-428-memory-usage-tp161327.… Sent from the Users mailing list archive at Nabble.com.

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