- sr-users - kamailio.org

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.

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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

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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

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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

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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

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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

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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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syslog CDR time format
by Yu Boot 18 Aug '17

18 Aug '17

Hello. Can't get 'date-time' format timestamps in syslog CDRs, while SQL CDRs are OK. According to acc module documentation: modparam("acc", "cdr_enable", 1) modparam("acc", "time_format", "%Y-%m-%d %H:%M:%S") modparam("acc", "time_mode", 3) modparam("acc", "cdr_facility", "LOG_DAEMON") syslog: NOTICE: acc [acc_cdr.c:352]: log_write_cdr(): start_time=1503068984.627; end_time=1503068991.250; duration=6.623; src_user=100; src_domain=a.b.c.d; src_ip=e.f.g.h; dst_ouser=200; dst_user=200; dst_domain=k.l.m.n

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On the fly Change
by Logeshwaran G 18 Aug '17

18 Aug '17

Hi, Is there any possibilities of "on the fly" option in kamailio.cfg. For Example: I need to change my carrier IP address in the lcr_gw table, kamailio.cfg should takes it on the fly. Without restarting the kamailio. Thanks & Kind Regards, Logeshwaran G

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Acc cdr duration
by yu＠yu-boot.ru 18 Aug '17

18 Aug '17

Hello. How to remove milliseconds from duration field in CDR? -- Отправлено из myMail для Android

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dispatch_rpc
by Logeshwaran G 18 Aug '17

18 Aug '17

Hi, Is there any other module need to be added for xmlrpc for using dispatch_rpc(). Because I am getting an below error: 0(22485) ERROR: <core> [core/cfg.y:3274]: yyparse(): cfg. parser: failed to find command dispatch_rpc (params 0) 0(22485) CRITICAL: <core> [core/cfg.y:3414]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio_final.cfg, line 640, column 17: unknown command, missing loadmodule? I have loaded xmlrpc as well as xhttp modules. Do I need to enable some other modules as well?? Please Guide me. Thanks & Kind Regards, Logeshwaran G

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SIP_DOMAIN in /etc/kamailio/kamctlrc
by Trent 17 Aug '17

17 Aug '17

Hello. I have not been able to find anything in the wiki, or documentation which covers this. The only item I have found is related to a module. >From the information, I have gleaned on this, it is clear an IP address, or a domain name is to be set here. What is not clear, is if the server is on multiple IP address whether they are on a WAN, LAN, or both, if multiple IP addresses are need, or can be added. I have a situation where I have a public WAN address, and a LAN address. Thanks a bunch!

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kamailio restart:errors
by Uzma Anjum 17 Aug '17

17 Aug '17

Hi,I am new to kamailio. when i restart kamailio,I found errors on log.please help through this. 6(5633) ERROR: mymodule [my_mod.c:373]: outbound_call_process(): [outbound_call_process] URI : sip:11111111112@107.170.69.191:1008 received ip 183.82.115.246 in network ips:107.170.69.191:6090|c1|c2 6(5633) ERROR: mymodule [my_mod.c:379]: outbound_call_process(): [outbound_call_process] Request URI : 11111111112 6(5633) ERROR: mymodule [my_mod.c:395]: outbound_call_process(): outbound_call_process, obcalls privilges :select u.ob_call_privileges, a.phone_number, replace(a.name,'.','-') from account_info a, users u, subscriber s where a.id=u.account_id and u.account_id=s.account_id and u.extension=s.extension and s.username='dyan-dayan' 6(5633) ERROR: mymodule [my_mod.c:425]: outbound_call_process(): outbound_call_process query from user:dyan-dayan, phone_type:1 acc_name:dayan 6(5633) ERROR: mymodule [my_mod.c:501]: outbound_call_process(): [outbound_call_process]NEW URI : sip:ivr-1-dayan@107.170.69.191:6090 6(5633) ERROR: <core> [udp_server.c:448]: udp_rcv_loop(): recvfrom:Received encrypted packet len:4, v▒..

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ClueCon Slides: Industrial Grade FreeSWITCH, Scaling, Balancing, High Availability, SIP, WebRTC
by Giovanni Maruzzelli 17 Aug '17

17 Aug '17

*Industrial Grade FreeSWITCH Scaling, Balancing and High Availability for SIP and WebRTC* Scaling your FreeSWITCH platform to serve a growing user base is a critical challenge. We'll go through the best techniques, practices, and implementations for Voice and Video Calls, Conferencing, WebRTC, SIP, Chatting, Presence and Instant Messaging Slides can be downloaded from http://opentelecom.it/cluecon2017.pdf Enjoy ClueCon! -giovanni -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18

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Re: [SR-Users] DMQ Not Discovering all Nodes
by Charles Chance 16 Aug '17

16 Aug '17

Hello, Can you share the rest of your config (at least, any dmq modparams and the main route block)? Cheers, Charles On 16 Aug 2017 22:48, "John Petrini" <jpetrini(a)coredial.com> wrote: Hello, I've just started doing some testing with dmq but I'm having trouble getting it to discover all of the nodes in my cluster. There are 7 kamailio instances with the following in their configs: Example from the notification server (10.0.10.211): listen=udp:10.0.10.211:5080 loadmodule "dmq.so" # ----- dmq params ----- modparam("dmq", "server_address", "sip:10.0.10.211:5080") modparam("dmq", "notification_address", "sip:10.0.10.211:5080") Example from one of the nodes (10.0.10.216). (Only the listen and server_address is changed to match the local ip of each node.) listen=udp:10.0.10.216:5080 loadmodule "dmq.so" # ----- dmq params ----- modparam("dmq", "server_address", "sip:10.0.10.216:5080"); modparam("dmq", "notification_address", "sip:10.0.10.211:5080") Output from kamcmd dmq.list_nodes { host: 10.0.10.211 port: 5080 resolved_ip: 10.0.10.211 status: 2 last_notification: 0 local: 0 } { host: 10.0.10.216 port: 5080 resolved_ip: 10.0.10.216 status: 2 last_notification: 0 local: 1 } I'm expecting to see all 7 nodes in the output but only see the local node and the notification node. I imagine I could use an SRV record for the notication address to add all of the nodes but I thought the notification node was supposed to share the rest of the nodes in the cluster. Can anyone explain what it is I'm doing wrong? Thank You, John Petrini _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Sipcentric Ltd. Company registered in England & Wales no. 7365592. Registered office: Faraday Wharf, Innovation Birmingham Campus, Holt Street, Birmingham Science Park, Birmingham B7 4BB.

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DMQ Not Discovering all Nodes
by John Petrini 16 Aug '17

16 Aug '17

Hello, I've just started doing some testing with dmq but I'm having trouble getting it to discover all of the nodes in my cluster. There are 7 kamailio instances with the following in their configs: Example from the notification server (10.0.10.211): listen=udp:10.0.10.211:5080 loadmodule "dmq.so" # ----- dmq params ----- modparam("dmq", "server_address", "sip:10.0.10.211:5080") modparam("dmq", "notification_address", "sip:10.0.10.211:5080") Example from one of the nodes (10.0.10.216). (Only the listen and server_address is changed to match the local ip of each node.) listen=udp:10.0.10.216:5080 loadmodule "dmq.so" # ----- dmq params ----- modparam("dmq", "server_address", "sip:10.0.10.216:5080"); modparam("dmq", "notification_address", "sip:10.0.10.211:5080") Output from kamcmd dmq.list_nodes { host: 10.0.10.211 port: 5080 resolved_ip: 10.0.10.211 status: 2 last_notification: 0 local: 0 } { host: 10.0.10.216 port: 5080 resolved_ip: 10.0.10.216 status: 2 last_notification: 0 local: 1 } I'm expecting to see all 7 nodes in the output but only see the local node and the notification node. I imagine I could use an SRV record for the notication address to add all of the nodes but I thought the notification node was supposed to share the rest of the nodes in the cluster. Can anyone explain what it is I'm doing wrong? Thank You, John Petrini

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Kamailio not forces TLS socket
by Yuriy Gorlichenko 16 Aug '17

16 Aug '17

Hi! I using uac for send MESSAGE request from kamailio to remote destination endpoint Before sending i setting up $fs seudovariable as $fs = "tls:12.3.4.5:443"; kamailio listeinig this interface and succesfully receives requests and sending responses on it When i call uac_req_send();There is error in log file: ERROR: tm [ut.h:327]: uri2dst2(): no corresponding socket found for "2.3.26.11" af 2 (udp:2.3.26.11:57050) ERROR: tm [uac.c:272]: t_uac_prepare(): t_uac: no socket found If I setting up udp interface for listening then kamailio ognores my $fs set up and sending message through udp socket. kamailio version 4.4.6 Now i trying to roll back to 4.4.4 version but not sure if it will be helpfull.

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Permission denied for /var/run/kamailio
by Ryan Wagoner 16 Aug '17

16 Aug '17

I'm running Kamailio 4.4.6-2.1 installed from RPM on CentOS 7, where /var/run is a tmpfs (by default). After every reboot Kamailio fails to start with the following error: Aug 16 00:27:39 sbc1 /usr/sbin/kamailio[7135]: ERROR: mi_fifo [fifo_fnc.c:72]: mi_init_fifo_server(): Can't create FIFO: Permission denied (mode=432) I have the following line in my kamailio.cfg modparam("mi_fifo", "fifo_name", "/var/run/kamailio/kamailio_fifo") At this point the /var/run/kamailio directory looks like this drwx------. 2 root root 80 Aug 16 00:30 . drwxr-xr-x. 26 root root 880 Aug 16 00:30 .. srw-------. 1 kamailio kamailio 0 Aug 16 00:30 kamailio_ctl If I run "chown kamailio /var/run/kamailio" and "systemctl start kamailio" I am good to go. Am I missing something in my config? It appears Kamailio is creating the /var/run/kamailio folder as root, writing the kamailio_ctl file, dropping root privileges, and then trying to write the kamailio_fifo file. Thanks, Ryan

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sipcapture/HEP msg sizes>1500
by Daniel Tryba 16 Aug '17

16 Aug '17

While looking into HOMER I saw that large INVITEs got truncated in the webinterface, looking at the HEP traffic these (UDP) messages got fragmented in transport (asterisk/res_hep) and I started to think these were not getting reassembled correctly. Only to find out the database schemas have set the msg size set to varchar(1500). Is there any good reason for not changing this to a bigger default?

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RTP loops
by Mauro Costantini 16 Aug '17

16 Aug '17

Hi everyone, we have an installation with: - 1 kamailio instance - with 2 interface, one with public and one with private IP (for local communication to the db and asterisk ). We use keepalived service for failover and so, every interface has 2 IPs: the real one and the “virtual” one for the keepalived service - some asterisk instances for transcoding and billing - with one interface with private IP - 2 RTP (rtpengine) instances - with 2 interfaces, one with public and one with private IP A few times happens a loop on the RTPengine and we could see a lot of lines like this in the log: “Too many packets in UDP receive queue (more than 50), aborting loop. Dropped packets possible” This loop doesn't stop until all the resources are exhausted and it results in the freeze of the machine. We don’t understand the reason why these loops are generated. We have 2 cases: 1) we send an INVITE to the CPE (a Bria on iOS) The CPE answers with a "100 Trying", a "180 Ringing" and a few seconds later sends us 3 packets of "200 ok" with SDP like this: SIP/2.0 200 OK Via: SIP/2.0/UDP 178.250.x.y;branch=z9hG4bKb842.29952898c30d6d06b75b916618357ccd.2 Via: SIP/2.0/UDP 192.168.x.y:5090;rport=5090;branch=z9hG4bK7fd100fb Record-Route: <sip:178.250.x.y;r2=on;lr=on;ftag=as26b82ac7;did=cb5.0ce;nat=yes> Record-Route: <sip:192.168.x.z;r2=on;lr=on;ftag=as26b82ac7;did=cb5.0ce;nat=yes> Contact: <sip:6051018****@158.148.x.y:3180;rinstance=ffdff5e0142dfd2d> To: <sip:6051018****@192.168.x.z>;tag=29859d3c From: <sip:051438****@192.168.x.y>;tag=as26b82ac7 Call-ID: 698200851849894c5874186163abe668@192.168.x.y:5090 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, BYE, REFER, INFO, NOTIFY, OPTIONS, UPDATE, PRACK, MESSAGE, SUBSCRIBE Content-Type: application/sdp Supported: replaces User-Agent: Bria iOS release 3.9.7 stamp 38887.38893 Content-Length: 230 v=0 o=- 1085831961 3 IN IP4 100.81.231.195 s=Cpc session c=IN IP4 100.81.231.195 t=0 0 m=audio 65454 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv Could the IP of the media part (100.81.231.195 in the CGNAT 100.64.0.0/10) be the cause of the loop? If that’s the case, how can we prevent the loop? 2) in the second case the CPE is an Avaya The Avaya send us an INVITE. (Also in this case in the SDP there is a private IP) INVITE sip:04818****@178.250.x.y SIP/2.0 From: <sip:0432415453@185.55.x.y>;tag=4500808-c0a8077d-13c4-50022-313081f-200d750c-313081f To: <sip:04818****@178.250.x.y> Call-ID: 44e8f20-c0a8077d-13c4-50022-313081f-31e99586-313081f CSeq: 1 INVITE Via: SIP/2.0/UDP 185.55.x.y:5060;rport;branch=z9hG4bK-313081f-257bc48-1472e4a5 Privacy: none Max-Forwards: 70 User-Agent: OfficeServ 7200 Contact: <sip:0432415453@185.55.z.y:5060;transport=udp> Allow: REGISTER,INVITE,ACK,BYE,REFER,NOTIFY,CANCEL,INFO,OPTIONS,PRACK,SUBSCRIBE,UPDATE Supported: 100rel Content-Type: application/sdp Content-Length: 289 v=0 o=SAMSUNG_SIP_GATEWAY 39304265 0 IN IP4 185.55.x.y s=SIP_CALL c=IN IP4 192.168.7.126 t=0 0 m=audio 30012 RTP/AVP 18 8 0 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv We reply with a “183 Session Progress” and the Avaya send US an “INFO” packet (dtmf) we replay with a “200 OK”; and then, after another “INFO” packet we replay with a “200 OK” with SDP. After that the Avaya replies with an ACK 2017/08/02 10:17:32.101121 185.55.x.y:5060 -> 178.250.x.y:5060 ACK sip:04818****@178.250.x.y SIP/2.0 From: <sip:0432415453@185.55.x.y>;tag=4500808-c0a8077d-13c4-50022-313081f-200d750c-313081f To: <sip:04818****@178.250.x.y>;tag=as783106e4 Call-ID: 44e8f20-c0a8077d-13c4-50022-313081f-31e99586-313081f CSeq: 1 ACK Via: SIP/2.0/UDP 185.55.x.y:5060;rport;branch=z9hG4bK-3130831-257ffea-84906e4 Max-Forwards: 70 User-Agent: OfficeServ 7200 Route: <sip:178.250.x.y;lr=on;r2=on;ftag=4500808-c0a8077d-13c4-50022-313081f-200d750c-313081f;did=adc.2f61> Route: <sip:192.168.x.y;lr=on;r2=on;ftag=4500808-c0a8077d-13c4-50022-313081f-200d750c-313081f;did=adc.2f61> Contact: <sip:043241****@185.55.x.y:5060;transport=udp> Content-Length: 0 And after that the chaos… Has anyone had a problem like this? Can somebody help us in order to prevent this loop? In the attached images you can see: [0] a call, our machines IPs, an ACK from public interface of our SBC to its private one; and than a lot of ACKs from the keepalived IP to the private IP of our SBC. [1] one of these strange ACKs, they are all different in content, but similar in structure: multiple Record-Route / Via, truncating too long message; [2] a "never ending" BYEs. After modifying the configuration adding some "listen=IP_ADDR:PORT" (and of course excluding the keepalive IP) we had no loops but is this enough? Why Kamailio generated the loops? Is there something else we should review in our config? Any advice would help a lot. Thanks. [0] https://goo.gl/rWh5db [1] https://goo.gl/MNShq9 [2] https://goo.gl/6mxepG

1 0

UAC reg_random_delay issue
by Ivo Vastert 16 Aug '17

16 Aug '17

Hi, I'm using the UAC module in Kamailio to setup about 500 outbound registration towards a single host. The issue I'm currently experiencing is that when we start Kamailio all this 500 registrations are initiated at once towards the host which can't handle all the requests at the same time. Therefor I tried to spread the registrations using the reg_random_delay function: modparam("uac", "reg_random_delay", 60) I would expect that this would spread the registrations over about a minute but in reality the registrations are spread as follow: Starttime / Registrations 21:59:14 151 21:59:15 166 21:59:16 127 21:59:18 60 21:59:22 35 So Kamailio is started at 21:59:14 and initiates 151 registrations, 1 second after start another 166 2 seconds after start another 127 4 seconds after start another 60 8 seconds after start the last 35 In reality it looks like some increment is applied instead of calculating a random starting offset for each outbound registration? Is there any solutions to this issue available? Best regards, Ivo Vastert

2 6

rtpengine : question about RTP/RTCP multiplexing (RFC 5761) and demultiplexing
by Abdoul Osséni 16 Aug '17

16 Aug '17

Hello All, I read rtpengine can multiplex and demultiplex RTP/RTCP. Cf.https://github.com/sipwise/rtpengine Example: UAC -> Kamailio + rtpengine -> Asterisk UAC and Asterisk don't support RFC 5761. Only rtpengine does. Does this mean that uac must be compliance with RFC 5761 if I want multiplex and demultiplex RTP/RTCP between UAC and rtpengine? Thanks. Abdoul.

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