- sr-users - kamailio.org

ERROR: dispatcher [dispatch.c:1402]: ds_load_remove(): cannot find load for
by Will Ferrer 22 Dec '14

22 Dec '14

Hi All We recently discovered that our dispatcher boxes have this error constantly at the end of our calls. ERROR: dispatcher [dispatch.c:1402]: ds_load_remove(): cannot find load for (<callid goes here>) We aren't seeing any issues arising from it that we can notice but wanted to see if any one knew what it meant and what it might cause. Thanks much in advance. All the best Will Ferrer Switchsoft Inc

2 7

Kamailio not start if redis server is down
by pars3c 22 Dec '14

22 Dec '14

Hi, i'm trying to use the ndb_redis connector , but i've a problem when redis server is down and i try to restart kamailio. Kamailio in this case not start and release this log: 2(19590) ERROR: ndb_redis [redis_client.c:112]: redisc_init(): error communicating with redis server [srv1] (xxxxxx/0): Connection refused 2(19590) ERROR: ndb_redis [ndb_redis_mod.c:126]: child_init(): failed to initialize redis connections 2(19590) ERROR: <core> [sr_module.c:927]: init_mod_child(): init_mod_child(): Error while initializing module ndb_redis (//lib/kamailio/modules/ndb_redis.so) 30(19622) ERROR: <core> [tcp_main.c:4962]: tcp_init_children(): ERROR: tcp_main: fork failed: Connection refused 31(19624) : <core> [pass_fd.c:293]: receive_fd(): ERROR: receive_fd: EOF on 11 31(19624) : <core> [pass_fd.c:293]: receive_fd(): ERROR: receive_fd: EOF on 31 0(19588) ALERT: <core> [main.c:775]: handle_sigs(): child process 19589 exited normally, status=255 0(19588) ALERT: <core> [main.c:775]: handle_sigs(): child process 19590 exited normally, status=255

2 2

rtpengine stats
by Marc Soda 22 Dec '14

22 Dec '14

Does anyone know how a can get stats from rtpengine? I see the $rtpstat pseudo variable in Kamailio, but from the documentation it looks like that will only give me stats on a particular call. I'm looking for overall stats like concurrent calls, bandwidth, etc... Thanks, Marc

3 2

Multi-tenant SBC Kamailio
by Kalala Alexander 22 Dec '14

22 Dec '14

3 5

Re: [SR-Users] sr-users Digest, Vol 115, Issue 16
by Kees Meijs - SIGNET B.V. 22 Dec '14

22 Dec '14

Hi there, We haven't used gdb on Kamailio yet but some time soon we'll most likely will. Obviously I'll post our findings. Thanks again! Cheers, Kees On 17-12-14 12:00, sr-users-request(a)lists.sip-router.org wrote: > did you get a core dump file? If yes, send the full backtrace taken with > gdb.

2 1

kamailio dispatcher module stops processing REGISTER
by José Seabra 22 Dec '14

22 Dec '14

Hello there, I'm doing stress testing against kamailio (sip load balancer role) sending sip registrar, my stress test software is sending 50 registrations per second, after 30 seconds and 1000 registers processed the function ds_select_dst (I'm using algorithm 4) returns null for all registers sent from my software, and all registrar sip servers are available and in active mode, if I reduce the software to send 20 registrar's per second, the issue only happens after 60 seconds and kamailio (sip load balancer role) process 4000 registrers, after this, function ds_select_dst starts returning null Anyone has any idea why it is happening? My kamailio version is 4.2 Best Regards -- Cumprimentos José Seabra

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The "Subscriber" table VS the "Location" table ?!
by Mahmoud Ramadan Ali 20 Dec '14

20 Dec '14

Hi Dears, I'm a little bit confused about the difference between the "Subscriber" table and the "Location" table.I read that the "Location" table is used for persistent user registration BUT i did NOT configured the "Location" table and can get persistent user registration with the "Subscriber" table ! So can anyone explain the difference please ? Best Regards and Thanks in Advance.

3 2

rtpengine returns 0.0.0.0 even with sendrecv in offer
by Juha Heinanen 19 Dec '14

19 Dec '14

i got mozilla to generate sdp with sendrecv, but still rtpengine does not replace 0.0.0.0 address on o and c lines. why? -- juha Dec 19 10:20:18 box /usr/bin/sip-proxy[5841]: INFO: ===== rtpengine_offer(ICE=force replace-session-connection replace-origin via-branch=1) Dec 19 10:20:18 box rtpengine[20171]: Got valid command from 127.0.0.1:39586: offer - { "sdp": "v=0#015#012o=Mozilla-SIPUA-34.0 25949 0 IN IP4 0.0.0.0#015#012s=SIP Call#015#012t=0 0#015#012a=ice-ufrag:f5511cb8#015#012a=ice-pwd:1d062e6b56db4f738c7613bc97086194#015#012a=fingerprint:sha-256 45:CA:2E:A5:12:4E:80:0C:AD:9D:11:94:C5:2D:ED:15:55:FD:83:70:91:59:3D:57:3D:F9:A5:77:74:85:BD:E8#015#012m=audio 9 RTP/SAVPF 109 9 0 8 101#015#012c=IN IP4 0.0.0.0#015#012a=rtpmap:109 opus/48000/2#015#012a=ptime:20#015#012a=rtpmap:9 G722/8000#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:101 telephone-event/8 ... Dec 19 10:20:18 box rtpengine[20171]: ... 000#015#012a=fmtp:101 0-15#015#012a=sendrecv#015#012a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level#015#012a=setup:actpass#015#012a=rtcp-mux#015#012a=candidate:0 1 UDP 2122252543 192.168.1.52 42190 typ host#015#012a=candidate:0 2 UDP 2122252542 192.168.1.52 54801 typ host#015#012a=candidate:1 1 UDP 1686110207 192.98.100.128 42190 typ srflx raddr 192.168.1.52 rport 42190#015#012a=candidate:1 2 UDP 1686110206 192.98.100.128 54801 typ srflx raddr 192.168.1.52 rport 54801#015#012", "ICE": "force", "replace": [ "session-connection", "origin" ], "call-i ... Dec 19 10:20:18 box rtpengine[20171]: ... d": "lr92g1vkbkvmgdt9m1dg", "via-branch": "z9hG4bK2313203", "received-from": [ "IP4", "192.98.100.128" ], "from-tag": "d03ug77pi6", "command": "offer" } Dec 19 10:20:18 box rtpengine[20171]: [lr92g1vkbkvmgdt9m1dg] Creating new call Dec 19 10:20:18 box rtpengine[20171]: [lr92g1vkbkvmgdt9m1dg] Opened ports 8056..8057 for media relay Dec 19 10:20:18 box rtpengine[20171]: [lr92g1vkbkvmgdt9m1dg] Opened ports 8012..8013 for media relay Dec 19 10:20:18 box rtpengine[20171]: [lr92g1vkbkvmgdt9m1dg] Returning to SIP proxy: d3:sdp725:v=0#015#012o=Mozilla-SIPUA-34.0 25949 0 IN IP4 0.0.0.0#015#012s=SIP Call#015#012t=0 0#015#012a=ice-lite#015#012m=audio 8056 RTP/SAVPF 109 9 0 8 101#015#012c=IN IP4 0.0.0.0#015#012a=rtpmap:109 opus/48000/2#015#012a=ptime:20#015#012a=rtpmap:9 G722/8000#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-15#015#012a=sendrecv#015#012a=rtcp:8057#015#012a=rtcp-mux#015#012a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:8FwYLdKkGvVH4Ykc+kTgrFN1Cwz0KZVpT2JK7+g0#015#012a=setup:actpass#015#012a=fingerprint:sha-1 ED:DE:59:BA:F7 ... Dec 19 10:20:18 box rtpengine[20171]: [lr92g1vkbkvmgdt9m1dg] ... :F8:78:9D:1D:78:42:AA:00:6C:88:85:A4:2B:36:42#015#012a=ice-ufrag:4dmKpmV3#015#012a=ice-pwd:75jY5mekvVULGZnnB9mJmHQoVa9i#015#012a=candidate:bVSyZZLCFRG94R1D 1 UDP 2130706431 192.26.111.29 8056 typ host#015#012a=candidate:bVSyZZLCFRG94R1D 2 UDP 2130706430 192.26.111.29 8057 typ host#015#0126:result2:oke

2 7

Initial REGISTER request/response in 3rd Party Register
by jyaim 19 Dec '14

19 Dec '14

Hello, I'm using Kamailio with a SIP Application Server, when a user registers on the IMS Core a 3rd Party REGISTER request is sent to the application server to start some logic. I'm trying to add in the 3rd party register request body the initial REGISTER request sent by user device, and the 200 OK response sent back by Kamailio. Do you think it is possible to configure Kamailio to get this behaviour ? Thank you!

1 0

Accounting: syslog vs db_flatfile
by DanB 19 Dec '14

19 Dec '14

Hey Guys, We are in the process to review the interoperability between Kamailio and CGRateS and was wondering what is the best/most recommended/maintained way to generate accounting from Kamailio: * Syslog * db_flatfile * other way maybe? Is CDR generation planned for files already, or just accounting events? Thanks in advance! DanB

6 8

SIP Fragments
by Marc Soda 19 Dec '14

19 Dec '14

I'm having a problem reassembling UDP packets on my Asterisk servers after passing through Kamailio (it appears to me an OS level issue, nothing to do with Kamailio). Is there a way, with Kamailio, to limit the size of a SIP message? I know I can just start removing headers, but that doesn't seem like a realistic solution. I see that Kamailio can compress the message body, but can Asterisk handle that? How do other people handle this? Thanks in advance, Marc

7 13

webrtc clients support using rtpengine
by Andrey Utkin 19 Dec '14

19 Dec '14

Hi! I need to establish calls between WebRTC and usual SIP clients (exactly, sipml/jssip and linphone-android). I used configs from https://github.com/caruizdiaz/kamailio-ws and latest git master HEAD of both kamailio and rtpengine. I got calls from webrtc to android working correctly (but only with Firefox browser), even with video. But in other directions i have some issues because of lack of RTP delivery or RTP timeouts. I have some logs to show you regarding this: https://gist.github.com/krieger-od/27c6f3e4924f5e21352e (works), https://gist.github.com/krieger-od/196bcfbd331d621427ef (doesn't work). I would really love to get some quick help from anyone. For direct manual fixing, I can give a couple of hundreds of bucks. Looking forward impatiently for reply from anyone having something to say. -- Andrey Utkin

3 10

Kamailio World 2015 – Call For Speakers
by Daniel-Constantin Mierla 18 Dec '14

18 Dec '14

Hello, a short note to announce for everyone interested that submissions of presentation proposals is now open for Kamailio World 2015 (Berlin, May 27-29). More details at: - http://conference.kamailio.com/k03/2014/12/kamailio-world-2015-call-for-spe… Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Kamailio CDRs
by Massimo Varriale (IPZeta) 18 Dec '14

18 Dec '14

Hi Guys! I'm trying to store CDRs for Kamailio calls. Following Siremis tutorial (http://siremis.asipto.com/install-accounting/) I'm able to gather Succesfull Calls and using the MySQL stored procedure I can have a CDR table. But what I can see now is that only outbound calls with a duration are stored and that failed attempts are not stored into the database. How to capture those attempts too? Here is my accounting configuration: # ----- acc params ----- modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) modparam("acc", "detect_direction", 0) modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;dst_user=$rU;dst_domain=$rd;src_ip=$si") modparam("acc", "cdr_enable", 1) #!endif Thanks Max

2 3

Any extension or solution on how to implement RPC over SIP
by Thierry Luo 18 Dec '14

18 Dec '14

Dear all, I'm trying to design a SIP based framework that can execute RPC end-to-end. Session is set up based on SIP, and then one user agent can remote call a method into the peer user agent. Anyone have any solution on this? Thanks. Best Regards. Thierry Luo

1 0

Maintenance work on kamailio.org - Wed, Dec 17
by Daniel-Constantin Mierla 17 Dec '14

17 Dec '14

Hello, on Wednesday (Dec 17) late afternoon in Europe (probably starting around 18:00, Berlin time), we plan some maintenance work on kamailio.org server. Hopefully the downtime will be short and barely noticeable. But as we know, sometime we have to expect the unexpected, therefore in case the server is unavailable a bit longer, keep in mind that it is a scheduled work. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 2

bug in allow_source_address?
by Henry Fernandes 17 Dec '14

17 Dec '14

I¹m using allow_source_address in my Kamailio config and have been for months/years. I¹ve had no issues with it until today. The issue is that allow_source_address("1²) returns ³1" (for $si=123.17.29.201 and $sp=5060) even though we don¹t have this IP address in the address¹ table for group 1. We have defined this IP address for group 2 but not group 1. select * from address order by mask; +----+-----+---------------+------+------+----------+ | id | grp | ip_addr | mask | port | tag | +----+-----+---------------+------+------+----------+ | 20 | 1 | 123.17.0.0 | 23 | 5061 | Testing | | 22 | 1 | 123.17.0.0 | 23 | 5062 | Testing | | 23 | 1 | 123.17.0.0 | 23 | 5063 | Testing | | 24 | 1 | 123.17.0.0 | 23 | 5064 | Testing | | 1 | 1 | 123.17.0.0 | 23 | 5060 | | | 4 | 1 | 20.5.176.0 | 24 | 5060 | | | 21 | 1 | 123.17.1.235 | 32 | 5060 | Override | | 46 | 2 | 123.17.29.201 | 32 | 5060 | customer | +----+-----+---------------+------+------+----------+ Can anyone explain this? I¹m wondering if there¹s a bug in how allow_source_address does its net mask calculations. -H

1 0

Pseudo-Variables Digest
by Kalala Alexander 17 Dec '14

17 Dec '14

2 2

What is raw 0.0.0.0:255 pn netstat??
by Luis Jimenez 17 Dec '14

17 Dec '14

Hello, i just installed Kamailio 4.2.1 on debian Wheezy, would like to know why when Kamailio starts i see this on netstat -an ?? raw 0 0 0.0.0.0:255 0.0.0.0:* 7 is it normal?? why? Thanks in advance

3 3

How to Fix the RTP Engine SDP Rewriting direction.
by Mahmoud Ramadan Ali 16 Dec '14

16 Dec '14

Hi Dears, I'm working on integrating the rtpengine to work with Kamailio as RTP proxy and i have successfully configured the rtpengine in multi home mode to proxy the media and rewrite the SDP message whenever it passes trough the Kamailio internal interface to the external interface and vice versa using this command. root@debian:/usr/local/etc/kamailio# rtpengine --interface=Kamailio-Internal/192.168.100.1 --interface=Kamailio-External/ 192.168.50.1 --listen-ng=127.0.0.1:22222 --pidfile=/var/run/ngcp-rtpengine-daemon.pid Also i have configured the "i" and "e" flags in the route[NATMANAGE] to take care of the SDP rewriting direction the same way it worked before with my rtpproxy module configuration. But the issue now is that whenever i make a call from the external to the internal the rtpengine can NOT detect the right direction as colored in red below so my question now is : How to set the direction of the rtpengine to designate that the interface 192.168.50.1 is the external and 192.168.100.1 is the internal ? Dec 16 09:28:09 debian rtpengine[4714]: Got valid command from 127.0.0.1:50639: offer - { "sdp": "v=0#015#012o=- 13063213690566395 1 IN IP4 192.168.50.2#015#012s=X-Lite release 4.7.1 stamp 74247#015#012c=IN IP4 192.168.50.2#015#012t=0 0#015#012m=audio 56078 RTP/AVP 125 100 0 9 8 101#015#012a=rtpmap:125 opus/48000/2#015#012a=fmtp:125 useinbandfec=1#015#012a=rtpmap:100 speex/16000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-15#015#012a=sendrecv#015#012", "direction": [ "internal", "external" ], "flags": [ "asymmetric", "trust-address", "symmetric" ], "replace": [ "session-connection", "orig ... Best Regards and Thanks in advance.

2 1

IPv6 socket issues
by Kees Meijs - SIGNET B.V. 16 Dec '14

16 Dec '14

Good evening, During pre-production testing, we discovered quite odd IPv6 socket issues. Problems arise when a SIP dialog is initiated from within Kamailio (e.g. a SIP INVITE message). Using ngrep, it seems very odd IPv6 sockets are opened, for example: > U 2001:beef:34a:10::234:5060 -> 2001:beef:285:1::3:5060 1573541542@0:1440 > U 2001:beef:34a:10::234 -> 2001:beef:285:1::3 +1573541542@1448:17 Obviously the "1573541542@0:1440" part is bogus, I expect output similar to: > U 2001:beef:34a:10::234:5060 -> 2001:beef:285:1::3:5060 > INVITE sip:s@[2001:beef:285:1::3]:5060 SIP/2.0. When I do a REGISTER from UA using IPv6, the dialog (and return packets) seem completely normal. Using IPv4, there's no problem at all. It seems the socket problems are triggered when: 1. Using the topoh module. 2. When altering From: headers e.g. using uac_replace_from(). When topoh is disabled and no headers are altered, IPv6 works as good as IPv4. We could reproduce the problem using Kamailio 4.1.4, 4.1.5 and 4.2.0 on Debian (64-bit). Please advice what to do. Thanks in advance! Best regards, Kees

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Re: [SR-Users] sr-users Digest, Vol 115, Issue 15
by Kees Meijs - SIGNET B.V. 16 Dec '14

16 Dec '14

Hi Daniel, Thank you very much for your prompt response. The output mentioned is generated by ngrep and can be seen by tcpdump as well. It seems a "space" and the bogus part ends up after the port used in the socket opened. For example "5060" becomes "5060 yadieyaboguspartdrdfg5r3245234". Obviously sending a packet towards an IPv6 address using such a port doesn't work. For the time being, we disabled the topoh.so module and rewrote code using uac_replace_from() and substituded the code with $fn = "string", $fd = "otherstring" equivalents. Thanks again! Cheers, Kees

1 0

Kamailio AVPs Radius
by Kalala Alexander 16 Dec '14

16 Dec '14

Radius server sends a response containing an attribute (Session-Timeout (27): 2342). How is the value saved in the AVPs using module auch_radius? Config: modparam("auth_radius", "radius_config","/etc/radiusclient-ng/radiusclient.conf") modparam("auth_radius", "service_type",15) modparam("auth_radius", "use_ruri_flag", 22) modparam("auth_radius", "auth_extra", "Calling-Station-Id=$fU;Called-Station-Id=$tU;Acct-Session-Id=$ci") if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!radius_www_authorize("$td")) { www_challenge("$td", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } ............................... if (from_uri==myself) { if (!radius_proxy_authorize("$fd", "$fU")) { proxy_challenge("$fd", "0"); exit; } -- С уважением Инженер по телекоммуникациям Калала Александр vel: 375291146285 life: 375256819996 skype: klistrod VoIP Network Engineer dCAP, CCNAVoice Kalala Alexander http://callcenters.by/ http://skytel.by/ http://voiplab.by/ http://asterisk-pbx.by/

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New reminder: about private emails
by Daniel-Constantin Mierla 16 Dec '14

16 Dec '14

Given that I noticed still a lot of emails sent directly after a discussion started on public mailing lists (subject with mailing lists prefix, but the sender header is not the mailing list daemon), I would like to remind that, at least in my case, writing emails directly without cc-ing the mailing list is highly not recommended. The rule is also suggested in the mailing lists presentation page: - http://www.kamailio.org/w/mailing-lists/ Unless I asked explicitly for a private email with some specific details, the message will not be noticed in decent time and therefore not replied. I use this email address for many public mailing lists and it is intended only for that usage, I am reacting based on the filters I created, unmatched messages getting the lowest priority. Therefore if you wrote me and expect a response, re-send the message to the appropriate public mailing list. Checking and answering the messages on Kamailio mailing lists are among the top of priorities on my daily activity, so it is high chance you get the response faster when you write there. Note that you can always mask sensitive data (e.g., replace passwords, IP addresses) before sending logs or config file. There are several common sense as well as technical reasons for keeping the discussion on public mailing lists. Here are some along with other suggestions to improve the likeliness of getting an answer: - we are not answering questions on mailing lists to help only one person, but also other people that may have similar issues in the future -- they can find the answers in the archive with a search engine. Also, other people subscribed to the list can learn from conversation or contribute to the discussion, resulting in better outcome - the amount of unexpected messages is very high -- it is practically a zero chance to get to the folder with messages that didn't match any filter (thus your message is not going to be replied) - whenever I find time, the first emails I answer are those coming on mailing lists. They are many there as well, people are traveling or having other personal or business projects, so sometime is good to send a reminder if a question does not get an answer - starting with negative approach or no technical content, like you are going to use something else if you don't get this instantly, or this application is missing what so ever tutorial giving exactly the solution you need, or simply asking for full config of complex requirements, is not helping at all, but actually making people to ignore your messages. Start with what you tried and where you got stuck. Everything you get is for free, have appropriate expectation and behavior. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Email notifications from Github commits
by Daniel-Constantin Mierla 15 Dec '14

15 Dec '14

Hello, those on sr-dev mailing list probably noticed the last commit notification - the email format can be seen at: - http://lists.sip-router.org/pipermail/sr-dev/2014-December/026513.html As discussed in the past, it tries to stay as close as possible to the old notification message, but it has some differences. I published the php script I use for generating this notification as well as synchronizing the git repository mirror on kamailio.org. It is available at: - https://github.com/miconda/gitpushub If people prefer, we can make a clone inside kamailio organization and tailor it to be more specific for kamailio repository, with more people having direct access to update it. On the other hand, any contribution to the script to make it better or more useful is welcome! Eventually it is going to be useful for various people with their other projects. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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