- sr-users - kamailio.org

Newbie starting - looking for docs / tutorials
by Antony Stone 28 Feb '17

28 Feb '17

Hi. I'm just starting with Kamailio - I've been to a few presentations about it, at Fosdem and KamailioWorld, but never used it so far. I'm pretty familiar with Asterisk, and I want to use them in combination. My first use case is to handle SIP registrations (both as a client, registering to a remote SIP server, and as a server, accepting registrations from handsets) in Kamailio instead of Asterisk, and still be able to route calls to/from the registered accounts through the Asterisk dialplan. I suspect this is a pretty common starting point for people who've used Asterisk and want to add Kamailio to the setup? So far I've been unsuccessful finding any Howto or Tutorial explaining how to do this - I can't help feeling I must simply be looking in the wrong place or in the wrong way. Searching for topics like "Asterisk Kamailio Getting Started" I find: http://kamailio.org/docs/ser-getting-started/SER-GettingStarted.pdf but with 12 chapters and 85 pages I'm not sure which bits are the essential parts to do what I need. I've also found http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x- asterisk-11.3.0-astdb but this seems to focus on a database backend, which I guess I'll want to use in due course, but doesn't seem the simplest place to start just to learn how Asterisk & Kamailio work together? So, please can someone suggest a beginner's guide to introducing Kamailio into an Asterisk setup, keeping things as simple as possible to start with, so I can learn about Kamailio one bit at a time (since there do seem to be a lot of bits to it) from a familiar environment of using it with Asterisk? Thanks in advance, Antony. -- What do you get when you cross a joke with a rhetorical question? Please reply to the list; please *don't* CC me.

2 3

multi domain kamailio
by przeqpiciel 28 Feb '17

28 Feb '17

Let's suppose that i have two machines with installed asterisk and one with kamailio. I would like have routing based on sip domain. for domain sip.domain1.com send sip signalling to asterisk#1 server and for sip.domain2.com send to asterisk#2 server. What i should use? Domain module or what ? Tomasz

3 4

Kamailio v5.0.0 Released - new major version is out
by Daniel-Constantin Mierla 27 Feb '17

27 Feb '17

Kamailio v5.0.0 is out – it comes with 6 new modules and a consistent set of improvements touching more than 50 existing modules. You can read detailed release notes at: * https://www.kamailio.org/w/kamailio-v5-0-0-release-notes/ Many thanks to all developers and community members that made possible this release. A consistent effort was directed to restructuring the source code tree to meet more modern patterns; introducing the KEMI framework that allows writing SIP routing blocks in other embedded interpreters such as Lua, JavaScript or Python; removal of obsoleted MI control mechanism, being replaced by RPC. Enjoy Kamailio v5.0.0! Thank you for flying Kamailio! Daniel -- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference 2017 - http://www.kamailioworld.com

1 0

problem with avpops and mongodb modules
by hdssdsdsdsfsdf hdssdsdsdsfsdf 27 Feb '17

27 Feb '17

1 1

Releasing Kamailio v5.0.0
by Daniel-Constantin Mierla 27 Feb '17

27 Feb '17

Hello, I am considering to release Kamailio v5.0.0 at the beginning of next week, depending how things evolve in the next days, either on Monday, Feb 27, or Tuesday, Feb 28. I haven't noticed any new issues specific for the new features added for 5.0 lately, so it looks pretty good so far. If you are aware of issues not reported to the bug tracker, register them as soon as possible at: - https://github.com/kamailio/kamailio/issues There are still some parts of documentation that have to be updated, I am planning to work on them during the next days, but help from others are very appreciated, especially details of what you had to do changes while upgrading from 4.4 to 5.0. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Mar 6-8 (Europe) and Mar 20-22 (USA) - www.asipto.com Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 1

Parallel Forking with Different Call-Id
by SamyGo 24 Feb '17

24 Feb '17

Hi, I've a very strange scenario to work on which requires me to parallel fork the call to the same Destination provider. The only problem here is that they think that the second INVITE with different branch tag is a re-transmission and hence only take one call forward. I do not have to modify any R-URI or any headers, hence just 'append_branch() before t_relay() is in the code. ... $ru = "sip:" + $rU + "@" + $avp(carrier_ip) + ":" + $avp(carrier_port); append_branch(); route(RELAY); ... *Question:* Is there anyway possible I can change the CallID of the forked INVITEs ? I have tried using Topoh module, but it still puts the same CallID before sending out. Different from the A-leg but the Sent out INVITEs have same Call-ID value. In other weird scenario, I've also tried branching, and looping call within Kamailio before sending out to carrier hoping that Kamailio would treat the two different calls and Topoh would change the Call-ID on both INVITEs before sending out.. ... if(is_present_hf("X-FORKED")) { $ru = "sip:" + $rU + "@" + $avp(carrier_ip) + ":" + $avp(carrier_port); route(RELAY); }else { append_hf("X-FORKED: 1\r\n"); append_branch(); route(RELAY); } ... Still I get same Call-ID on outgoing branched call. I could branch out one INVITE to a MediaServer, say FreeSWITCH/Asterisk but again I don't want to have that component bottleneck the throughput. That could be my very last option. Looking for some ideas. Regards, Sammy

3 6

Kamailio 4.3.x location_attrs update issue
by Mridul B 24 Feb '17

24 Feb '17

Hi, We've noticed a case where the *location_attrs* table isn't updated when there's an unexpected shutdown of a SIP device (here there'll be no REGISTER message sent with expires=0). We're using Kamailio v4.3.x When the SIP device shuts down ungracefully. The location table is updated using DELETE query This is done by the timer which deletes the expired contacts, by default the timer_interval is set as 60. Here, the logs of auto-removal of expired contacts entries by the timer > Feb 17 17:16:13 localhost kamailio[23273]: DEBUG: db_postgres > [km_dbase.c:272]: db_postgres_submit_query(): sending query ok: > 0x7f34969433c8 (1) - [delete from location where expires<'2017-02-17 > 17:16:14' AND expires<>'1970-01-01 00:00:00'] > Feb 17 17:16:14 localhost kamailio[23272]: DEBUG: db_postgres > [km_dbase.c:272]: db_postgres_submit_query(): sending query ok: > 0x7f34969433c8 (1) - [delete from location where expires<'2017-02-17 > 17:16:15' AND expires<>'1970-01-01 00:00:00'] The delete operation is only on location table, shouldn't we have the same implementation on the location_attrs table based upon ruid or username? We're getting duplicate entries in the location_attrs table of the same sip user after it registers. But when the SIP device un-registers, the entries in the location as well as in the location_attrs table is deleted Feb 17 18:14:32 localhost kamailio[23249]: DEBUG: usrloc [ucontact.c:1725]: > uldb_delete_attrs(): trying to delete location attributes > Feb 17 18:14:32 localhost kamailio[23249]: DEBUG: db_postgres > [km_dbase.c:272]: db_postgres_submit_query(): sending query ok: > 0x7f3496941888 (1) - [delete from location_attrs where username='mridul' > AND ruid='uloc-58a72243-5ace-2'] > Feb 17 18:14:32 localhost kamailio[23249]: DEBUG: db_postgres > [km_dbase.c:272]: db_postgres_submit_query(): sending query ok: > 0x7f3496941888 (1) - [delete from location where username='mridul' AND > contact='sip:mridul@127.0.0.1:37881;rinstance=b841a8c916e47dfa'] I think the usrloc module should have the implementation of deleting the entries of expired sip users based on its ruid/username in the location_attrs table also. Thanks for the help. Best regards, MRIDUL

2 3

want to see active users SIP users
by Mandar Nagarkar 24 Feb '17

24 Feb '17

Hello Everyone, I want to see active users SIP users for this purpose I checked the location table of kamailio database it does have active users entries however as per my observation there is 45 secs to 1 min delay before the entry actually added to this table i.e. suppose if I am registering the user through pjspi,jssip or any client I cannot see entry for that user in location table. I have to wait up-to 1 min to see the user in location table. Is there any way we can Identify if user is online without delayed entry? Regards, Mandar nagarkar The information in this email is confidential and may be legally privileged. It is intended solely for the addressee and access to it by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, distribution or any action taken or omitted to be taken based on it, is strictly prohibited and may be unlawful.

2 1

Fetching proxy-autheticate header from 407 Response
by vivek＠advaitamtech.com 23 Feb '17

23 Feb '17

Hi All, Am not using the authentication in the kamailio. So I will be receiving the "407 Proxy Authentication Required" response from the third party server for the first INVITE. I want to fetch the Proxy-Authenticate header present in the 407 response. I browsed and tried using the pseudo-variables "$adu/$aa/$ar". It dint work. Please let me know if there is a way to fetch this header. Thanks, Vivek.

3 4

How to release call without having to use fromTag and toTag, and resume a suspended call
by Cassandra Yurekli 23 Feb '17

23 Feb '17

To whom it may concern, There are several points I don’t quite understand regarding a couple of kamailio modules. I have a main application which controls calls by sending command messages via a socket (for example reject this sip invites). I communicate with the socket client via evapi module and using evapi_asynch_relay() in order to be sure that communication are suspended and kamailio waiting an order for doing something with them (Accepte, reject, release, ...). I want to release a call, using a command “release this call”. I’m actually doing this with dialog module (using get_dialog(callid,fromTag,toTag) and dlg_bye(‘all’)) but is there anything else (like another module) which could release a call without necessarily having to use fromTag and toTag to get the context ? I didn’t find it on modules documentations. Also after an INVITE I’m waiting for the main app to accept or reject the call. For accepting and continuing the transaction I found “t_continue(tindex,tlabel,route)” on tmx module, but at this moment $T(id_index) and $T(id_label) are null, is it normal or I did something wrong ? How can I simply resume the suspended call ? Thanks; Best regards,

2 1

Need help with tcpops
by Abdoul Osséni 22 Feb '17

22 Feb '17

Hi, My call flow is: Softphone --> TLS ---> Kamailio --> UDP --> ASTERISK --> PSTN I want to use tcpops module between the softphone and kamailio. http://www.kamailio.org/docs/modules/4.4.x/modules/tcpops - enable tcpops : no problem - disable tcpops : two cases - cancel or bye from softphone: no problem. - cancel or bye to softphone via kamailio: how disable tcpops? I can't use $avp(bye_conid) because it is asterisk thant sending the sip message. Can we use $avp(caller_conid)? In my case, the softphone is always at the origin of the call establishment. Regards Abdoul.

2 1

Automated integration tests for individual modules
by A Lai 22 Feb '17

22 Feb '17

Hi, I just started working with the Kamailio code base. I am wondering if there are some common strategies in having automated tests on proper functioning of modules when loaded into cfg.

2 1

problem with avpops and mongodb modules
by hsdsdssdfsdf dsasd 22 Feb '17

22 Feb '17

2 1

How to handle CANCEL if corresponding INVITE transaction is missing
by Venkatesh Macha 22 Feb '17

22 Feb '17

Hi all, I am using Kamailio 4.4.5, It is working fine. But in one case. If i make a video call between the two users and cancels the call then Call is not disconnecting on the callee side. Please look the following scenario. 1000 made one video call to 2000 2000 is ringing. Now 1000 disconnected the call by sending CANCEL. But 2000 is not disconnecting. This is happening because Kamailio is unable to find the corresponding INVITE transaction for the CANCEL so it is simply dropping the requests. Why Kamailio unable to find the transaction. And is there any possibility to handle this case and send the CANCEL to callee side. I tried with forwad() and someother things but it is just generating the loops. I am also attaching the ngrep trace. Thank you in Advance. CANCEL_not_working_nabble.CANCEL_not_working_nabble <http://sip-router.1086192.n5.nabble.com/file/n156243/CANCEL_not_working_nab…> -- View this message in context: http://sip-router.1086192.n5.nabble.com/How-to-handle-CANCEL-if-correspondi… Sent from the Users mailing list archive at Nabble.com.

2 2

kamailio memory leak in private memory
by david 22 Feb '17

22 Feb '17

Hello all We are observing an issue in one kamailio proxy we use. The consecuence of an unknow cause is that the private memory of some UDP receiver processes in where we listen from a particular IP, seems to be draining along the time while processing traffic. Quite "slowly" but maybe aroung 200K in 3-4hours, depending on the moment of the day. This is quite weird for us, because it seemed like it was caused by some particular traffic profile, since we have several IP listening (8 children each) and we only see the issue in the UDP receiver processes of some of those IPs. Besides we have moved the kamailio service listening on the same IP to several different physical servers, and the behaviour is the same. We look at it ussing pkg.stats kamcmd command and for instance if we track one of the process affected pid: 23784 rank: 53 used: 2838616 free: 13005408 real_used: 3771808 pid: 23784 rank: 53 used: 2842584 free: 12999280 real_used: 3777936 This 2 outputs were taken with 10min diff between each other. But we dont know where or how to look at it. We have been these last days trying to search for patterns to allocate some particular traffic source causing something weird, but no luck. We have checked all the SIP INVITE messages looking for anything weird at any field but no luck, besides we sometimes see some bad headers with blank spaces in the uri or thing like that but those cases we have an error and we dont think it can cause a memory leak. Shared memory seems to be stable. We also have checked the config we are using and we have been discarding possible issues since we dont think the $var variables we are using could cause that issue (but you never know....) We tried to debug the kamailio on one affected IP compiling it with DBG_SYS_MALLOC , and doing the dump of the pkg mem status when the kamailio was restarted and no traffic was processed, and also when it had traffic processed. in this last case, we removed traffic, waited for 20min and made the dump i attach you both files, we have the debug logs, but they are very big to be passed here, and we dont know what to look for :) could you please give us some guidance or clue to at least start to searching for something that can cause this issue? thanks alot and regards david

3 6

parse_addr_spec(): unexpected char [^M] in status 6:
by Abdoul Osséni 21 Feb '17

21 Feb '17

Hello, Could you help us to fix this error? <core> [parser/parse_addr_spec.c:662]: parse_addr_spec(): unexpected char [^M] in status 6: Regards Abdoul.

3 5

Transaction-persistent data and uac_req_send()
by Alex Balashov 21 Feb '17

21 Feb '17

Hello, I am using uac_req_send() to send INVITEs to a redirect server. However, upon receiving the redirect, I need access to some state information that was last available at the time of sending the request. Does uac_req_send() allow me to persist AVPs like a normal TM transaction? If not, what are my options, other than building my own state using htable or what have you? I would really rather not maintain—and be responsible for garbage-collecting—externally constructed state like that if possible. 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/

3 10

Kamailio 4.4 with 2 Asterisk 13
by Graham Brown 20 Feb '17

20 Feb '17

I'm trying to load balance 2 Asterisk servers with Kamailio offloading the registration to Kamailio using the Asterisk real-time database tables. I have a problem with the dispatcher module, if there is a t_on_failure route it always goes into that, if I remove that then I would expect every call to round robin but I'm only ever hitting the same server from the same extension (seems like sticky session based on source ip). 1. how do I fix my failure route so I can reboot server 1 and server 2 will be used 2. how to I change the sticky session behaviour. Another issue I am facing with extensions registering to Asterisk are only avaliable on the server they routed to My assumption was if using the Asterisk real-time database I would get the contact on both servers. Maybe my architecture is incorrect, I'm open to suggestions. Currently Kamailio is not storing locations (assuming that is the extension contact) maybe that is a better solution, but I dont know how to configure Asterisk to forward everything to Kamailio then or integrate via the database or something. Thanks in advance for any help Regards Graham

2 1

Cleaning wiki accounts
by Daniel-Constantin Mierla 20 Feb '17

20 Feb '17

Hello, I noticed a lot of random usernames for the accounts created to edit the wiki. Apart of several which did some which did some edits to add spam, most of them are inactive. Not sure what would be the real purpose behind them, but I am going to remove anything that doesn't look as a community user. Shall I remove an account with random id, but for a good user, let me know, it can be readded/account recreated. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Mar 6-8 (Europe) and Mar 20-22 (USA) - www.asipto.com Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 1

memory: Free fragment not found
by Dmitri Savolainen 17 Feb '17

17 Feb '17

Hi. My kamailio v4.4.5 installation ( -m 512 -M 64 ) stop sip processing with repeatable error: Feb 15 21:45:05 kamailio[14173] ERROR: <core> [mem/f_malloc.c:415]: > fm_search_defrag(): fm_search_defrag(0x7f39917fa010, 952); Free fragment > not found! > Feb 15 21:45:05 kamailio[14173] ERROR: <core> [mem/f_malloc.c:498]: > fm_malloc(): fm_malloc(0x7f39917fa010, 952) called from core: rvalue.c: > rval_new_empty(224), module: > core; Free fragment not foun while it was enough free shared memory according to kamcmd> core.shmmem m > { > total: 512 > free: 472 > used: 27 > real_used: 39 > max_used: 51m > fragments: 37035 > } whether it was about package memory or smth else? Thanks! -- Savolainen Dmitri

2 4

MSILO offline messages are coming in reverse order.
by Venkatesh Macha 17 Feb '17

17 Feb '17

Hi all, I am using MSILO module for the offline messaging feature. It is working good. But If I send more than one message to the offline user. Those messages are delivering in the reverse order. Please consider the following scenario. A sending B ( B is offline, so messages are stored in DB) hi hello how are you Now B comes online and registers. Kamailio sending the messages but App is getting them in reverse order. how are you hello hi Is there any possibility to tell Kamailio to send them in order. ? Thank you in advance. -- View this message in context: http://sip-router.1086192.n5.nabble.com/MSILO-offline-messages-are-coming-i… Sent from the Users mailing list archive at Nabble.com.

2 2

Problem with dispatcher on Kamailio 4.2 and Debian 8
by przeqpiciel 17 Feb '17

17 Feb '17

Hi, I'm trying to setup dispatcher module to working with mysql database but when i put to kamailio.cfg below lines then kamailio wont starts. loadmodule "dispatcher.so" modparam("dispatcher", "db_url", "mysql://kamailioro:kamailioro@localhost /kamailio") modparam("dispatcher", "table_name", "dispatcher") i tried connect to mysql with credentials which you can see in mod_param line and i could query all kamailio database. im really new in this and i dont know how to debug this, is anybody could help me with it ? Best regards, Tomasz

2 4

Kamailio 4.2 on Debian 8 and dispatcher problem
by przeqpiciel 16 Feb '17

16 Feb '17

Hi, I'm trying to setup dispatcher module to working with mysql database but when i put to kamailio.cfg below lines then kamailio wont starts. loadmodule "dispatcher.so" modparam("dispatcher", "db_url", "mysql://kamailioro:kamailioro@localhost /kamailio") modparam("dispatcher", "table_name", "dispatcher") i tried connect to mysql with credentials which you can see in mod_param line and i could query all kamailio database. im really new in this and i dont know how to debug this, is anybody could help me with it ? Best regards, Tomasz

2 1

Re: [SR-Users] Timejump causes kamailio to hang/stall
by erik 16 Feb '17

16 Feb '17

Thanks Daniel, I will look into that. Indeed I found the logic that corrects the time ticks for any drifting, using the time of day. I guess that causes the registrations to expire. I am out of the office for a couple of days, but will get back to you. Best Regards, Erik Hello, On 15/02/2017 14:47, Erik Visser wrote: > After I have kamailio up and running, with UAs properly registrered and sessions routed successfully, my platform faces a timejump. > The platform starts up with date&time set to 1-1-2000 and after kamailio is up and running, the date&time jumps to the current date&time, causing two effects: > - I loose all my registrations (since they expired in 3600 seconds) > - kamailio does not respond to any SIP messages for a couple of minutes > > Is there a way to have kamailio use timeticks rather than relative time for the registrations? > Does anybody recognize the hanging behavior? Any suggestions? > kamailio is using unixtime for registrations, not relative time. But the timer callbacks are using relative intervals to know when to be executed (e.g., timer callbacks are defined to be executed like very 1 minute, or every 90 seconds). It may be that some of the timers try to catch up on missed intervals. It is strange that you say no sip traffic is processed. Can you simulate again the situation and get the backtrace with gdb from few sip workers? If your traffic is over udp, then do 'kamctl ps', get the pid of processes with index 1, 2 and 3, then use gdb to attach to those PIDs and grab the output of: bt full info locals list Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Mar 6-8 (Europe) and Mar 20-22 (USA) - www.asipto.com Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 0

remove offline message prefix of Offline message of MSILO module
by Venkatesh Macha 16 Feb '17

16 Feb '17

Hi all, I am implementing Offline messages functionality using the MSILO module. Everything went well. I am able to store and send those offline messages when the user comes online. But my present problem is, I don't want the offline message prefix. Please look at the following example. This is what Kamailio sending. [Offline message - Wed Feb 15 05:43:38 2017] Test2 I want only Test2 Is there any possibility to remove that offline message prefix ? Thank you in advance ! -- View this message in context: http://sip-router.1086192.n5.nabble.com/remove-offline-message-prefix-of-Of… Sent from the Users mailing list archive at Nabble.com.

2 2

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users