- sr-users - kamailio.org

Using kamailio to play early media announcements
by Adnan 28 Aug '13

28 Aug '13

Hi, I know that this is not the purpose of the proxy but there is a situation in which our kamailio needs to playback early media for a small number of calls. Which module will be most suitable for this? One way is to use rtpproxy and just functions such asrtpproxy_stream2... to play back the prompts to the caller. Note that the caller is not behind NAT or behind any symmetric firewall. So we a free to choose. We know we could do it easily by dispatching to asterisk where we use the Progress() app or similar for freeswitch but we don't want add the overhead of a new media server installation and configuration. We need a kamailio based one box solution.

3 3

Can I use kamailio replace of asterisk?
by 黑了，白了！ 28 Aug '13

28 Aug '13

Hello, I have a question about the load balancer module of kamailio. As the site http://kb.asipto.com/ say, Kamailio is as a SIP proxy router to scale Asterisk. Can I run a kamailio instance as the load balancer, and other several instances as voice service replace of Asterisk? If I can do that, could you give me a tutoral? We are using kamailio as our server. Thank you very much. ------------------ Best Regards!

2 1

Documentation for ims_registrar_pcscf
by Sai Krishna Kota 28 Aug '13

28 Aug '13

Hello, Can any one please provide the documentation for integrating PCSCF with Kamailio. The link for the documentation for ims_registrar_pcscf is not found. ( http://kamailio.org/docs/modules/devel/modules/ims_registrar_pcscf.html) Please tell me what are the modules are to be loaded before pcscf. Thank you, Kota Sai Krishna

2 1

Re: [SR-Users] hanging active dialog...
by Daniel-Constantin Mierla 28 Aug '13

28 Aug '13

The example is the raw command to send, but kamctl takes care to format it. You should do: kamctl fifo profile_list_dlg callers or kamctl fifo profile_list_dlg callers _URI_ Replace the _URI_ with the value that was in $fu. Cheers, Daniel On 8/23/13 3:34 PM, Gertjan Wolzak wrote: > > I have tried with kamctl... > > But because I found the documentation not very clear, I think I am > doing it wrong. > > Let's take the 8.6: > > :profile_list_dlgs:_reply_fifo_file_ > > inbound_calls > > _empty_line_ > > Where the parameters are profile and value, whereby the value is optional. > > I tried : kamctl fifo :profile_list_dlgs: > _/reply_fifo_file/_inbound_calls callers > > Im pretty sure that is not correct. > > Been looking for the correct why on google, but can find a clear > explanation. > > Gertjan. > > *From:*sr-users-bounces@lists.sip-router.org > [mailto:sr-users-bounces@lists.sip-router.org] *On Behalf Of > *Daniel-Constantin Mierla > *Sent:* vrijdag 23 augustus 2013 15:06 > *To:* Kamailio (SER) - Users Mailing List > *Subject:* Re: [SR-Users] hanging active dialog... > > Have you listed the profile via kamctl or kamcmd? The commands are in > the readme: > > http://kamailio.org/docs/modules/stable/modules/dialog.html#idp3737432 > > Is the afferent profile listing any dialog? > > Cheers, > Daniel > > On 8/23/13 2:50 PM, Gertjan Wolzak wrote: > > Hello Carlos, > > I have been looking at your module, can't wait to work with it, > especially the prepaid part. > > But as this system is in production, can't just start > experimenting, want to.... But cant. > > So just have to wait for some tips on the cli.... > > But thanks. > > Rgds, > > Gertjan > > *From:*sr-users-bounces@lists.sip-router.org > <mailto:sr-users-bounces@lists.sip-router.org> > [mailto:sr-users-bounces@lists.sip-router.org] *On Behalf Of > *Carlos Ruiz Díaz > *Sent:* vrijdag 23 augustus 2013 14:37 > *To:* Kamailio (SER) - Users Mailing List > *Subject:* Re: [SR-Users] hanging active dialog... > > I don't know what exactly happened in your case but if you don't > have time to investigate you can try cnxcc module [1] channel > control to achieve the same goal. > > Take a look at the sample configuration file located in [2] > > > > xlog("L_INFO", "Setting up channel based > > credit control")/;/ > > > > > > $var(max_chan) *=* 2; > > > > $var(retcode) = cnxcc_set_max_channels("$var(client)", "$var(max_chan)"); > > > > > > if ($var(retcode) *=*= -1) { > > > > xlog("Error setting up credit control"); > > return; > > > > } > > > > > > $var(count) *=* -1; > > > > > > if (!cnxcc_get_channel_count("$var(client)", "$var(count)")) { > > > > xlog("Error getting customer's channel count" > > )/;/ > > > > } > > > > > > xlog("L_INFO", "CNXCC ROUTE: $var(client) has $var(count) call(s)")/;/ > > > > > > if ($var(retcode) < -1) { > > > > xlog("Too many channels for custome > > r")/;/ > > > > sl_send_reply(403, "Forbidden")/;/ > > > > > > if (!cnxcc_terminate_all("$var(client)")) { > > > > xlog("Error terminating customer's calls")/;/ > > > > } > > > > > > exit/;/ > > } > > > > > > [1]http://kamailio.org/docs/modules/devel/modules/cnxcc.html > > [2]https://github.com/caruizdiaz/cnxcc/blob/master/example/kamailio-cnxcc.cfg > > > > Regards, > > Carlos > > > > On Fri, Aug 23, 2013 at 5:16 AM, Gertjan Wolzak > <g.wolzak(a)foize.com <mailto:g.wolzak@foize.com>> wrote: > > Goodmorning All, > > I use the following route to check for concurrent calls by the > same user, if a concurrent call is tried it is not allowed. > > route[CONCURRENT] > > { > > xlog("SCRIPT: Conccurrent call check"); > > if(!get_profile_size("caller","$fu","$avp(nrcalls)")) > > { > > sl_send_reply("403", "Call not matching profile"); > > exit; > > } > > xlog("SCRIPT: caller value for $fu is $avp(nrcalls)"); > > if($avp(nrcalls)>= 1) > > { > > sl_send_reply("403", "Active calls limit exceeded"); > > exit; > > } > > dlg_manage(); > > if(!set_dlg_profile("caller","$fu")) > > { > > sl_send_reply("500", "No new channels in this profile"); > > exit; > > } > > xlog("SCRIPT: caller value for $fu is now $avp(nrcalls)"); > > } > > Now I had a situation where a user could not call because the > get_profile_size for this user gave the value 1. So another call > was not allowed. > > But the user did not have a call active. As no dialogs were > active, checked that. > > I assume there can be a lot of reasons why this happens, also I > want to use the "I don't want to know the cause" method to solve this. > > So I looked at the dialog module documentation, to see if I can > just clear the profile size for this specific user. > > I wanted to use the profile_list_dlgs(8.6) to get the dialog > details, then the dlg_terminate_dlg(8.4) to, you guessed it, > terminate that dialog. > > But it's not clear to me how I can give those commands.... > > Is there a way to give those commands on the cli, if so can > someone please write down some examples... > > Would be really appreciated. > > Rgds, > > Gertjan > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing > list > sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > > -- > Carlos > > http://caruizdiaz.com > > +595981146623 > > > > > _______________________________________________ > > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > > sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> > > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > > -- > Daniel-Constantin Mierla -http://www.asipto.com > http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> -http://www.linkedin.com/in/miconda -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

2 1

ims_registrar_pcscf
by Sai Krishna Kota 28 Aug '13

28 Aug '13

Hello, Can any one please provide the documentation for integrating PCSCF with Kamailio. The link for the documentation for ims_registrar_pcscf is not found. ( http://kamailio.org/docs/modules/devel/modules/ims_registrar_pcscf.html) Please tell me what are the modules are to be loaded before pcscf. Thank you, Kota Sai Krishna

1 0

cleanup database
by Sebastian Grosse 27 Aug '13

27 Aug '13

Hallo All, I use the module sipcapture and this will fill a MySQL-database. Is there a better / an other way to cleanup the database frequently as to create an own cronjob? I want to set a maximum interval of time or table-rows, because I don't want my database to grow eternal. Any idea how to configure kamailio, sipcapture or an other module or other ideas? Sebastian G.

2 3

Presence related tables are not populated
by Rupayan Dutta 27 Aug '13

27 Aug '13

Dear All Users, While logging into kamailio server via SIP over websocket using my sipml5 client (PRESENCE & XCAP enabled along with WEBSOCKET) the watcher,xcap and presentity tables are not populated.Therefore I can't get presence status though I can successfully running my video conferencing along multiple VC clients.I can't find any problem in my kamailio configuration file.I am attaching the kamailio.cfg file.can you please help where is the problem?eagerly waiting for your suggestion. Thanks in advance

1 0

dispatcher, get attrs on incoming request from nodes
by I.Pavlov 27 Aug '13

27 Aug '13

Hi all. We use kamailio v.3.3.3 with dispatcher and some nodes: #pstn gw group 1 sip:10.189.19.1:5060 0 2 weight=80;limit=120,node=sv6 # SV6 1 sip:10.189.19.29:5060 0 1 weight=20;limit=120;node=mr64 # MR64 I use ds_is_from_list: if (ds_is_from_list("1")) { route(FROM_PSTN); }; route[FROM_PSTN] { xlog("L_INFO","dsattrs: $avp(dsattrs) $avp(i:271) $avp(i:272) $avp(i:273)"); :.. }; Result: INFO: <script>: dsattrs: <null> <null> <null> <null> My question - Is it possible to get attrs on incoming request from GW ?

3 5

Re: [SR-Users] Web socket Feature
by Rupayan Dutta 27 Aug '13

27 Aug '13

Please have a look at https://gist.github.com/jesusprubio/4066845. At the same time you can download kamailio source and in the example directory of the source folder you will find websocket.cfg file. Hope this will help you On Mon, Aug 26, 2013 at 7:02 PM, Gertjan Wolzak <g.wolzak(a)foize.com> wrote: > Hello Vijay, > > What kind of answer do you expect? > > Rupayan has attached an example Kamailio configuration, you just have to > edit it to comply with your situation. > > Or do you have other questions? > > Please then ask the questions, elaborate on what you have tried and what > are > the problems you run into. > > Then give the guru's some time to answer, you sound a bit pushy when you > mention that you do not get an answer on your question you have posted just > three hours earlier. > > Ok, when you have configured the Kamailio based on the config that Rupayan > has attached, BTW you can also find an example in the source of Kamailio, > then use http://sipml5.org/ as the web sip client. Click the demo link, > enter the details for your server and see what happens. > > Should you run in to challenges, tell us what you have done, what the log > files tell you and wait to see if someone has run into the same challenge > and knows how to solve it. > > Good luck. > > > > -----Original Message----- > From: sr-users-bounces(a)lists.sip-router.org > [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of Vijay Thakur > Sent: maandag 26 augustus 2013 14:46 > To: sr >> "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" > Subject: [SR-Users] Web socket Feature > > Dear All Members, > > > I am not getting the answer of my queries from anywhere. So sorry to post > again. > I have installed Kamailio 4.0 in Ubuntu 12.04 and i am able to test chat, > voip calls, screen sharing with Jitsi client. This is a test server. I > have > to test web socket feature of Kamailio 4.0. I have got no material from > google or any forum about web socket. My motive is that i want to make a > call from web browser. For this purpose what changes are required in > kamailio.cfg and what web application can be used as web voip client. > > > Thanks in advance. > > Vijay Thakur > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >

1 0

What does this mean...
by arun Jayaprakash 26 Aug '13

26 Aug '13

This is a very basic question, but can someone let me know what this line means? if($si==$sel(cfg_get.pstn.gw_ip)) return 1; return -1; Thank you, A

2 4

Config debugging
by Marc Soda 26 Aug '13

26 Aug '13

(Sorry for the double post) I trying to debug why Kamailio is dropping certain ACKs. I have debug level set to 4 and cfgtrace on. The last thing I see in the log for my missing ACKs is: Aug 26 12:22:29 eng-reg1 kamailio: ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=789 a=16 n=if Aug 26 12:22:29 eng-reg1 kamailio: ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=786 a=25 n=check_route_param Aug 26 12:22:29 eng-reg1 kamailio: DEBUG: rr [loose.c:974]: check_route_param(): params are <;alias=172.16.3.65~5060~1> Aug 26 12:22:29 eng-reg1 kamailio: ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=795 a=16 n=if Which corresponds to these lines in the config: if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) { return; } if (is_request()) { <---------LINE 795 if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } Why would that be the last line that is traced? Is Kamailio really not processing past here? -- Marc

2 1

[OT] Video processing of Kamailio Word recordings
by Daniel-Constantin Mierla 26 Aug '13

26 Aug '13

Hello, I have few GB with the recordings of presentations from Kamailio World, but they are in full HD format and need to be processed for uploading to one of the web sites hosting videos. Not being familiar with video processing, I am asking here to see if anyone can give some hits about what I could use to prepare the videos for upload. I would need to do: - merge some video files - cut from video files - eventually add few frames at the beginning with title of presentation/name of presenter - resize/transcode to right format for web publishing Let me know if you know some good/easy to use toolkit for such operations (free or at low cost), running on Linux or Mac OS X. Also, if anyone volunteers to help and has already an adequate infrastructure for video editing, maybe it is easier to transfer the raw content via snail mail on a usb stick (unless you are in Berlin area where we can meet in person). Another point would be suggestions about what web sites to use for publishing the videos. Youtube is one of them, but maybe there are better alternatives. If that matters, the average time of presentations should be 25-30min. Thanks, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Trainings - Berlin, Oct 21-24; Miami, Nov 11-13, 2013 - more details about Kamailio trainings at http://www.asipto.com -

4 3

(no subject)
by Marc Soda 26 Aug '13

26 Aug '13

I trying to debug why Kamailio is dropping certain ACKs. I have debug level set to 4 and cfgtrace on. The last thing I see in the log for my missing ACKs is: Aug 26 12:22:29 eng-reg1 kamailio: ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=789 a=16 n=if Aug 26 12:22:29 eng-reg1 kamailio: ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=786 a=25 n=check_route_param Aug 26 12:22:29 eng-reg1 kamailio: DEBUG: rr [loose.c:974]: check_route_param(): params are <;alias=172.16.3.65~5060~1> Aug 26 12:22:29 eng-reg1 kamailio: ERROR: *** cfgtrace: c=[/etc/kamailio/kamailio.cfg] l=795 a=16 n=if Which corresponds to these lines in the config: if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) { return; } if (is_request()) { <---------LINE 795 if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } Why would that be the last line that is traced? Is Kamailio really not processing past here? -- Marc

1 0

Web socket Feature
by Vijay Thakur 26 Aug '13

26 Aug '13

Dear All Members, I am not getting the answer of my queries from anywhere. So sorry to post again. I have installed Kamailio 4.0 in Ubuntu 12.04 and i am able to test chat, voip calls, screen sharing with Jitsi client. This is a test server. I have to test web socket feature of Kamailio 4.0. I have got no material from google or any forum about web socket. My motive is that i want to make a call from web browser. For this purpose what changes are required in kamailio.cfg and what web application can be used as web voip client. Thanks in advance. Vijay Thakur

2 1

Websocket Feature Test
by Vijay Thakur 26 Aug '13

26 Aug '13

Dear All, Recently, I have install Kamailio 4.0. In this new release i want to test websocket feature in available in this version. How i can i make calls from Web browser. Is there any one in the list, share his experience. Thanks in advance. Vijay Thakur

2 1

Carrierroute
by Lucky Santiago 26 Aug '13

26 Aug '13

I have kamailio 4.0.2 installed on Ubuntu 12.04.1 and I already have user/subscriber registered and PSTN routing already working fine (outgoing call) i just need to add config on kamailio.cfg (modparam, ifdefs, routing logic, etc) to make this carrierroute module work based on the db entries i created, if someone can provide sample config entries that needs to add it would be great, since i believe this is not included on the default installation of kamailio. Sorry for disturbing you. kamctl cr show cr carrier names +----+------------+ | id | carrier | +----+------------+ | 1 | 61.8.XX.XX | +----+------------+ cr domain names +----+------------+ | id | domain | +----+------------+ | 1 | 61.8.XX.XX | +----+------------+ cr routes +----+---------+--------+-------------+-------+------+------+-------+--------------+----------------+----------------+------------------+ | id | carrier | domain | scan_prefix | flags | mask | prob | strip | rewrite_host | rewrite_prefix | rewrite_suffix | description | +----+---------+--------+-------------+-------+------+------+-------+--------------+----------------+----------------+------------------+ | 1 | 1 | 1 | 03 | 0 | 0 | 1 | 0 | 61.8.XX.XX | 0 | 0 | CR to voip gw | +----+---------+--------+-------------+-------+------+------+-------+--------------+----------------+----------------+------------------+ My current kamailio.cfg was based on this, http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob;f=etc/kama… with just few modification for PSTN route usage (IP addr of PSTN and allowed prefix) Thanks in Advance, -- Lucky Santiago M: +63.932.88.LUCKY (58259) Skype: luckysantiago

1 0

Finding Destination IP:Port for 200 OK
by SamyGo 26 Aug '13

26 Aug '13

Dear list, Its just a little thing to ask but I somehow can't seem to figure out which psuedo variable to use to find out where a particular 200 OK is destined to go. the scenario is I've multiple media servers in dispatcher and calls are distributed. I need to capture a 200OK that is destined to go to a particular ip! I've tried using the dst_ip, $dd, $du but none of them are giving any results for the 200 OK. I just need to know where do kamailio is going to send this 200 since it is already a part of an on-going INVITE/call. Thanks, Sammy

2 4

Websockets problem in Kamailio
by Ruben Antunes 24 Aug '13

24 Aug '13

Hello, I've tried to configure *Websockets *server in *Kamailio *(i've installed kamailio with these steps: http://www.kamailio.org/wiki/install/4.0.x/git), but I think that is *not working*, because when trying to connect to the *Websockets* server in *JsSip * http://tryit.jssip.net/ says: <http://gyazo.com/f37d9f42caf1915223611f6f1ad18b84.png> That's the *Websockets server*: ws://184.75.243.217:10080 * * * * The *configuration file* that I use is that: https://gist.github.com/jesusprubio/4066845 I've changed the basic things.. directory of the modules, port of the wss:// and the mysql user and password. That's a printscreen of the *Kamailio* server running: (there are more that just 1 IP because my server have 4 IP address because of ICE Server. The* Kamailio is running* pretty well over programs like *Jitsi*, but not in the browser because of Websockets, that's why I want to run a Websockets Server. <http://gyazo.com/ea6635a8162513c150e35fe2f8e0e381.png>

1 0

Change "from user "
by arun Jayaprakash 24 Aug '13

24 Aug '13

Hello, can someone let me know how I can modify the "from user" when making an outbound PSTN call. For example, when I make a PSTN call now the caller ID shows up as my 4 digit extension but I would like my PSTN number displayed instead. Thank you. Arun

2 2

UAC module fails to refresh registrations
by neumann 24 Aug '13

24 Aug '13

First register is OK, but after 60 seconds time to update register, kamailio after 401 Unauthorized dont send second register with Authorization. In log i see only ERROR: uac [uac_reg.c:537]: uac_reg_tm_callback(): authentication failed for <kamailio> mysql> select * from uacreg; +----+----------+------------+----------------+------------+----------------+----------+---------------+---------------+--------------------+---------+ | id | l_uuid | l_username | l_domain | r_username | r_domain | realm | auth_username | auth_password | auth_proxy | expires | +----+----------+------------+----------------+------------+----------------+----------+---------------+---------------+--------------------+---------+ | 1 | kamailio | aster | 192.168.16.201 | kamailio | 192.168.16.200 | asterisk | kamailio | 1234 | sip:192.168.16.200 | 60 | +----+----------+------------+----------------+------------+----------------+----------+---------------+---------------+--------------------+---------+ 1 row in set (0.00 sec) #!KAMAILIO # # Kamailio (OpenSER) SIP Server v4.0 - default configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: <sr-users(a)lists.sip-router.org> # # Refer to the Core CookBook at http://www.kamailio.org/wiki/ # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: #-!define WITH_DEBUG # # *** To enable mysql: #!define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql #!define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication #!define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql #!define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: #!define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Include Local Config If Exists ######### import_file "kamailio-local.cfg" ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!endif #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ #auto_aliases=no /* add local domain aliases */ #alias="sip.mydomain.com" /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:10.0.0.10:5060 /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 #!ifdef WITH_TLS enable_tls=yes #!endif # life time of TCP connection when there is no traffic # - a bit higher than registration expires to cope with UA behind NAT tcp_connection_lifetime=3605 ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" pstn.gw_port = "" desc "PSTN GW Port" #!endif #!ifdef WITH_VOICEMAIL # VoiceMail Routing on offline, busy or no answer # # - by default Voicemail server IP is empty to avoid misrouting voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules_k:modules" #!else mpath="/usr/local/lib/kamailio/modules_k/:/usr/lib64/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "mi_rpc.so" loadmodule "acc.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif loadmodule "uac.so" # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 1) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # set it to 1 to enable GRUU modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) #!endif modparam("uac", "reg_db_url", "mysql://kamailio:kamailiorw@localhost/kamailio") modparam("uac", "reg_timer_interval", 60) modparam("uac", "reg_db_table", "uacreg") modparam("uac", "reg_contact_addr", "192.168.16.201:5060") ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); } route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); exit; } } # USER location service route[LOCATION] { #!ifdef WITH_SPEEDDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") if(sd_lookup("speed_dial")) route(SIPOUT); #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) route(SIPOUT); #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } route(RELAY); exit; } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { add_contact_alias(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage("co"); if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { add_contact_alias(); } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } if (strempty($sel(cfg_get.pstn.gw_port))) { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); } else { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # route to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n"); return; } if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); route(RELAY); exit; #!endif return; } # manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); } # manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { $du = $null; route(TOVOICEMAIL); exit; } #!endif }

1 0

Re: [SR-Users] 477 Unfortunately error on sending to next hop occurred
by Daniel-Constantin Mierla 23 Aug '13

23 Aug '13

I'll check the logs and most probably have to do some test myself, but it's going to take a bit. Daniel On 8/23/13 2:50 PM, Steve Davies wrote: > > > > On 23 August 2013 14:24, Daniel-Constantin Mierla <miconda(a)gmail.com > <mailto:miconda@gmail.com>> wrote: > > Oh, I forgot that the group of t_set_* functions have one > parameter, to allow set/unset the flag. Use: > > t_set_disable_internal_reply(1); > > > Hi Daniel, > > Doesn't seem to work... > > > Kamailio sends back 500, 477, 477, 477 > > Return code from t_relay was still -1: > > t_set_disable_internal_reply(1); > $avp(senttoast) = 0; > $var(rr) = t_relay(); > xlog("L_NOTICE","SLD: in RELAY, t_relay returned $var(rr) > err.rcode is $err.rcode t_r_c is $T_reply_code sent = $avp(senttoast)\n"); > if ($var(rr) < 0) { > sl_reply_error(); > } > > > >> Aug 23 14:34:11 ubuntu /usr/local/sbin/kamailio[22785]: NOTICE: > <script>: SLD: in RELAY, t_relay returned -1 err.rcode is <null> t_r_c > is 100 sent = 0 > > > Thanks, > Steve > > > -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

2 4

Kamailio Presence module
by Kethzer Docteur 23 Aug '13

23 Aug '13

I'm using kamailio 4.1 Im having problem to see witch user is online, I have the module presence.so and presence_xml.so load into the config file of Kamailio.cfg. #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) modparam("presence", "presentity_table", "presentity") modparam("presence", "active_watchers_table", "active_watchers") modparam("presence", "watchers_table", "watchers") modparam("presence", "db_update_period", 20) modparam("presence", "clean_period", 50) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif # handle presence related requests route(PRESENCE); Any help please? -- Kethzer Docteur

1 0

477 Unfortunately error on sending to next hop occurred
by Steve Davies 23 Aug '13

23 Aug '13

Hi, I see that if 4.0.3 has a network error in sending a request, it sends back a 477 response, but does not execute the t_on_failure route block. Which I want to use to route the call via a local alternative gateway. I saw a bug that talked about the failure_exec_mode modparam, but that's not listed in the 4.0.3 documentation. Is there a way to get this to work? Thanks, Steve -- Steve Davies: Technical Director, Connection Telecom (Pty) Ltd Email is preferred, but: Phone: 0878200200

4 12

Error message installing kamailio websocket
by Kethzer Docteur 23 Aug '13

23 Aug '13

CC (gcc) [M websocket.so] ws_frame.o In file included from ws_frame.c:25: /usr/include/unistr.h:189: error: expected ‘;’, ‘,’ or ‘)’ before ‘_UNUSED_PARAMETER_’ /usr/include/unistr.h:259: error: expected ‘;’, ‘,’ or ‘)’ before ‘_UNUSED_PARAMETER_’ make[1]: *** [ws_frame.o] Error 1 make: *** [modules] Error 1 -- Kethzer Docteur

2 5

Re: [SR-Users] hanging active dialog...
by Daniel-Constantin Mierla 23 Aug '13

23 Aug '13

Have you listed the profile via kamctl or kamcmd? The commands are in the readme: http://kamailio.org/docs/modules/stable/modules/dialog.html#idp3737432 Is the afferent profile listing any dialog? Cheers, Daniel On 8/23/13 2:50 PM, Gertjan Wolzak wrote: > > Hello Carlos, > > I have been looking at your module, can't wait to work with it, > especially the prepaid part. > > But as this system is in production, can't just start experimenting, > want to.... But cant. > > So just have to wait for some tips on the cli.... > > But thanks. > > Rgds, > > Gertjan > > *From:*sr-users-bounces@lists.sip-router.org > [mailto:sr-users-bounces@lists.sip-router.org] *On Behalf Of *Carlos > Ruiz Díaz > *Sent:* vrijdag 23 augustus 2013 14:37 > *To:* Kamailio (SER) - Users Mailing List > *Subject:* Re: [SR-Users] hanging active dialog... > > I don't know what exactly happened in your case but if you don't have > time to investigate you can try cnxcc module [1] channel control to > achieve the same goal. > > Take a look at the sample configuration file located in [2] > > > xlog("L_INFO", "Setting up channel based credit control")/;/ > > > $var(max_chan) *=* 2; > > $var(retcode) = cnxcc_set_max_channels("$var(client)", "$var(max_chan)"); > > > if ($var(retcode) *=*= -1) { > > xlog("Error setting up credit control"); > return; > > } > > > $var(count) *=* -1; > > > if (!cnxcc_get_channel_count("$var(client)", "$var(count)")) { > > xlog("Error getting customer's channel count")/;/ > > } > > > xlog("L_INFO", "CNXCC ROUTE: $var(client) has $var(count) call(s)")/;/ > > > if ($var(retcode) < -1) { > > xlog("Too many channels for customer")/;/ > > sl_send_reply(403, "Forbidden")/;/ > > > if (!cnxcc_terminate_all("$var(client)")) { > > xlog("Error terminating customer's calls")/;/ > > } > > > exit/;/ > } > > > [1]http://kamailio.org/docs/modules/devel/modules/cnxcc.html > [2]https://github.com/caruizdiaz/cnxcc/blob/master/example/kamailio-cnxcc.cfg > > Regards, > Carlos > > > On Fri, Aug 23, 2013 at 5:16 AM, Gertjan Wolzak <g.wolzak(a)foize.com > <mailto:g.wolzak@foize.com>> wrote: > > Goodmorning All, > > I use the following route to check for concurrent calls by the same > user, if a concurrent call is tried it is not allowed. > > route[CONCURRENT] > > { > > xlog("SCRIPT: Conccurrent call check"); > > if(!get_profile_size("caller","$fu","$avp(nrcalls)")) > > { > > sl_send_reply("403", "Call not matching profile"); > > exit; > > } > > xlog("SCRIPT: caller value for $fu is $avp(nrcalls)"); > > if($avp(nrcalls)>= 1) > > { > > sl_send_reply("403", "Active calls limit exceeded"); > > exit; > > } > > dlg_manage(); > > if(!set_dlg_profile("caller","$fu")) > > { > > sl_send_reply("500", "No new channels in this profile"); > > exit; > > } > > xlog("SCRIPT: caller value for $fu is now $avp(nrcalls)"); > > } > > Now I had a situation where a user could not call because the > get_profile_size for this user gave the value 1. So another call was > not allowed. > > But the user did not have a call active. As no dialogs were active, > checked that. > > I assume there can be a lot of reasons why this happens, also I want > to use the "I don't want to know the cause" method to solve this. > > So I looked at the dialog module documentation, to see if I can just > clear the profile size for this specific user. > > I wanted to use the profile_list_dlgs(8.6) to get the dialog details, > then the dlg_terminate_dlg(8.4) to, you guessed it, terminate that dialog. > > But it's not clear to me how I can give those commands.... > > Is there a way to give those commands on the cli, if so can someone > please write down some examples... > > Would be really appreciated. > > Rgds, > > Gertjan > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > > -- > Carlos > > http://caruizdiaz.com > > +595981146623 > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

2 1

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