- sr-users - kamailio.org

sip code and sip reason read only
by Kelvin Chua 14 Mar '14

14 Mar '14

How can i alter at least the sip reason in transit? $rr is read only for example: what i wanted to do is when an egress switch returned, 404 i am ugly i wanted to return this to ingress 404 he is ugly Kelvin Chua

1 1

bug in textops module
by Juha Heinanen 14 Mar '14

14 Mar '14

according to textops readme: 2.1. Kamailio Modules The following modules must be loaded before this module: * sl -- Stateless replies. however, i get these errors if pv module is not loaded before textops module: Mar 14 10:14:52 wheezy1 sip-proxy[4707]: ERROR: <core> [pvapi.c:790]: pv_parse_spec2(): error searching pvar "avp" Mar 14 10:14:52 wheezy1 sip-proxy[4707]: ERROR: <core> [pvapi.c:994]: pv_parse_spec2(): wrong char [s/115] in [$avp(s:rpid)] at [5 (5)] Mar 14 10:14:52 wheezy1 sip-proxy[4707]: ERROR: siputils [rpid.c:73]: init_rpid_avp(): malformed or non AVP $avp(s:rpid) AVP definition Mar 14 10:14:52 wheezy1 sip-proxy[4707]: ERROR: siputils [siputils.c:235]: mod_init(): failed to init rpid AVP name Mar 14 10:14:52 wheezy1 sip-proxy[4707]: ERROR: <core> [sr_module.c:970]: init_mod(): init_mod(): Error while initializing module siputils (/usr/lib/sip-proxy/modules/siputils.so) please fix textops module so that it does not depend on pv module if rpid_avp is not set. -- juha

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SIREMIS redirecting to login prompt issue
by malik sherif 14 Mar '14

14 Mar '14

I configured apache web server and also installed siremis 4.1.1 successfully but i couldn't login and my problem is after " welcome to Siremis installation wizard" I went through steps 1to 4 without issue. On the last step I even receive installation complete congratulation also display username and login password but after 10 seconds it goes back to step 1. I was expecting to see a login window but I am stack. Did anybody experience this issue? Any help greatly appreciated. Thanks Abdulmalik

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Kamailio World 2014 - Interactive sessions
by Daniel-Constantin Mierla 14 Mar '14

14 Mar '14

This year we scheduled more interactive sessions, to reveal better how Kamailio and open source RTC applications can be used out there. During the afternoon of April 2, a group of technical tutorials target to teach how to use various platforms for typical use cases: sip:provider CE, Kamailio for IMS and prepaid systems, SEMS. At the end of April 3, we have the open debate moderated by Randy, the host of weekly VoIP Users Conference. Before noon on April 4, James Body challenges everyone around to do a dangerous demo, impress the audience and win some goodies -- prepare yours in time! More details are presented in the schedule: - http://conference.kamailio.com/k02/schedule/ About two weeks and a half till the start of the event, there is no other meeting alternative in Europe for networking with professionals in large real time communications deployments and open source telephony, be sure you don't miss it. Registration details are available at: - http://conference.kamailio.com/k02/registration/ Credits to event sponsors that ensured a good location and excellent content by participation of world wide speakers - Asipto, FhG Fokus, Sipwise, IT Center, Sipgate, Simwood, Digium, NG Voice, Core Network Dynamics: - http://conference.kamailio.com/k02/sponsors/ See you very soon in Berlin! Meanwhile work on your idea for a dangerous demo! -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference - April 2-4, 2014, Berlin, Germany http://www.kamailioworld.com

1 0

401 after a 302
by Marc Soda 14 Mar '14

14 Mar '14

Can someone tell me how to handle a 401 from a 302 redirect? I am attempting to register with the uac module. Normally, I set a failure route for the 401 and call auc_auth(). In this case, I am receiving a 302, sending the register with get_redirects(), then getting a 401. However, Kamailio seems to be ignoring the 401. I see it in an onrely_route, but not a failure route. failure_route[REG_AUTH] { xlog("L_NOTICE","route[REG_AUTH]\n"); # Handle redirects if ($T_reply_code == 302) { get_redirects("*", "Redirected"); t_on_failure("REG_AUTH"); route(RELAY); } else if ($T_reply_code == 401) { uac_auth(); t_on_reply("REG_REPLY"); route(RELAY); } } route[RELAY] { xlog("L_NOTICE","route[RELAY] ($rm)\n"); if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if (!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if (!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if (!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } xlog("L_NOTICE","t_relay()'ing ($rm)\n"); if (!t_relay()) { sl_reply_error(); } exit; } Thanks! Marc

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siremis doesn't works with postgresql
by Rene Montilva 14 Mar '14

14 Mar '14

Hi List i want to config siremis with postgresql and really to much things fail for the install, at the end y left the siremis db with mysql and kamailio with postgresql but the problems continue, when i want to show subscriber list from siremis interface, i get an error and check the postgresql log and found this LOG: sentencia: SET NAMES 'utf8' ERROR: error de sintaxis en o cerca de «`» en carácter 101 SENTENCIA: SELECT T0.id, T0.username, T0.domain, T0.password, T0.email_address, T0.ha1, T0.ha1b, T0.rpid FROM `subscriber` T0 LIMIT 10 what could be the right config?

2 1

Inter domain routing
by Owais ul Haq 14 Mar '14

14 Mar '14

Hello, I have two kamailio servers running independently with different domain names. I want to route calls between those two servers. Can it be done ? If yes then How ? Thanks.

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Soft phone Issue
by Owais ul Haq 14 Mar '14

14 Mar '14

Hello, I have configured a kamailio server on my local network, gave it a dummy domain like "abcd.tk". I have a number of computers on my local network. I want to run eyebeam on them and connect to my kamailio server. Server is running fine. When I try to connect Eyebeam to Server, it goes to DNS to find IP for "abcd.tk". I have entered the value "192.168.1.10 = abcd.tk" in "hosts" file of all machines. File is located at "C:\WINDOWS\system32\drivers\etc\hosts". On other hand, when I ping abcd.tk from client computers. They ping 192.168.1.10. So Operating systems verify the entry "192.168.1.10 = abcd.tk", but problem lies with soft phones. I have tried X-Lite and Eyebeam. How can I make softphone understand that abcd.tk = 192.168.1.10 and please go to this adress, not DNS ?? Help please :( Regards, Awais.

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Softphone not connecting
by Owais ul Haq 14 Mar '14

14 Mar '14

Hello, I have deployed a server. It is starting fine. Connection with MySQL database is also okay. Siremis is also working fine. But softphone "Eyebeam" can't login as a user to the server. Client functionality is ok. As same client with same setting is working good with other kamailio server. How can I trace down the error ?? Please guide. Regards, Owais

2 1

Two kamailio servers sharing database
by Owais ul Haq 14 Mar '14

14 Mar '14

Hello, I have deployed two kamailio servers. Both are sharing a common database deployed on Windows 2008 Server using MySQL. Now domain of one kamailio server is lets say "abc.com" and other's is "xyz.com". I want to know that if a user is created by user id "user1(a)abc.com" with password "qazwsx@123" in the database. And a user try to login from 2nd server "xyz.com" using user id "user1(a)xyz.com" with password "qazwsx@123". Will he able to login using this shared database ? Also please tell which modules controls this ? As I have few more questions also concerning this.

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help with SIP Notify message, to make a endpoint (SPA504G and similar) to reboot.
by Pedro Niño 13 Mar '14

13 Mar '14

Sorry for intruding, but this is puzzling me a lot. I followed the guide to make Kamailio work with asterisk and realtime, Kamailio version 4 and asterisk 11. (http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb ) up until now its working, and now want to get the phones (CISCO SPA504G, former linksys) to get the SIP notify command and reload to get their new config, from the asterisk cli the cli command is "sip notify spa-reboot <username>" The problem is that, using the setup in the guide, the phones registers with kamailio, kamalio then strip the password and register with no password to the asterisk, as supposed to be. by that, when asterisk sends the SIP Notify with the reboot commands, it arrives to the kamailio, and is forwarded to phone, and the phone answers with a '401' and a digest. In the flow, the kamailio (in the middle) receives the 401 and sends again the sip message to asterisk, and asterisk uses the digest to cipher the password and reply, but as asterisk is configured with no password, the reply contains as 'wrong' password (the empty/blank one) is there any way to make kamailio intercepts again that '401' and reply with a crafted message, with the right digest, password, (the one in the ul table) and user, so the phone can accept the order and reboots? The other (ugly) option, is to remove the auth from the phone, for the Sip Provisioning, but that would leave and open door to a reboot attack without auth needed from any IP. And I dont like that option. As we talk, I managed to create a routing logic, but is that the right approach? : =================== request_route { if (is_method("NOTIFY")) { #xlog("NOTIFY received: $rs"); t_on_failure("MANAGE_FAILURE_401"); #try to intercep a 401 } ..... } failure_route[MANAGE_FAILURE_401] { if (t_check_status("401")) { #401 detectado dentro de un Notify, #probablemente ha sido por una orden de reboot. #hay que construir un paquete e intentar enviar, con el digest adecuado.... $uac_req(method)="NOTIFY"; $uac_req(ruri)="sip:"; $uac_req(furi)="sip:"; $uac_req(turi)="sip:"; $uac_req(hdrs)="Contact: <sip:"+ ">\r\n"; #Paquete construido, debe ser enviado uac_req_send(); xlog("401 detectado"); # exit; } =================== Thanks in advance.....

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Problem install siremis and kamailio with postgresql
by Rene Montilva 13 Mar '14

13 Mar '14

Hi list i tried install siremis with postgresql db, but when i was in step 2 and select postgresql driver and fill all the required fields, but i saw siremis installer try to connect to postgresql with www-data user, why this happend

1 0

Bizarre dialog profile tracking behaviour
by Alex Balashov 13 Mar '14

13 Mar '14

Hello, We've got an issue we've been trying to track for days with 'dialog', where the dialogs go through the following life cycle: 1. INVITE --> 2. <-- 407 challenge 3. ACK --> 4. INVITE --> 5. <-- 100 Trying 6. <-- 180 Ringing 7. <-- 183 Session Progress 8. <-- 200 OK 9. ACK --> 10. BYE --> 11. <-- 200 OK Somewhere around steps #6-7 (18x messages), the dialogs no longer appear to be counted in the dialog profile they're attached to, as gleaned from 'kamctl fifo profile_get_size <profile> <key>'. The dialog count in the profile falls to 0. Naturally, I suspected a SIP issue that was preventing the dialog state from being tracked correctly. However, subsequent investigation revealed that these very same dialogs are tracked just fine throughout their entire lifetime. The 'kamctl fifo dlg_list' command shows them to have the right states at the right times. So, the dialog itself isn't going away from tracking. Its attachment to the given profile seems to be the issue. I can't find any logical explanation for this in the code, nor any technical reasoning to support this hypothesis. However, that's what seems to be happening. I'm not sure that this is the issue; it might be another issue masquerading as having this effect. I'm just trying to rule possibilities out. Are there any situations that can cause a dialog that is otherwise being normally tracked to be dumped from a profile of which it was previously part? Thanks! -- Alex -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

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Problemas with kamctl ul show - Kamailio 4.1
by Raphael Borges 13 Mar '14

13 Mar '14

Hello, I'm having a problem with the command kamctl ul show, and I do not understand what is happening. On my server when I run just "kamctl ul show" command works fine but when I run "kamctl ul show [<username>]" an error happens. Before when I was using Kamailio 3.1, its work ok. I looked for some documentation and found nothing about this. For example: *$ /home2/local/kamailio/sbin/kamctl ul show* Domain:: location table=512 records=1 max_slot=1 AOR:: 2501 Contact:: sip:2501@192.168.XXX.XXX:5060;transport=udp Q= Expires:: 3594 ... Domain:: aliases table=512 records=0 max_slot=0 *$ /home2/local/kamailio/sbin/kamctl ul show 2501* 404 AOR not found Thanks for Help. Best regards. Raphael

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Reminder: about private emails
by Daniel-Constantin Mierla 12 Mar '14

12 Mar '14

A message sent almost one year ago, but based on volume of private emails related to free support on SIP, Kamailio or other applications, perhaps it is time to refresh, with couple of extra bits for clarification: - http://lists.sip-router.org/pipermail/sr-users/2013-April/077700.html >>> I would like to remind that, at least in my case, writing emails directly without cc-ing the mailing list is 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 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. There are several common sense as well as technical reasons. 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, nobody here owe you anything. Therefore, if anyone sent me a private message asking about kamailio or continuing a discussion started on mailing lists, then the messages have to be resent on mailing list if one wants an answer. A quick look in my unfiltered folder revealed hundreds of messages during the past year, soon many will be deleted anyhow. Note that you can always mask sensitive data (e.g., replace passwords, IP addresses) before sending logs or config file. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference - April 2-4, 2014, Berlin, Germany http://www.kamailioworld.com

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trixbox or asterisk as a media server for kamailio
by malik sherif 12 Mar '14

12 Mar '14

I am looking documentation as to how to integrate kamailio and Asterisk, I am not sure if I use the correct term "media server" but I would like to configure 3-way call, call waiting, call transfer , and call forwarding. Your help is greatly appreciated. Thanks Abdul

2 2

Kamailio LCR unable to relay new calls, memory issue
by Camila Troncoso 12 Mar '14

12 Mar '14

Hi, Last Friday I presented a increase of load in kamailio by a client. The load was almost twice as usual. By that time I encountered with many issues in kamailio. The calls were not relay to gateways, and a lot of messages about memory were appearing in log file. Kamailio didn't crash. When I remove the extra traffic all went back to normal. Is there a issue about memory leak or about to few memory resources to manage this amount of traffic? ERROR: tm [t_reply.c:1799]: ERROR: relay_reply: cannot alloc reply shmem ERROR: <core> [msg_translator.c:1782]: ERROR: build_req_buf_from_sip_req: out of memory ERROR: tm [t_fwd.c:431]: ERROR: print_uac_request: no shm mem ERROR: tm [t_fwd.c:1534]: ERROR: t_forward_nonack: failure to add branches ERROR: tm [tm.c:1369]: ERROR: w_t_relay_to: t_relay_to failed ERROR: tm [t_reply.c:598]: ERROR: _reply_light: cannot allocate shmem buffer ERROR: tm [t_msgbuilder.c:371]: ERROR: cannot allocate shared memory ERROR: tm [t_msgbuilder.c:521]: ERROR: build_local_reparse: cannot build ACK request MERROR: tm [t_msgbuilder.c:371]: ERROR: cannot allocate shared memory ERROR: tm [t_msgbuilder.c:521]: ERROR: build_local_reparse: cannot build ACK request ERROR: sl [sl_funcs.c:282]: ERROR: sl_reply_error used: No error (2/SL) ERROR: <core> [msg_translator.c:1782]: ERROR: build_req_buf_from_sip_req: out of memory ERROR: tm [t_fwd.c:431]: ERROR: print_uac_request: no shm mem ERROR: tm [t_fwd.c:1534]: ERROR: t_forward_nonack: failure to add branches ERROR: tm [tm.c:1369]: ERROR: w_t_relay_to: t_relay_to failed ERROR: sl [sl_funcs.c:282]: ERROR: sl_reply_error used: No error (2/SL) ERROR: <core> [sip_msg_clone.c:506]: ERROR: sip_msg_cloner: cannot allocate memory ERROR: tm [t_lookup.c:1338]: ERROR: new_t: out of mem: ERROR: tm [t_lookup.c:1478]: ERROR: t_newtran: new_t failed ERROR: sl [sl_funcs.c:282]: ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) Regards, Camila *Camila Troncoso Solar **|* Ingeniero de Desarollo +56 2 22408535 *·* +56 9 97327220 | ctroncoso(a)redvoiss.net Badajoz 130, piso 16, Las Condes | Santiago - CHILE | *www.redvoiss.com <http://www.redvoiss.com>*

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Re: [SR-Users] Routing based on destination and source
by Keith 12 Mar '14

12 Mar '14

Hi Yeah I do already use the lcr module for routing to external carriers but is probably a little overkill for what I need to route that traffic internally. Is that my only real option? Cheers Keith

1 0

Kamailio Proxy for Freeswitch
by jay binks 12 Mar '14

12 Mar '14

So im using Kamailio as a load balancer and proxy in-front of a few Freeswitch boxes. Im not going to use RTP Proxy, as my FS boxes will be public on the internet ( Direct RTP connection is fine ). I want ALL SIP to go via Kamailio, but the issue im running into is that the SIP Contact header has the FS IP address in it in some cases. ( 180's etc ) Every bit of documentation I can find about re-writing the Contact header is geared around NAT, and thats not what im trying to do here.. Can anyone give some pointers on how to basically topology hide the FS boxes on the SIP front but leave their IP in the SDP. im going to have the firewalls setup so the SIP Can ONLY come from kamailio, so its quite important I get this all right. Any pointers would be appreciated. Thanks Jay

2 3

(no subject)
by malik sherif 12 Mar '14

12 Mar '14

I send couple of email to Asipto.com about Siremis login issue but I didn't get any response. I configured apache web server and also installed siremis 4.1.1 successfully but i couldn't login and my problem is after " welcome to Siremis installation wizard" I went through step 1to 4 without issue on the last step I even receive installation complete congratulation also display username and login password but after 10 seconds it goes back to step 1. I was expecting to see a login window but I stack. Did anybody experience this issue? or if there is a mailing list, can someone point to the right group. Any help greatly appreciated. Thanks Abdulmalik

2 1

Re: [SR-Users] Routing based on destination and source
by Keith 12 Mar '14

12 Mar '14

Hi all, I am looking for the best module to route calls based on the authenticated user that call came from and the destination number. Is there any current module to do this? Many thanks Keith

2 1

Fail Safe Environment for kamailio
by Owais ul Haq 12 Mar '14

12 Mar '14

Hello, Guys, I need some help. I am tasked to deploy "Fail Safe Architecture" for already running Kamailio Server in our organization. Presently one Kamailio Server is serving the cause with one internet connection and database on MySQL running on Windows 2008 Server R2(which is attached to kamailio server on other LAN card). Now the requirement is to deploy redundancy for internet connection and kamailio server. Like they want two kamailio servers running parallel, so if one fails, other will automatically take over and service will be available to users without any obstruction. At any cost, my organization want to absolutely minimize any possible blackout. To further elaborate my point. Here is the network we want. 2 incoming internet connections to 2 kamailio servers. Both attached to a single database placed at the backend. Now is it possible ?? If not what are constraints ?? What workaround is possible ?? I seriously need help as it is matter of job. Thanking in advance for help. P.S : kamailio server is running of Fedora machine in our environment. Cheers.

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Web interface for kamailio
by malik sherif 11 Mar '14

11 Mar '14

Any recommendation web interface for gathering statistics other than SIREMIS? I installed SIREMIS but unable to resolve login problem and my request for help on ASIPTO didn't get any response also unable to find siremis mailing list. Your help is greatly appreciated. Thanks Abdulmailk

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kamailio 4.1.X and ast 11.8
by lore 11 Mar '14

11 Mar '14

hello folks, i'm really new with kamailio (a couple fo days) and I need some suggestion/samples/idea/HELP to realize my environment. I've been starting to read the docs on website but there are a lot and i don't know where start and which kind of modules need to use I''d like to realize a trunk from kamailio and asterisk. Each one has their own sip users. kamailio sip users are some inside network and some outside network (behind nat) Could someone be son kind to show me the right way to configure kamailio for incoming/ougoing calls from/to asterisk and for using sip peers (behind nat and not)?. Also same samples are really preciated. Thanks a lot in advance Kind regards Lorenzo -- "Chi vive sperando muore cagando ... Lo Russo isoletta dell'Egeo che non conta un cazzo, 1941 ... sono anche un autore"

1 0

milliseconds on $Ts
by Kelvin Chua 11 Mar '14

11 Mar '14

Hi guys, Is there a code somewhere in the project i can reuse to implement milliseconds on $Ts? for example introduce a "$Tms" or a "$TmS" to refer to a pv that stores milliseconds instead of seconds? Kelvin Chua

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how to get value of route param?
by Juha Heinanen 10 Mar '14

10 Mar '14

rr module has check_route_param(re) function that can be used to check if local route header has a param that matches re. however, there does not exist a function to find out what is the value of a given rr param. for example, if route header has param ;param=foo, how to extract value foo of param? there is @rr.uri.params select that together with {param.value,name} transformation might be able to do the job? -- juha

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How to encode domain in username for Asterisk REGISTER forwarding (was: How to configure Kamailio + Asterisk (on same server) to route between several disjoint networks?)
by Alex Villacís Lasso 10 Mar '14

10 Mar '14

On the previous issue (disjoint networks), I eventually settled for binding Asterisk on localhost:5080, and using rtpproxy to route media to the networks as required. The only snag is that I had to write a script to rewrite the Kamailio configuration in order to take current IPs into account. Now, for the same project. What I want now is to be able to express multiple domains with an Asterisk backend. That is: given pbx.company1.com and pbx.company2.com, two DNS addresses that point to the same server, I want to have separate accounts bob(a)pbx.company1.com and bob(a)pbx.company2.com, with separate registration and accounting, without collisions. I know about the Kamailio "domain" module, and I have enabled WITH_MULTIDOMAIN and filled the kamailio.domain table for it to work. Now the authentication is taking domains into account. Next, I want to express two non-colliding registrations for Asterisk. What comes to mind is username rewrite - Kamailio sees bob(a)company1.com, and forwards a REGISTER to Asterisk using bob_company1_com as the SIP username. I am looking right now at the WITH_ASTERISK block that uses the uac module: # Forward REGISTER to Asterisk route[REGFWD] { if(!is_method("REGISTER")) { return; } $var(rip) = $sel(cfg_get.asterisk.bindip); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.asterisk.bindport); $uac_req(furi)="sip:" + $au + "@" + $var(rip) + ":5060"; $uac_req(turi)="sip:" + $au + "@" + $var(rip) + ":5060"; $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n"; if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n"; uac_req_send(); } #!endif The above is the non-encoding version. What is the best way to modify it to do a domain encoding in the username? Or, if a better solution exists, what is it?

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Debian init script "log_faiulre_msg"
by Corey Edwards 08 Mar '14

08 Mar '14

There's a typo in the Debian init script. Is this the correct place to report packaging bugs? --- /etc/init.d/kamailio 2014-03-06 13:42:23.000000000 -0700 +++ /tmp/kamailio 2014-03-07 08:41:05.000000000 -0700 @@ -52,7 +52,7 @@ log_failure_msg "Not starting $DESC: invalid configuration file!" log_failure_msg log_failure_msg "$out" - log_faiulre_msg + log_failure_msg exit 1 fi } Corey

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Billing solutions for Kamailio?
by Sergey Kharlamov 07 Mar '14

07 Mar '14

Hi, Can someone recommend me good billing solution for kamailio? Ideally it should contain admin panel, user cabinet etc... -- Sergey Kharlamov

2 1

p-cscf - out of mem
by Daniel Ciprus 07 Mar '14

07 Mar '14

Hi, Has anybody seen this : Mar 2 17:25:16 kamailio kam-pcscf[20741]: ERROR: <core> [msg_translator.c:2011]: generate_res_buf_from_sip_res(): out of mem Mar 2 17:25:16 kamailio kam-pcscf[20741]: ERROR: tm [t_reply.c:1943]: relay_reply(): ERROR: relay_reply: no mem for outbound reply buffer followed by: Mar 2 17:27:03 kamailio kam-pcscf[20741]: ERROR: <core> [msg_translator.c:2011]: generate_res_buf_from_sip_res(): out of mem Mar 2 17:27:03 kamailio kam-pcscf[20741]: ERROR: tm [t_reply.c:1943]: relay_reply(): ERROR: relay_reply: no mem for outbound reply buffer Mar 2 17:27:03 kamailio kam-pcscf[20741]: ERROR: <core> [msg_translator.c:2168]: build_res_buf_from_sip_req(): ERROR: build_res_buf_from_sip_req: out of memory ; needs 386 Mar 2 17:27:53 kamailio kam-pcscf[20741]: ERROR: <core> [parser/contact/contact.c:194]: parse_contacts(): parse_contacts(): No memory left Mar 2 17:27:53 kamailio kam-pcscf[20741]: ERROR: <core> [parser/contact/parse_contact.c:59]: contact_parser(): contact_parser(): Error while parsing contacts Mar 2 17:27:53 kamailio kam-pcscf[20741]: ERROR: <core> [parser/contact/parse_contact.c:88]: parse_contact(): parse_contact(): Error while parsing However, seems like we still had enough mem: [root@kamailio kamailio]# vmstat 1 procs -----------memory---------- ---swap-- -----io---- --system-- -----cpu----- r b swpd free buff cache si so bi bo in cs us sy id wa st 3 0 0 9526348 230308 13282472 0 0 0 4 0 1 6 0 94 0 0 1 0 0 9526216 230308 13282524 0 0 0 0 1313 84424 17 8 75 0 0 1 0 0 9526092 230308 13282616 0 0 0 0 1482 86882 16 9 75 0 0 [root@kamailio monit.d]# kamailio -v version: kamailio 4.2.0-dev0 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 17:42:42 Dec 18 2013 with gcc 4.4.7 Unfortunately I had to reload p-cscf to get service working again. -- Daniel Ciprus Integration engineer http://www.acision.com 9954 Mayland Dr Suite 3100 Richmond, VA 23233 USA T: +1 804 762 5601 E: daniel.ciprus(a)acision.com<mailto:daniel.ciprus@acision.com> ________________________________ This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you for understanding.

2 6

(no subject)
by malik sherif 07 Mar '14

07 Mar '14

Hello, I am using kamailio 4.1.1, I am wondering if kamailio supports call forwarding , three-way call, and call transfer natively? Does it expect the endpoints to do the RTP mixing? Thanks Absul

3 3

500 Forking capacity exceeded (12/SL)
by Camila Troncoso 07 Mar '14

07 Mar '14

Hi, I´m having problems with kamailio as LCR . I have a route with 14 different gateways, each one with priority 1 and all with the same weight. If one gateway is full, busy, down or answers with any failed response to LCR , I jump to the next gateway in the list. I´m presenting the problem that its stops jumping after 12 gateways and returns the message 500 Forking capacity exceeded (12/SL). I need to increase this limit , so all the gateways in the list give the call a try. Regards, Camila *Camila Troncoso Solar **|* Ingeniero de Desarollo +56 2 22408535 *·* +56 9 97327220 | ctroncoso(a)redvoiss.net Badajoz 130, piso 16, Las Condes | Santiago - CHILE | *www.redvoiss.com <http://www.redvoiss.com>*

3 5

RV: 500 I'm terribly sorry, server error occurred (1/SL)
by Camila Troncoso 07 Mar '14

07 Mar '14

*De:* Camila Troncoso [mailto:ctroncoso@redvoiss.net] *Enviado el:* jueves, 06 de marzo de 2014 16:03 *Para:* sr-users(a)lists.sip-router.org *Asunto:* 500 I'm terribly sorry, server error occurred (1/SL) Hi, I have kamailio 3.2 as LCR with many gateways, weights and priorities. Out of no ware It presented this error today: 500 I'm terribly sorry, server error occurred (1/SL). This wasn't happening for all the calls , and was arbitrary. I read on google ,that this error comes form t_relay, so the server was unable to relay the call to the gateway. But all the gateways are reachable and active. The server was on load , almost 30 cps. The same configuration have been running for more than 2 years with any problem. Could these happened for heavy load? I have children=32 in config file. Has anyone an idea of what may happened? Regards, Camila Troncoso *Camila Troncoso Solar **|* Ingeniero de Desarollo +56 2 22408535 *·* +56 9 97327220 | ctroncoso(a)redvoiss.net Badajoz 130, piso 16, Las Condes | Santiago - CHILE | *www.redvoiss.com <http://www.redvoiss.com>*

1 0

KAMAILIO 4.1.1 Forward SIP REGISTER
by Francesco Maria Magnini 07 Mar '14

07 Mar '14

Hi, I would like to have a Kamailio Registrar/Location server on my private network and another one facing the public side: KAM1 (Private Registrar) <—> KAM2 (Public Proxy) I would like also to configure the proxy in this way: - REGISTRATION TO Public Proxy KAM2 going to Private Registrar KAM1 In the reference guides I’ve seen almost always application specific configuration (WITH_ASTERISK, WITH_FREESWITCH). Basically, how it is possible to forwards REGISTER messages to a generic upstream Registrar? Which modules are involved? I’m using Kamailio 4.1.1 Regards, Francesco

3 4

RPM packages for v4.1.2, v4.0.6 and v3.3.7
by Daniel-Constantin Mierla 07 Mar '14

07 Mar '14

Hello, based on the releases made yesterday, the rpm packages for various Linux distributions are available via opensuse build services - the list of distributions and links to the repositories are published at: - http://www.kamailio.org/wiki/packages/rpms Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference - April 2-4, 2014, Berlin, Germany http://www.kamailioworld.com

1 0

Newbie looking for resources
by richard.seguin＠marisec.ca 07 Mar '14

07 Mar '14

Hi folks, I'm new to Kamailio, and I am looking for some documentation on how to write config files, what should be included... etc.. I'm pretty much looking for product documentation and tutorials, not info on SIP (already went through the sip school). - How do I write a basic config file - Basic auth and forward examples I'm familiar with the cookbooks, however they don't really provide context as to what the recommendations do. Any resources that people could provide would be greatly appreciated. Thanks, Richard Seguin

3 2

500 I'm terribly sorry, server error occurred (1/SL)
by Camila Troncoso 06 Mar '14

06 Mar '14

Hi, I have kamailio 3.2 as LCR with many gateways, weights and priorities. Out of no ware It presented this error today: 500 I'm terribly sorry, server error occurred (1/SL). This wasn't happening for all the calls , and was arbitrary. I read on google ,that this error comes form t_relay, so the server was unable to relay the call to the gateway. But all the gateways are reachable and active. The server was on load , almost 30 cps. The same configuration have been running for more than 2 years with any problem. Could these happened for heavy load? I have children=32 in config file. Has anyone an idea of what may happened? Regards, Camila Troncoso *Camila Troncoso Solar **|* Ingeniero de Desarollo +56 2 22408535 *·* +56 9 97327220 | ctroncoso(a)redvoiss.net Badajoz 130, piso 16, Las Condes | Santiago - CHILE | *www.redvoiss.com <http://www.redvoiss.com>*

1 0

Kamailio v4.1.2 Released
by Daniel-Constantin Mierla 06 Mar '14

06 Mar '14

Hello, Kamailio SIP Server v4.1.2 stable release is out. This is a maintenance release of the latest stable branch, 4.1, that includes fixes since release of v4.1.1. There is no change to database schema or configuration language structure that you have to do on installations of v4.1.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.1.2. For more details about version 4.1.2 (including links and guidelines to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2014/03/kamailio-v4-1-2-released/ RPM, Debian/Ubuntu packages will be available soon as well. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Kamailio v4.0.6 Released
by Daniel-Constantin Mierla 06 Mar '14

06 Mar '14

Hello, Kamailio SIP Server v4.0.6 stable release is out. This is a maintenance release of the stable branch 4.0, that includes fixes since release of v4.0.5. There is no change to database schema or configuration language structure that you have to do on installations of v4.0.x. Deployments running previous v4.0.x versions are strongly recommended to be upgraded to v4.0.6. For more details about version 4.0.6 (including links and guidelines to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2014/03/kamailio-v4-0-6-released/ Note that 4.0.x is previous stable release series, the latest one is now 4.1.x, with v4.1.2 released today as well. In addition, 3.3.7 was packaged in order to have it included in the new APT repository and make transition to newer versions easier for those still on branch 3.3. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Incorrect IP address in path header
by Rob Moore 06 Mar '14

06 Mar '14

Hi All, So an interesting one here. I'm doing pass through registrations using the dispatcher module in Kamailio and I'm using the Path Module to set the Path header In Register request so that Freeswitch knows to follow the route via Kamailio to find the sipPhone in future. In summary: [SIP Phone]------[NAT]------[(externalip)Kamailio(internalip)]-------[FS-Registrar] The problem I'm having is that the Path header is being added with the external ip address of the Kamailio instance not the internal ip e.g. "Path: sip:(externalip);transport=tcp" which mean FS doesn't root new calls out to the SIPPhone correctly via kamailios internal ip. I have noticed that if I remove the ;transport=tcp from the FS-Register address in my dispatcher group so that the last hop (from kamailio to FS-Register) is done over UDP then the problem corrects itself. (although this does present new issues whilst calling out back along the path from FS, and we also have a prerequisite to use TCP so this doesn't help). Has anyone had any experience of this issue? I can't see what I've done wrong here. Many Thanks Rob

2 2

sqlops crash after out of memory
by Juha Heinanen 06 Mar '14

06 Mar '14

pkg memory run out here: Mar 4 09:00:20 localhost /usr/sbin/sip-proxy[31923]: ERROR: sqlops [sql_api.c:318]: sql_do_query(): no more memory then sip proxy crashed here: (gdb) where #0 0xb5fce89b in sql_reset_result (res=0xb6ce3438) at sql_api.c:233 #1 0xb5fd18db in sql_do_query (con=0xb6cd1e28, query=0xbfecd248, res=0xb6ce3438) at sql_api.c:410 #2 0xb5fd630d in sql_query (msg=0xb6ce8788, dbl=0xb6cd1e28 "\230\035Í¶\t", query=0xb70436a8 "\310Y\004\267P", res=0xb6ce3438 "\245\246=|\r3Î¶\006") at sqlops.c:209 ... goto error was executed when if(res->vals[i]==NULL) was true and at error sql_reset_result was called. there the crash happened here: if(res->vals) { for(i=0; i<res->nrows; i++) { for(j=0; j<res->ncols; j++) { if(res->vals[i][j].flags&PV_VAL_STR && res->vals[i][j].value.s.len>0) /* here */ looks to me that it should be tested if res->vals[i] is not null before executing the if statement. -- juha ps. it there any chance to find out from the core dump, why memory run out, i.e., how was most of it used? amount of pkg memory in this host was defined to be 8 MB. -- juha

2 2

dialog module with DB Backend
by jay binks 06 Mar '14

06 Mar '14

Hi all, im using the dialog module with db_cassandra backend.. I dont believe this issue is related to cassandra, but its worth mentioning anyways. so... I run kamailio, make calls, see dialogs in the DB.. and I Can use "kamctl mi dlg_list" and see that dialogs go away when I hangup a call.. When I query the DB Backend, I still see the queries, but they have a state of 5. I Initially thought this was a bug, but it seems dialogs in state 5 get cleaned up after a period. so I moved on. now , lets restart kamailio.. kamailio loads all dialogs on startup, after kamailio starts I call "kamctl mi dlg_list" again, and it shows all my dialogs from the DB. they DO show as "State 5" but for some reason, these dialogs appear to stick around for a long time, and the bigger issue it causes me is that my channel limiting ( using get_profile_size ) seems to consider these dialogs ( in state 5 ) as being active calls. Please someone point me in the right direction... :) what am I doing wrong ? ( or is this a bug somewhere ) Sincerely Jay

3 12

geoip crash
by Juha Heinanen 06 Mar '14

06 Mar '14

i made a typo in my script and $gip(src=>cc) was referenced before geoip_match() call. the reference resulted in crash. to reproduce, execute statement $var(test) = $gip(src=>cc); as first thing in your script. -- juha

2 17

Releasing 4.1.2, 4.0.6 and 3.3.7
by Daniel-Constantin Mierla 06 Mar '14

06 Mar '14

Hello, we are considering to release new versions from stable branches of Kamailio starting with Thursday, March 6. First is v4.1.2 from latest stable branch and if the time allows in the same day, the v4.0.6. The reason to release v3.3.7 is to have a proper debian packages built and hosted on the new repositories -- the specs had a conflict for latest 3.3 version of some sub-packages. If there are known issues that were not addressed, please report them to the tracker. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference - April 2-4, 2014, in Berlin, Germany http://www.kamailioworld.com

3 4

How to add some text at end of body line with subst_body
by Barthel Marco (CI/AFU1) 05 Mar '14

05 Mar '14

Hi, I'm trying to add some string at the end of some line in the body/sdp of a SIP INVITE for testing some interoperability. The following line a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:NwC8z5MsyCWbpJJo1n2BDS8DkGuUU2cIt9KRUqPU should be converted to a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:NwC8z5MsyCWbpJJo1n2BDS8DkGuUU2cIt9KRUqPU UNENCRYPTED_SRTCP I tried with this: subst_body('/(a=crypto.*)\r/\1 UNENCRYPTED_SRTCP\r/g'); but it does not match just using subst_body('/(a=crypto.*)/\1 UNENCRYPTED_SRTCP/g'); matches but results in this (carriage return between key and "UNENCRYPTED_SRTCP"): a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:NwC8z5MsyCWbpJJo1n2BDS8DkGuUU2cIt9KRUqPU\rUNENCRYPTED_SRTCP using Kamailio 3.3.6 Any ideas? Mit freundlichen Grüßen / Best regards Marco Barthel Robert Bosch GmbH (CI/AFU1) Postfach 30 02 20 70442 Stuttgart GERMANY www.bosch.com<http://www.bosch.com> Sitz: Stuttgart, Registergericht: Amtsgericht Stuttgart, HRB 14000; Aufsichtsratsvorsitzender: Franz Fehrenbach; Geschäftsführung: Dr. Volkmar Denner, Dr. Stefan Asenkerschbaumer, Dr. Rolf Bulander, Dr. Stefan Hartung, Dr. Dirk Hoheisel, Christoph Kübel, Uwe Raschke, Wolf-Henning Scheider, Dr. Werner Struth, Peter Tyroller

2 1

msilo config
by Peter Villeneuve 05 Mar '14

05 Mar '14

Hi, Can someone share a working msilo config for 4.1.1? I've tried following older tutorials out there but they no longer seem to apply. I just want to test out msilo with the default kamailio config in 4.1.1, but the current example in http://kamailio.org/docs/modules/4.1.x/modules/msilo.html doesn't work for me. Specifically, I get an error regarding the syntax in t_on_failure("1"); Any help or pointers appreciated. Thanks

2 5

Incorrect Contact address
by Marc Soda 05 Mar '14

05 Mar '14

I have Kamailio setup as a proxy in front of a backend server (Asterisk). When I make a call through the proxy, the Contact header in the 200 OK that is returned to the client has the IP of the backend server in it. Thus, the client is sending it's ACK directly to the backend server. Is there a special method to rewrite the Contact header to be Kamailio's IP? Where is a good place in the config to do this? (my config is loosely based on this: http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb) Thanks! Marc

4 5

Re: [SR-Users] [Kamailio-Business] [sr-dev] New official Debian and Ubuntu repository
by Vik Killa 05 Mar '14

05 Mar '14

Can someone update the wiki page? I was confused for days on this: http://www.kamailio.org/dokuwiki/doku.php/packages:debs On Wed, Mar 5, 2014 at 8:21 AM, Vik Killa <vipkilla(a)gmail.com> wrote: > Can someone update the wiki page? I was confused for days on this: > http://www.kamailio.org/dokuwiki/doku.php/packages:debs > > > On Tue, Mar 4, 2014 at 6:54 AM, Klaus Darilion < > klaus.mailinglists(a)pernau.at> wrote: > >> Thanks to all involved >> >> >> On 03.03.2014 14:16, Victor Seva wrote: >> >>> The new build system for Debian and Ubuntu packages is now in place. >>> This service is kindly sponsored by SipWise [0] thanks to Andreas >>> Granig [1]. Sipwise is providing the hosting and man power to create >>> and manage this new system. >>> >>> deb.kamailio.org is based on jenkins-debian-glue[2] project running on >>> AWS EC2 environment thanks to Michael Prokop [3] and myself. All the >>> needed files, scripts and info to reproduce this system is kept public >>> at [4]. >>> >>> In [5] is described all the repositories available. >>> >>> * nightly builds are been built if a change is detected in the branch, >>> once by night. >>> - kamailiodev-nightly >>> branch: >>> - 'master' >>> distributions: jessie, wheezy, squeeze, precise >>> >>> - kamailio41-nightly >>> branch: >>> - '4.1' >>> distributions: jessie, wheezy, squeeze, precise >>> >>> - kamailio40-nightly >>> branch: >>> - '4.0' >>> distributions: lenny, squeeze, wheezy, lucid, precise >>> >>> - kamailio33-nightly >>> branch: >>> - '3.3' >>> distributions: lenny, squeeze, wheezy, lucid, precise >>> >>> * tags are been built if a new tag is detected once by night. >>> - kamailio41 >>> branch: >>> - '*/tags/4.1*' >>> distributions: jessie, wheezy, squeeze, precise >>> >>> - kamailio40 >>> branch: >>> - '*/tags/4.0*' >>> distributions: lenny, squeeze, wheezy, lucid, precise >>> >>> - kamailio33 >>> branch: >>> - '*/tags/3.3*' >>> distributions: lenny, squeeze, wheezy, lucid, precise >>> >>> Now there is no 4.1.1 package built so the default kamailio repository >>> is pointing to a empty repository. As soon as we release 4.1.2 this >>> will be fixed. >>> >>> Regards, >>> Victor Seva ( kamailio admin hat on ) >>> >>> [0] http://www.sipwise.com/ >>> [1] http://www.kamailio.org/w/andreas-granig/ >>> [2] https://github.com/mika/jenkins-debian-glue >>> [3] http://michael-prokop.at/ >>> [4] https://github.com/sipwise/kamailio-deb-jenkins >>> [5] http://deb.kamailio.org/index.html >>> >>> _______________________________________________ >>> sr-dev mailing list >>> sr-dev(a)lists.sip-router.org >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev >>> >>> >> _______________________________________________ >> Kamailio (OpenSER) - Business mailing list >> Business(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/business >> http://lists.openser-project.org/cgi-bin/mailman/listinfo/business >> > >

3 3

Reload memory leaks issues.
by Uri Shacked 05 Mar '14

05 Mar '14

Hi, kamailio ver 3.3.6. I load approximately 100Mb of data into memory. And my shared memory size is 4Gb. It is CARRIERROUTE, DIALPLAN, MTREE and HTABLE combimations. My system has to be synced with DB that is being updated frequently. So, I reload the data every 1-2 minutes. I notice that the shared memory usage is increasing at 1% every 18-19 reloads. I use mem_join = 1 and no mem_join at all. I compiled with C_DEFS+= -DMEM_JOIN_FREE and even with -DF_MALLOC. Tried different options, still, the issue keeps on going and it is very frustrating... Anyone has the same problem? BR, Uri

1 1

MSRP Relay
by Nandini madhu 05 Mar '14

05 Mar '14

Dear All, Can anybody please tel me, are there any MSRP relay (RFC 4976) compliant SIP clients for Android Platform ? Sorry for asking this question which is out of relevant here. Waiting for somebody's comments on this Regards, Nandini

1 0

Kamailio World 2014 - Exhibitors
by Daniel-Constantin Mierla 05 Mar '14

05 Mar '14

During Kamailio World 2014, the companies IT Center, Digium, Sipgate, NG Voice, Asipto and FhG Fokus will provide all day long demos and presentations of their products and services. You can contact them in advance to schedule an appointment if you want to reserve proper time to discuss about their offerings: - http://conference.kamailio.com/k02/exhibitors/ It is less than one month till the start of the event, the agenda is pretty much finalized: - http://conference.kamailio.com/k02/schedule/ It has over 30 sessions presented by world wide speakers, from technical tutorials to business use cases, with topics such WebRTC, IMS, anti-fraud & VoIP security, scalability with OpenStack, real time charging, various Kamailio use cases, Asterisk, FreeSwitch and SEMS. Once again we are grateful to event sponsors -- Asipto, FhG Fokus, Sipwise, IT Center, Sipgate, Simwood, Digium, NG Voice -- without their contributions is impossible to bring such an excellent group of speakers and support event logistics: - http://conference.kamailio.com/k02/sponsors/ Hurry up if you don't want to miss this event, it is no alternative in Europe for meeting professionals in large real time communications deployments and open source telephony. Registration details are available at: - http://conference.kamailio.com/k02/registration/ See you in Berlin! -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

invalid xmlrpc response from pkg.stats?
by Juha Heinanen 05 Mar '14

05 Mar '14

i tried make xmlrpc query on pkg.stats, but my xmlrpc lib was not able to parse the response (below). according to xmlrpc spec: The body of the response is a single XML structure, a <methodResponse>, which can contain a single <params> which contains a single <param> which contains a single <value>. according to the spec <value> is scalar value or <struct> or <array>. in sip proxy response below, there a several <struct>s in the <value>, which according to my understanding is violation of the spec. instead, the response should be like this: <?xml version="1.0"?> <methodResponse> <params> <param> <value> <array> <data> <value> <struct>...</struct> </value> <value> <struct>...</struct> </value> ... </data> </array> </param> </methodResponse> comments? -- juha -------------------------------------------------------------------------- Received the following response: HTTP/1.0 200 OK Via: SIP/2.0/TCP 127.0.0.1:36643 Server: OpenXg SIP Proxy (4.1.1-2 (i386/linux)) Content-Length: 6311 <?xml version="1.0"?> <methodResponse> <params> <param> <value><struct><member><name>entry</name><value><int>0</int></value></member><member><name>pid</name><value><int>5846</int></value></member><member><name>rank</name><value><int>0</int></value></member><member><name>used</name><value><int>4553004</int></value></member><member><name>free</name><value><int>559936</int></value></member><member><name>real_used</name><value><int>5731520</int></value></member></struct> <struct><member><name>entry</name><value><int>1</int></value></member><member><name>pid</name><value><int>5848</int></value></member><member><name>rank</name><value><int>1</int></value></member><member><name>used</name><value><int>4557328</int></value></member><member><name>free</name><value><int>554652</int></value></member><member><name>real_used</name><value><int>5736804</int></value></member></struct> <struct><member><name>entry</name><value><int>2</int></value></member><member><name>pid</name><value><int>5849</int></value></member><member><name>rank</name><value><int>2</int></value></member><member><name>used</name><value><int>4557328</int></value></member><member><name>free</name><value><int>554652</int></value></member><member><name>real_used</name><value><int>5736804</int></value></member></struct> <struct><member><name>entry</name><value><int>3</int></value></member><member><name>pid</name><value><int>5850</int></value></member><member><name>rank</name><value><int>3</int></value></member><member><name>used</name><value><int>4557328</int></value></member><member><name>free</name><value><int>554652</int></value></member><member><name>real_used</name><value><int>5736804</int></value></member></struct> <struct><member><name>entry</name><value><int>4</int></value></member><member><name>pid</name><value><int>5851</int></value></member><member><name>rank</name><value><int>4</int></value></member><member><name>used</name><value><int>4557328</int></value></member><member><name>free</name><value><int>554652</int></value></member><member><name>real_used</name><value><int>5736804</int></value></member></struct> <struct><member><name>entry</name><value><int>5</int></value></member><member><name>pid</name><value><int>5852</int></value></member><member><name>rank</name><value><int>4294967295</int></value></member><member><name>used</name><value><int>4556356</int></value></member><member><name>free</name><value><int>556296</int></value></member><member><name>real_used</name><value><int>5735160</int></value></member></struct> <struct><member><name>entry</name><value><int>6</int></value></member><member><name>pid</name><value><int>5853</int></value></member><member><name>rank</name><value><int>4294967295</int></value></member><member><name>used</name><value><int>4556356</int></value></member><member><name>free</name><value><int>556296</int></value></member><member><name>real_used</name><value><int>5735160</int></value></member></struct> <struct><member><name>entry</name><value><int>7</int></value></member><member><name>pid</name><value><int>5854</int></value></member><member><name>rank</name><value><int>4294967294</int></value></member><member><name>used</name><value><int>4563012</int></value></member><member><name>free</name><value><int>549112</int></value></member><member><name>real_used</name><value><int>5742344</int></value></member></struct> <struct><member><name>entry</name><value><int>8</int></value></member><member><name>pid</name><value><int>5857</int></value></member><member><name>rank</name><value><int>4294967295</int></value></member><member><name>used</name><value><int>4556368</int></value></member><member><name>free</name><value><int>556236</int></value></member><member><name>real_used</name><value><int>5735220</int></value></member></struct> <struct><member><name>entry</name><value><int>9</int></value></member><member><name>pid</name><value><int>5863</int></value></member><member><name>rank</name><value><int>4294967295</int></value></member><member><name>used</name><value><int>4556368</int></value></member><member><name>free</name><value><int>556236</int></value></member><member><name>real_used</name><value><int>5735220</int></value></member></struct> <struct><member><name>entry</name><value><int>10</int></value></member><member><name>pid</name><value><int>5873</int></value></member><member><name>rank</name><value><int>4294967295</int></value></member><member><name>used</name><value><int>4557616</int></value></member><member><name>free</name><value><int>554844</int></value></member><member><name>real_used</name><value><int>5736612</int></value></member></struct> <struct><member><name>entry</name><value><int>11</int></value></member><member><name>pid</name><value><int>5876</int></value></member><member><name>rank</name><value><int>5</int></value></member><member><name>used</name><value><int>4642360</int></value></member><member><name>free</name><value><int>467988</int></value></member><member><name>real_used</name><value><int>5823468</int></value></member></struct> <struct><member><name>entry</name><value><int>12</int></value></member><member><name>pid</name><value><int>5877</int></value></member><member><name>rank</name><value><int>6</int></value></member><member><name>used</name><value><int>4618200</int></value></member><member><name>free</name><value><int>493396</int></value></member><member><name>real_used</name><value><int>5798060</int></value></member></struct> <struct><member><name>entry</name><value><int>13</int></value></member><member><name>pid</name><value><int>5878</int></value></member><member><name>rank</name><value><int>7</int></value></member><member><name>used</name><value><int>4618200</int></value></member><member><name>free</name><value><int>493396</int></value></member><member><name>real_used</name><value><int>5798060</int></value></member></struct> <struct><member><name>entry</name><value><int>14</int></value></member><member><name>pid</name><value><int>5884</int></value></member><member><name>rank</name><value><int>4294967292</int></value></member><member><name>used</name><value><int>4035956</int></value></member><member><name>free</name><value><int>497228</int></value></member><member><name>real_used</name><value><int>5307880</int></value></member></struct> </value> </param> </params> </methodResponse>

2 3

INVITE proxy auth
by Marc Soda 04 Mar '14

04 Mar '14

Hey all, I have a pretty general SIP question that I'm hoping some of you can shed some light on. I hope this ok for the list. I am setting up a SIP proxy with Kamailio. The backend server (Asterisk in my case) requires authentication. Is it standard/best practice to require a proxy to authenticate to a backend server for INVITEs? I am already registering, with authentication, on behalf of the client, i.e, the client registers to the proxy, then the proxy registers to the backend server, all with authentication. I was hoping not to have to do this on INVITEs, but if I don't I'm left with the following: client proxy backend | -------INVITE-------> | | | <--------407--------- | | | ---------ACK--------> | | | ----INVITE(auth)----> | | | <--------100--------- | | | | -------INVITE-------> | | | <--------401--------- | | | ---------ACK--------> | | <--------401--------- | | | ---------ACK--------> | | | ----INVITE(auth)----> | | | <--------100--------- | | | | ----INVITE(auth)----> | | | <--------100--------- | | | <---------OK--------- | | <---------OK--------- | | | ---------ACK--------> | | | | ---------ACK--------> | It works, but, it's terrible... Before I try to make it work differently, what do you all think it should do? Marc

2 1

New official Debian and Ubuntu repository
by Victor Seva 04 Mar '14

04 Mar '14

The new build system for Debian and Ubuntu packages is now in place. This service is kindly sponsored by SipWise [0] thanks to Andreas Granig [1]. Sipwise is providing the hosting and man power to create and manage this new system. deb.kamailio.org is based on jenkins-debian-glue[2] project running on AWS EC2 environment thanks to Michael Prokop [3] and myself. All the needed files, scripts and info to reproduce this system is kept public at [4]. In [5] is described all the repositories available. * nightly builds are been built if a change is detected in the branch, once by night. - kamailiodev-nightly branch: - 'master' distributions: jessie, wheezy, squeeze, precise - kamailio41-nightly branch: - '4.1' distributions: jessie, wheezy, squeeze, precise - kamailio40-nightly branch: - '4.0' distributions: lenny, squeeze, wheezy, lucid, precise - kamailio33-nightly branch: - '3.3' distributions: lenny, squeeze, wheezy, lucid, precise * tags are been built if a new tag is detected once by night. - kamailio41 branch: - '*/tags/4.1*' distributions: jessie, wheezy, squeeze, precise - kamailio40 branch: - '*/tags/4.0*' distributions: lenny, squeeze, wheezy, lucid, precise - kamailio33 branch: - '*/tags/3.3*' distributions: lenny, squeeze, wheezy, lucid, precise Now there is no 4.1.1 package built so the default kamailio repository is pointing to a empty repository. As soon as we release 4.1.2 this will be fixed. Regards, Victor Seva ( kamailio admin hat on ) [0] http://www.sipwise.com/ [1] http://www.kamailio.org/w/andreas-granig/ [2] https://github.com/mika/jenkins-debian-glue [3] http://michael-prokop.at/ [4] https://github.com/sipwise/kamailio-deb-jenkins [5] http://deb.kamailio.org/index.html

4 3

Re: [SR-Users] Regd. Stored procedure support in Kamailio
by Alex Balashov 04 Mar '14

04 Mar '14

You can executed any SQL statements you want through 'sqlops', including invocation of stored procedures. Our Kamailio-based Class 4 routing/trunking product is built almost entirely on Postgres stored procedures. We have dozens of them. As to whether they are "really needed", I don't know how to answer that. :-) In our case, we decided they were really needed because they offer the opportunity to execute complex business logic beyond what Kamailio's relatively primitive route scripting language can offer, while eliminating unnecessary back-and-forth data interchange between Kamailio and the database. Stored procedures and triggers also execute in a transactional manner, which is additionally beneficial. Shankar <shankar.rk(a)plintron.com> wrote: >Hello All, > > > >Does Kamailio support invoking stored procedures instead of sending >plain DB >queries already? > > > >We have been asked to migrate to stored procedures. I am trying to >estimate >the effort involved in migrating Kamailio code to support stored >procedure. > > > >Other question that comes to my mind is whether stored procedures are >really >needed in Kamailio. Can someone please clarify? > > > >Regards, > >Shankar > > > > > >------------------------------------------------------------------------ > >_______________________________________________ >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 -- Sent from my mobile, and thus lacking in the refinement one might expect from a fully fledged keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0671 Web: http://www.evaristesys.com/, http://www.alexbalashov.com

1 0

Regd. Stored procedure support in Kamailio
by Shankar 04 Mar '14

04 Mar '14

Hello All, Does Kamailio support invoking stored procedures instead of sending plain DB queries already? We have been asked to migrate to stored procedures. I am trying to estimate the effort involved in migrating Kamailio code to support stored procedure. Other question that comes to my mind is whether stored procedures are really needed in Kamailio. Can someone please clarify? Regards, Shankar

1 0

REGISTER failure_route 401 problem
by Marc Soda 03 Mar '14

03 Mar '14

I have a Kamailio server setup which is registers to a back end server on behalf of endpoints. The endpoints can register to Kamailio but Kamailio is failing to register to the server when I put a NAT device in front of it. Without the NAT device it works fine. The problem is the 401 that comes back seems to be ignored by Kamailio. I have a failure route setup to auth, but it is never hit. I see the 401 in onrely_route, but not the failure_route. I'm assuming it's a NAT issue because removing the device fixes the issue. Anyone have any ideas? The 401 being ignored: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.10.11;branch=z9hG4bKe5d6.178378f7.0;received=198.XXX.XXX.XXX Via: SIP/2.0/UDP 127.0.0.1:12354 ;rport=6545;received=198.XXX.YYY.YYY;branch=z9hG4bK-1879-1-3 From: <sip:sip7878_spqa@64.YYY.YYY.YYY>;tag=1 To: <sip:sip7878_spqa@64.YYY.YYY.YYY>;tag=as00e32130 Call-ID: 1-1879(a)127.0.0.1 CSeq: 2 REGISTER User-Agent: CoreDialPBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="fe-c7c5-9o.domain.com", nonce="151e4f60" Content-Length: 0 Thanks, Marc

2 6

usrloc preload
by Victor V. Kustov 03 Mar '14

03 Mar '14

Hi all! Please give expanded tips about usrloc preload(). What exactly it doing? modparam("usrloc", "db_mode", 0) modparam("usrloc", "preload", "location") This case kamailio load from db on startup and operate in memory after? -- WBR, Victor JID: coyote(a)bks.tv JID: coyote(a)bryansktel.ru I use FREE operation system: 3.12.9-calculate GNU/Linux

3 5

Ubuntu precise apt repositories unavailable
by Josef Stautner 03 Mar '14

03 Mar '14

Goor morning community, I received a notification that the repositories for Ubuntu Precise are not available anymore. /W: Failed to fetch //http://deb.kamailio.org/kamailio/dists/precise/Release.gpg// Could// //not connect to deb.kamailio.org:80 (87.106.140.63). - connect (110: Connection timed out)// / / //W: Failed to fetch //http://deb.kamailio.org/kamailio/dists/precise/main/source/Sources//// //Unable to connect to deb.kamailio.org:http:// / / //W: Failed to fetch// / /http://deb.kamailio.org/kamailio/dists/precise/main/binary-i386/Packages// Unable to// //connect to deb.kamailio.org:http:// / / //W: Failed to fetch// / /http://deb.kamailio.org/kamailio/dists/precise/main/i18n/Translation-en_US// Unable to// //connect to deb.kamailio.org:http:// / / //W: Failed to fetch// / /http://deb.kamailio.org/kamailio/dists/precise/main/i18n/Translation-en// Unable to// //connect to deb.kamailio.org:http:// / After I tried manually to fetch new updates I got a connection and downloaded some data but then the connection timed out. Is this just a temporary probleme and are there any fallback mirrors? Thanks in advance Josef Stautner

2 2

Adding Path header for the current request.
by MAXX ... 02 Mar '14

02 Mar '14

I found the solution myself, anyone having a similar issue in the future can use msg_apply_changes() from textopsx module after append_hf line.

1 0

Adding Path header for the current request.
by MAXX ... 02 Mar '14

02 Mar '14

Hello, I need to add a Path header to the current request so the save method of registrar module will save the location with the path value to the db. I made the necessary settings for the registrar module but I couldn't be able to add the header. Neither "append_hf" nor "add_path" of path module works since they don't add the header to the currently processing request. $hdr pseudovariable is readonly so I couldn't modify it as well. What can I do to make this happen? Also the reason I need this is to let people registered to different registrars to be able to call each other via their current tls connections to their respective registrars. Thanks In Advance.

1 0

XAVP in 3.1.6
by Jijo 01 Mar '14

01 Mar '14

Hi I'm trying to use XAVP but the script is showing parsing error. Do i need to include any module? I rebuild kamailio using the flag WITH_XAVP but didn' help. route[0] { $xavp(a=>bar) = 'bar'; } S-154:~ # /usr/sbin/kamailio -c -f /etc/kamailio/kam86x2.cfg loading modules under /usr/lib/kamailio/modules_k/:/usr/lib/kamailio/modules/ 0(20184) ERROR: <core> [pvapi.c:516]: error searching pvar "xavp" 0(20184) ERROR: <core> [pvapi.c:720]: wrong char [a/97] in [$xavp(a=>bar)] at [6 (5)] 0(20184) : <core> [cfg.y:3461]: parse error in config file /etc/kamailio/kam86x2.cfg, line 555, column 5-17: unknown script pseudo variable $xavp(a=>bar) ERROR: bad config file (1 errors) Segmentation fault (core dumped) S-154:~ # /usr/sbin/kamailio -V version: kamailio 3.1.6 (i386/linux) af51bc flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, PKG_SIZE 25MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: af51bc compiled on 14:11:43 Feb 28 2014 with gcc 4.7.1 THANKS

1 0

How to configure Kamailio + Asterisk (on same server) to route between several disjoint networks?
by Alex Villacís Lasso 01 Mar '14

01 Mar '14

As part of a project, I have installed a CentOS 6 test system (a virtual machine) with Asterisk 11.7.0 and Kamailio 4.1.1 downloaded from http://download.opensuse.org/repositories/home:/kamailio:/telephony/CentOS_…. I am trying to setup a combination of Kamailio and Asterisk that will route SIP calls between all the configured networks in the test setup, in addition to being capable of using Asterisk in order to handle PSTN and IAX2 calls. I am using the following online guide to modify my kamailio.cfg: http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb . Based on this, I generated the attached patch for my Kamailio configuration My test setup has the following network interfaces: eth0: 10.1.0.3, on network 10.1.0.0/24 eth1: 192.168.5.18, on network 192.168.0.0/16 eth2: 10.0.0.2, on network 10.0.0.0/24 lo: 127.0.0.1, on network 127.0.0.0/8 I first configured Asterisk with SIP realtime support (with no Kamailio), and tested that all configured accounts could register from all interfaces, and that Asterisk could properly route media between any two disjoint networks. After installing Kamailio, the guide called for disabling Asterisk SIP authentication by setting passwords to NULL, and moving Asterisk SIP to a different port (I chose 5080) so that Asterisk and Kamailio could run on the same machine. At this point, the SIP clients (one softphone and one VoIP phone) can now register at port 5080 without authentication. In the process of changing my Kamailio configuration according to the attached patch, the guide says that I should configure the IP of the network interface as the value of asterisk.bindip and kamailio.bindip. After performing all required changes, Kamailio does take over authentication at the default port of 5060. Testing shows that for all SIP clients with IPs belonging to the same network as the configured asterisk.bindip, both registration and media exchange work correctly, and that the SIP clients are still capable of calling into the Asterisk dialplan, and therefore, routing into Asterisk resources. For SIP clients in disjoint networks, the failure mode depends on whether mhomed is enabled or disabled in kamailio.cfg. For mhomed=0 (or unset), I have the following situation between the two SIP clients (one at 10.1.0.1, the other at 10.0.0.3), as shown by "sip show peers" in Asterisk (when asterisk.bindip is set to 192.168.5.18): Privilege escalation protection disabled! See https://wiki.asterisk.org/wiki/x/1gKfAQ for more details. Name/username Host Dyn Forcerport ACL Port Status Description Realtime gatitoscomx64am_100/gatit 10.1.0.3 D N A 5060 OK (16 ms) Cached RT gatitoscomx64am_101/gatit 10.0.0.2 D N A 5060 OK (36 ms) Cached RT gatitoscomx64am_IM101 (Unspecified) D N A 0 UNREACHABLE Cached RT 3 sip peers [Monitored: 2 online, 1 offline Unmonitored: 0 online, 0 offline] If I try to call from one SIP client to an extension in the Asterisk dialplan that does NOT map to a SIP client in a disjoint network, the media exchange works (with negotiatied media IP in the same network as the SIP client), regardless of whether the calling client belongs in the same network as asterisk.bindip. If I try to call from the same SIP client to an extension that maps to a SIP client in a disjoint network, the call fails, and I get the spoken message about the user at extension such-and-such being unavailable. Additionally, I get the following error message in the Asterisk logs: [Feb 25 16:53:14] NOTICE[13807][C-00000003] chan_sip.c: Call from 'gatitoscomx64am_101' (10.0.0.2:5060) to extension 'gatitoscomx64am_101' rejected because extension not found in context 'gatitoscomx64am-from-internal'. For mhomed=1, the output of "sip show peers" changes to the following (when asterisk.bindip is set to 192.168.5.18): Privilege escalation protection disabled! See https://wiki.asterisk.org/wiki/x/1gKfAQ for more details. Name/username Host Dyn Forcerport ACL Port Status Description Realtime gatitoscomx64am_100/gatit 192.168.5.18 D N A 5060 OK (19 ms) Cached RT gatitoscomx64am_101/gatit 192.168.5.18 D N A 5060 OK (34 ms) Cached RT gatitoscomx64am_IM101 (Unspecified) D N A 0 UNREACHABLE Cached RT 3 sip peers [Monitored: 2 online, 1 offline Unmonitored: 0 online, 0 offline] From wireshark sniffing, I can see that the SDP payload sent from the client to Kamailio contains the IP address of the client, which is accessible by both Kamailio and Asterisk. However, the SDP payload in the OK response sent back to the client contains a media port with the IP address of asterisk.bindip (the one that appears in the "Host" column in the "sip show peers" report), not the IP address of the interface that received the INVITE. This results in broken media negotiation for all SIP clients belonging to networks other than the one that contains asterisk.bindip. In either case, I have to hardcode an IP address in kamailio.cfg, which is not satisfactory. IPs assigned to interfaces can and do change, especially if the interface is managed with DHCP. To escape this, I tried setting asterisk.bindip to 127.0.0.1, but since apparently localhost is also a disjoint network, all of the above described problems apply. Related to these issues, I am not satisfied with leaving Asterisk running unauthenticated SIP at the nonstandard port. Somebody suggested blocking the port with iptables, but I do not want to rely on this alone. I tried setting bindaddr=127.0.0.1 so that only Kamailio gets to talk to Asterisk, but this also has the side effect of restricting the media negotiation to localhost only. I am asking for help in building a Kamailio/Asterisk configuration that will support all of the networks and route media between all of them, just as if Asterisk were the only program running. Ideally, the configuration should not encode the current IP of any interface (except, maybe, localhost). What is the official name (if any) for the setup I am describing above? Does it have a standard setup procedure? How is Asterisk secured so that clients cannot bypass authentication using the Asterisk SIP port directly?

2 4

Replacing an ACME Packet Net-Net SBC
by Francesco Maria Magnini 01 Mar '14

01 Mar '14

Hi, I would like to have some suggestions about a full replacement of an ACME Packet Net-Net Session Border Controller. By now, ACME SBC performs all the SBC functionalities, mainly: - it is used as a SIP endpoint for SIP client registrations - it is used as a SIP endpoint for interconnection to multiple SIP carriers via SIP trunks - it is used for NAT traversal In this deployment, the SIP Server communicates only with the SBC and this one takes care of the communication between the SIP Server and the external SIP entities (UA clients, SIP Trunks). In this scenario, can I consider to replace the SBC with Kamailio?

13 22

prevent relaying to myself
by Henry Fernandes 01 Mar '14

01 Mar '14

Is there a way to prevent relaying requests to myself in route[RELAY]? -H

3 3

fresh kamailio installation unable to connect through jitsi
by Michelle Jun 01 Mar '14

01 Mar '14

hi i just finished installation of kamailio on centos 6 64 bit following the docs on http://www.fredposner.com/voip/1457/kamailio-behind-nat/ the user able to chat through jitsi just fine, but when calling, one user able to connect, while the other showing only "connecting" (audio/video) any idea what did i do wrong? here is the /var/log./messages thank you Feb 26 17:43:34 black96 rtpproxy[4293]: INFO:handle_command: new session 2f6e045de764ae9e9252352d952359ae@0:0:0:0:0:0:0:0, tag 2d9e2107;1 requested, type strong Feb 26 17:43:34 black96 rtpproxy[4293]: INFO:handle_command: new session on a port 25982 created, tag 2d9e2107;1 Feb 26 17:43:34 black96 rtpproxy[4293]: INFO:handle_command: pre-filling caller's address with 192.168.168.1:5000 Feb 26 17:43:34 black96 rtpproxy[4293]: INFO:handle_command: new session 2f6e045de764ae9e9252352d952359ae@0:0:0:0:0:0:0:0, tag 2d9e2107;2 requested, type strong Feb 26 17:43:34 black96 rtpproxy[4293]: INFO:handle_command: new session on a port 25192 created, tag 2d9e2107;2 Feb 26 17:43:34 black96 rtpproxy[4293]: INFO:handle_command: pre-filling caller's address with 192.168.168.1:5002 Feb 26 17:43:39 black96 rtpproxy[4293]: INFO:handle_command: lookup on ports 25982/27044, session timer restarted Feb 26 17:43:39 black96 rtpproxy[4293]: INFO:handle_command: pre-filling callee's address with 192.168.168.1:5000 Feb 26 17:43:39 black96 rtpproxy[4293]: INFO:handle_command: lookup on ports 25192/22432, session timer restarted Feb 26 17:43:39 black96 rtpproxy[4293]: INFO:handle_command: pre-filling callee's address with 192.168.168.1:5002 Feb 26 17:43:39 black96 /usr/local/sbin/kamailio[4456]: NOTICE: acc [acc.c:279]: acc_log_request(): ACC: transaction answered: timestamp=1393465419;method=INVITE;from_tag=2d9e2107;to_tag=7175e4ad;call_id=2f6e045de764ae9e9252352d952359ae@0 :0:0:0:0:0:0:0;code=200;reason=OK;src_user=handy;src_domain=184.105.148.231;src_ip=192.168.168.1;dst_ouser=harri;dst_user=harri;dst_domain=192.168.113.2 Feb 26 17:43:39 black96 rtpproxy[4293]: INFO:handle_command: lookup on ports 25982/27044, session timer restarted Feb 26 17:43:39 black96 rtpproxy[4293]: INFO:handle_command: lookup on ports 25192/22432, session timer restarted Feb 26 17:43:40 black96 rtpproxy[4293]: INFO:handle_command: lookup on ports 25982/27044, session timer restarted Feb 26 17:43:40 black96 rtpproxy[4293]: INFO:handle_command: lookup on ports 25192/22432, session timer restarted Feb 26 17:43:42 black96 rtpproxy[4293]: INFO:handle_command: lookup on ports 25982/27044, session timer restarted Feb 26 17:43:42 black96 rtpproxy[4293]: INFO:handle_command: lookup on ports 25192/22432, session timer restarted Feb 26 17:44:43 black96 rtpproxy[4293]: INFO:process_rtp: session timeout Feb 26 17:44:43 black96 rtpproxy[4293]: INFO:remove_session: RTP stats: 0 in from callee, 0 in from caller, 0 relayed, 0 dropped Feb 26 17:44:43 black96 rtpproxy[4293]: INFO:remove_session: RTCP stats: 0 in from callee, 0 in from caller, 0 relayed, 0 dropped Feb 26 17:44:43 black96 rtpproxy[4293]: INFO:remove_session: session on ports 25982/27044 is cleaned up Feb 26 17:44:43 black96 rtpproxy[4293]: INFO:process_rtp: session timeout Feb 26 17:44:43 black96 rtpproxy[4293]: INFO:remove_session: RTP stats: 0 in from callee, 0 in from caller, 0 relayed, 0 dropped Feb 26 17:44:43 black96 rtpproxy[4293]: INFO:remove_session: RTCP stats: 0 in from callee, 0 in from caller, 0 relayed, 0 dropped Feb 26 17:44:43 black96 rtpproxy[4293]: INFO:remove_session: session on ports 25192/22432 is cleaned up Feb 26 17:45:23 black96 rtpproxy[4293]: INFO:handle_command: delete request failed: session 2f6e045de764ae9e9252352d952359ae@0:0:0:0:0:0:0:0, tags 2d9e2107/7175e4ad not found Feb 26 17:45:53 black96 /usr/local/sbin/kamailio[4460]: NOTICE: acc [acc.c:279]: acc_log_request(): ACC: transaction answered: timestamp=1393465553;method=BYE;from_tag=2d9e2107;to_tag=7175e4ad;call_id=2f6e045de764ae9e9252352d952359ae@0:0:0:0:0:0:0:0;code=408;reason=Request Timeout;src_user=handy;src_domain=184.105.148.231;src_ip=192.168.168.1;dst_ouser=harri;dst_user=;dst_domain=184.105.148.231

3 7

What is required to enable TLS on non-standard port?
by info＠vintageelectronics.ca 01 Mar '14

01 Mar '14

Kamailio running in Fedora 19 box. The client (Jitsi) can only connect over TCP. If connecting over UDP, the clients connect, but cannot send/receive messages. Cannot connect over TLS at all. Ultimately I will need to run with TLS over 5065 port or higher. Tried using different listen directives, different port directive, but kamailio either does not start, or only listening on one TCP port. I've set up two SRV records, one for _sip on port 5065 and another for _sips on port 5066. Both pointing to the Fedora box. In Jitsi I have to specify port # and when I set it to 5065 over TCP it works, but 5066 over TLS never connects. What am I doing wrong? Thank you.

3 8

Kamailio crashes with double qm_free
by Tuan Viet Nguyen 28 Feb '14

28 Feb '14

Hello, I have another kamailio crash with 4.1.1. The scenario is simple: + call a number, fail + retry on another peer, fails In this case, I use t_reply("404", "Not found") then exit. It seems that we have a double qm_free. Log file Feb 28 13:15:56 kamailio23 /usr/local/sbin/kamailio[29595]: WARNING: tm [t_lookup.c:1536]: t_unref(): WARNING: script writer didn't release transaction Feb 28 13:15:56 kamailio23 /usr/local/sbin/kamailio[29583]: : <core> [mem/q_malloc.c:468]: qm_free(): BUG: qm_free: freeing already freed pointer (0x7f9dedb3e110), called from tm: h_table.c: free_cell(157), first free tm: h_table.c: free_cell(157) - aborting Feb 28 13:15:56 kamailio23 /usr/local/sbin/kamailio[29627]: : <core> [pass_fd.c:293]: receive_fd(): ERROR: receive_fd: EOF on 19 Feb 28 13:15:56 kamailio23 /usr/local/sbin/kamailio[29576]: ALERT: <core> [main.c:775]: handle_sigs(): child process 29583 exited by a signal 6 Feb 28 13:15:56 kamailio23 /usr/local/sbin/kamailio[29576]: ALERT: <core> [main.c:778]: handle_sigs(): core was generated core.kamailio.29583 (gdb) bt full #0 0x00007f9df6afb475 in raise () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #1 0x00007f9df6afe6f0 in abort () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #2 0x0000000000548253 in qm_free (qm=0x7f9ded878000, p=0x7f9dedb3e110, file=0x7f9df515b62d "tm: h_table.c", func=0x7f9df515b7d8 "free_cell", line=157) at mem/q_malloc.c:470 f = 0x7f9dedb3e0e0 size = 140316274568736 next = 0x0 prev = 0x0 __FUNCTION__ = "qm_free" #3 0x00007f9df50ee2b3 in free_cell (dead_cell=0x7f9dedb39490) at h_table.c:157 b = 0x0 i = 2 rpl = 0x0 tt = 0x177dd596e30 foo = 0x7fffdd597400 cbs = 0x7f9dedb39490 cbs_tmp = 0x7fffdd596f10 __FUNCTION__ = "free_cell" #4 0x00007f9df511c662 in t_unref (p_msg=0x7f9df68bfdb0) at t_lookup.c:1546 kr = 12 __FUNCTION__ = "t_unref" #5 0x00007f9df5146f4d in w_t_unref (foo=0x7f9df68bfdb0, flags=2147483649, bar=0x0) at tm.c:765 No locals. #6 0x00000000004d6f27 in exec_post_script_cb (msg=0x7f9df68bfdb0, type=REQUEST_CB_TYPE) at script_cb.c:195 cb = 0x7f9df68a97d0 flags = 2147483649 #7 0x00000000004a6e24 in receive_msg (buf=0x921620 "ACK sip:0123456789@10.100.8.7 SIP/2.0\r\nVia: SIP/2.0/UDP 10.100.8.94;branch=z9hG4bKbcd8.9b0d4b2682a07fefaa22406761659624.0\r\nMax-Forwards: 15\r\nFrom: \"via_test\" <sip:0310193301@10.100.8.12>;tag=as4d03043"..., len=375, rcv_info=0x7fffdd5970d0) at receive.c:228 msg = 0x7f9df68bfdb0 ctx = {rec_lev = -581341280, run_flags = 32767, last_retcode = 5, jmp_env = {{__jmpbuf = {8857080, 0, 0, 0, 0, 140316433164786, 17179869189, 0}, __mask_was_saved = 8856992, __saved_mask = {__val = {1, 140316271892032, 1656210553, 3713626240, 1024, 8008593472, 140316271892032, 140736907014208, 5472467, 1577608202, 140316271892032, 50195, 140316271892032, 140316426284544, 8282899704, 140736907014272}}}}} ret = 0 inb = {s = 0x921620 "ACK sip:0123456789@10.100.8.7 SIP/2.0\r\nVia: SIP/2.0/UDP 10.100.8.94;branch=z9hG4bKbcd8.9b0d4b2682a07fefaa22406761659624.0\r\nMax-Forwards: 15\r\nFrom: \"via_test\" <sip:0310193301@10.100.8.12>;tag=as4d03043"..., len = 375} __FUNCTION__ = "receive_msg" #8 0x000000000053c0d8 in udp_rcv_loop () at udp_server.c:536 len = 375 buf = "ACK sip:0123456789@10.100.8.7 SIP/2.0\r\nVia: SIP/2.0/UDP 10.100.8.94;branch=z9hG4bKbcd8.9b0d4b2682a07fefaa22406761659624.0\r\nMax-Forwards: 15\r\nFrom: \"via_test\" <sip:0310193301@10.100.8.12>;tag=as4d03043"... tmp = 0x13e0751a0eb8aa4f <Address 0x13e0751a0eb8aa4f out of bounds> from = 0x7f9df68a5408 fromlen = 16 ri = {src_ip = {af = 2, len = 4, u = {addrl = {1577608202, 4290224}, addr32 = {1577608202, 0, 4290224, 0}, addr16 = {25610, 24072, 0, 0, 30384, 65, 0, 0}, addr = "\nd\b^\000\000\000\000\260vA\000\000\000\000"}}, dst_ip = {af = 2, len = 4, u = {addrl = { 117990410, 0}, addr32 = {117990410, 0, 0, 0}, addr16 = {25610, 1800, 0, 0, 0, 0, 0, 0}, addr = "\nd\b\a", '\000' <repeats 11 times>}}, src_port = 5060, dst_port = 5060, proto_reserved1 = 0, proto_reserved2 = 0, src_su = {s = {sa_family = 2, sa_data = "\023\304\nd\b^\000\000\000\000\000\000\000"}, sin = {sin_family = 2, sin_port = 50195, sin_addr = {s_addr = 1577608202}, sin_zero = "\000\000\000\000\000\000\000"}, sin6 = {sin6_family = 2, sin6_port = 50195, sin6_flowinfo = 1577608202, sin6_addr = {__in6_u = {__u6_addr8 = '\000' <repeats 15 times>, __u6_addr16 = {0, 0, 0, 0, 0, 0, 0, 0}, __u6_addr32 = {0, 0, 0, 0}}}, sin6_scope_id = 0}}, bind_address = 0x7f9df6704da8, proto = 1 '\001'} __FUNCTION__ = "udp_rcv_loop" #9 0x000000000046ec98 in main_loop () at main.c:1617 i = 5 pid = 0 si = 0x7f9df6704da8 si_desc = "udp receiver child=5 sock=10.100.8.7:5060\000\000\000\001", '\000' <repeats 19 times>, "\020\000\000\000\000\000\000\000y\304\267b\000\000\000\000\260vA\000\000\000\000\000\000tY\335\377\177", '\000' <repeats 18 times>, "@rY\335\377\177\000\000\002\266K\000\000\000\000" nrprocs = 8 __FUNCTION__ = "main_loop" #10 0x0000000000471c38 in main (argc=5, argv=0x7fffdd597408) at main.c:2533 cfg_stream = 0x164c010 c = -1 r = 0 tmp = 0x7fffdd597438 "\211~Y\335\377\177" tmp_len = 0 port = 5 proto = 0 options = 0x5de800 ":f:cm:M:dVIhEeb:l:L:n:vKrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:" ret = -1 seed = 4192398120 rfd = 4 debug_save = 0 debug_flag = 0 dont_fork_cnt = 0 n_lst = 0xbf p = 0x416bd9 "H\203\304\b\303" <Address 0x416bde out of bounds> __FUNCTION__ = "main" core.kamailio.29576 (gdb) bt full #0 0x00007f9df6afb475 in raise () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #1 0x00007f9df6afe6f0 in abort () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #2 0x0000000000548253 in qm_free (qm=0x7f9ded878000, p=0x7f9dedb3e110, file=0x7f9df515b62d "tm: h_table.c", func=0x7f9df515b7d8 "free_cell", line=157) at mem/q_malloc.c:470 f = 0x7f9dedb3e0e0 size = 140316274568736 next = 0x0 prev = 0x0 __FUNCTION__ = "qm_free" #3 0x00007f9df50ee2b3 in free_cell (dead_cell=0x7f9dedb39490) at h_table.c:157 b = 0x7f9df464b480 "dialog: dlg_cb.c" i = 0 rpl = 0x7f9df464b8e0 tt = 0x7f9df4a8cc75 foo = 0x5000548a29 cbs = 0xd000000001 cbs_tmp = 0x1edb086c8 __FUNCTION__ = "free_cell" #4 0x00007f9df50ef480 in free_hash_table () at h_table.c:441 p_cell = 0x7f9dedb39490 tmp_cell = 0x7f9deda23d90 i = 36299 __FUNCTION__ = "free_hash_table" #5 0x00007f9df5102d35 in tm_shutdown () at t_funcs.c:122 __FUNCTION__ = "tm_shutdown" #6 0x00000000004f8101 in destroy_modules () at sr_module.c:817 t = 0x7f9df670fd50 foo = 0x7f9df670f588 __FUNCTION__ = "destroy_modules" #7 0x00000000004689b2 in cleanup (show_status=1) at main.c:560 memlog = 32669 __FUNCTION__ = "cleanup" #8 0x0000000000469aab in shutdown_children (sig=15, show_status=1) at main.c:702 __FUNCTION__ = "shutdown_children" #9 0x000000000046b146 in handle_sigs () at main.c:793 chld = 0 chld_status = 134 memlog = 0 __FUNCTION__ = "handle_sigs" #10 0x000000000046f549 in main_loop () at main.c:1746 i = 8 pid = 29627 si = 0x0 si_desc = "udp receiver child=7 sock=91.213.79.31:5060\000\001", '\000' <repeats 19 times>, "\020\000\000\000\000\000\000\000y\304\267b\000\000\000\000\260vA\000\000\000\000\000\000tY\335\377\177", '\000' <repeats 18 times>, "@rY\335\377\177\000\000\002\266K\000\000\000\000" nrprocs = 8 __FUNCTION__ = "main_loop" #11 0x0000000000471c38 in main (argc=5, argv=0x7fffdd597408) at main.c:2533 cfg_stream = 0x164c010 c = -1 r = 0 tmp = 0x7fffdd597438 "\211~Y\335\377\177" tmp_len = 0 port = 5 proto = 0 options = 0x5de800 ":f:cm:M:dVIhEeb:l:L:n:vKrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:" ret = -1 seed = 4192398120 rfd = 4 debug_save = 0 debug_flag = 0 dont_fork_cnt = 0 n_lst = 0xbf p = 0x416bd9 "H\203\304\b\303" <Address 0x416bde out of bounds> __FUNCTION__ = "main" Thanks for your help

2 2

kamailio crashes
by Tuan Viet Nguyen 28 Feb '14

28 Feb '14

Hello, I have a crash with the following scenario: I try to route an INVITE to a remote host, it fails, I route to the second suing the dispatcher module. It fails the second time and I stop the call. It seems that I did not release the transaction in this case 93454 Feb 28 11:39:08 kamailio23 /usr/local/sbin/kamailio[20225]: WARNING: tm [t_lookup.c:1536]: t_unref(): WARNING: script writer didn't release transaction And then, Kamailio receives BYE of other dialogs and it crashes Here is core bt of 2 kamailio processes (4.1.1) (gdb) bt full #0 0x0000000000534e4e in timer_list_expire (t=1283469267, h=0x7ff9219e99b8, slow_l=0x7ff9219e9c88, slow_mark=12) at timer.c:883 tl = 0x7ff921c6b8f0 ret = 32767 #1 0x00000000005351ba in timer_handler () at timer.c:959 saved_ticks = 1283469267 run_slow_timer = 0 i = 12 __FUNCTION__ = "timer_handler" #2 0x0000000000535453 in timer_main () at timer.c:998 No locals. #3 0x000000000046efa2 in main_loop () at main.c:1688 i = 8 pid = 0 si = 0x0 si_desc = "udp receiver child=7 sock=91.213.79.31:5060\000\001", '\000' <repeats 19 times>, "\020\000\000\000\000\000\000\000\243Mgf\000\000\000\000\260vA\000\000\000\000\000\020\201\351a\377\177", '\000' <repeats 18 times>, "P\177\351a\377\177\000\000\002\266K\000\000\000\000" nrprocs = 8 __FUNCTION__ = "main_loop" #4 0x0000000000471c38 in main (argc=5, argv=0x7fff61e98118) at main.c:2533 cfg_stream = 0x1970010 c = -1 r = 0 tmp = 0x7fff61e98148 "\211\236\351a\377\177" tmp_len = 0 port = 5 proto = 0 options = 0x5de800 ":f:cm:M:dVIhEeb:l:L:n:vKrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:" ret = -1 seed = 519948236 rfd = 4 debug_save = 0 debug_flag = 0 dont_fork_cnt = 0 n_lst = 0xbf p = 0x416bd9 "H\203\304\b\303" <Address 0x416bde out of bounds> __FUNCTION__ = "main" (gdb) bt full #0 0x00007ff92abf3475 in raise () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #1 0x00007ff92abf66f0 in abort () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #2 0x0000000000548253 in qm_free (qm=0x7ff921970000, p=0x7ff921c3b940, file=0x7ff92925362d "tm: h_table.c", func=0x7ff9292537d8 "free_cell", line=178) at mem/q_malloc.c:470 f = 0x7ff921c3b910 size = 5600 next = 0x7fff61e97ac0 prev = 0x7ff9291e571e __FUNCTION__ = "qm_free" #3 0x00007ff9291e64a9 in free_cell (dead_cell=0x7ff921c6b870) at h_table.c:178 b = 0x0 i = 1 rpl = 0x0 tt = 0x0 foo = 0x5000548a29 cbs = 0x0 cbs_tmp = 0x7ff921c3b940 __FUNCTION__ = "free_cell" #4 0x00007ff9291e7480 in free_hash_table () at h_table.c:441 p_cell = 0x7ff921c6b870 tmp_cell = 0x7ff921a159f0 i = 37066 __FUNCTION__ = "free_hash_table" #5 0x00007ff9291fad35 in tm_shutdown () at t_funcs.c:122 __FUNCTION__ = "tm_shutdown" #6 0x00000000004f8101 in destroy_modules () at sr_module.c:817 t = 0x7ff92a807d50 foo = 0x7ff92a807588 __FUNCTION__ = "destroy_modules" #7 0x00000000004689b2 in cleanup (show_status=1) at main.c:560 memlog = 32761 __FUNCTION__ = "cleanup" #8 0x0000000000469aab in shutdown_children (sig=15, show_status=1) at main.c:702 __FUNCTION__ = "shutdown_children" #9 0x000000000046b146 in handle_sigs () at main.c:793 chld = 0 chld_status = 139 memlog = 0 __FUNCTION__ = "handle_sigs" #10 0x000000000046f549 in main_loop () at main.c:1746 i = 8 pid = 20250 si = 0x0 si_desc = "udp receiver child=7 sock=91.213.79.31:5060\000\001", '\000' <repeats 19 times>, "\020\000\000\000\000\000\000\000\243Mgf\000\000\000\000\260vA\000\000\000\000\000\020\201\351a\377\177", '\000' <repeats 18 times>, "P\177\351a\377\177\000\000\002\266K\000\000\000\000" nrprocs = 8 __FUNCTION__ = "main_loop" #11 0x0000000000471c38 in main (argc=5, argv=0x7fff61e98118) at main.c:2533 cfg_stream = 0x1970010 c = -1 r = 0 tmp = 0x7fff61e98148 "\211\236\351a\377\177" tmp_len = 0 port = 5 proto = 0 options = 0x5de800 ":f:cm:M:dVIhEeb:l:L:n:vKrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:" ret = -1 seed = 519948236 rfd = 4 debug_save = 0 debug_flag = 0 dont_fork_cnt = 0 n_lst = 0xbf p = 0x416bd9 "H\203\304\b\303" <Address 0x416bde out of bounds> __FUNCTION__ = "main" Thank you for your help

2 1

Add Header
by Tony Turner 28 Feb '14

28 Feb '14

Hi, customers PBX----trunk-----kamailio------------freeswitch-------------carrier We already add Trusted IP’s via SIREMIS 4.0 is it possible with Kamailio to have customers calls come in on a trusted IP and when they send calls we add a header with a [client id] which we can remove on the next ip switch upstream So example Call comes in from customer with IP: their trusted IP source xx.xx.xx.xx add header with their client ID route to PSTN Then my upstream switch will process the customers call and remove the header before sending to our voip carrier. Basically my ip switch will evaluate the client id header for billing, process and strip the header back out. Thanks Tony

2 2

proposed patch for the registrar module - crash fix
by Dragos Oancea 28 Feb '14

28 Feb '14

Hi We think that we found and fix a bug in the registrar module. The bug is hard to reproduce, and it crashes our Kamailio from time to time (once at 2-3 weeks for us) . In save.c, function update_contacts() there are two places where we free() a pointer and then we reuse it, line 700 and line 730: while(ptr){ ptr0 = ptr; if(ptr!=c) ul.delete_ucontact(_r, ptr); ptr=ptr0->next; } And then from inside delete_ucontact(_r,ptr) we call mem_delete_ucontact(_r, _c) which calls free_ucontact(_c) , which calls shm_free(_c) . _c is actually our ptr. If another process writes at the location pointed by ptr during the "while" loop , the current process will crash. The bug affects 4.0.4, 4.0.5 and maybe older versions. We keep the location table in memory, no database backend, and we do alot of REGISTER/un-REGISTER in our environment, and maybe that's why this bug was not spotted by the community before. Just to clarify, this fix is in relation with this post: http://lists.sip-router.org/pipermail/sr-dev/2014-February/022934.html Please take into consideration the attached patch provided by the Libon Voice Team. Regards, Dragos Oancea

2 1

Mobile 3G Block from provider
by Rene Montilva 28 Feb '14

28 Feb '14

Hi List I'm trying to call over 3G networks, but doesn't work, i think my mobile provider block voip traffic, is there any way to hide voip traffic or jump the provider block rule ??

2 1

keep buddy list - best way
by Mihai Marin 27 Feb '14

27 Feb '14

Hello Sirs, I'm a little confused getting a decision about implementing buddy list (step: presence & IM). What is the best way for keeping the contacts list: xmpp or xcap? You will say probably xcap server but even if I intend to use in the future an xmpp gateway in order to be able to communicate sip with with xmpp servers like google - google talk? And why xcap? Because most of the clients has the possibility to connect to a xcap server to get the buddy list? From web perspective I think both are OK because of the xhttp module power in kamailio to get the buddy list xml. Thank you. Best regards, Mihai M

1 0

kamailio error, failed to start.
by Arya Farzan 27 Feb '14

27 Feb '14

I just installed kamailio on Debian 7 and when I start it with init.d I get the following messages. root@voip:/etc/kamailio# /etc/init.d/kamailio start Starting Kamailio: loading modules under /usr/lib64/kamailio/modules_k/:/usr/lib64/kamailio/modules/ Listening on udp: 127.0.0.1:5060 udp: MYSERVERIP:5060 tcp: 127.0.0.1:5060 tcp: MYSERVERIP:5060 tls: 127.0.0.1:5061 tls: MYSERVERIP:5061 Aliases: tls: voip:5061 tls: localhost:5061 tcp: voip:5060 tcp: localhost:5060 udp: voip:5060 udp: localhost:5060 kamailio error, failed to start. root@voip:/etc/kamailio# How can I find out what the problem is?

3 4

Re: [SR-Users] Replacing an ACME Packet Net-Net SBC
by Raúl Alexis Betancor Santana 27 Feb '14

27 Feb '14

Sorry ... I disagree at all .... All that features you mention, could be done with kamalio ... BUT you have to work on your cfg to get them. You don't get then out of the box. Kamailio it's a powerfull tool ... It's not a final product as the AP Net SBC Best regards Luis Silva <luisfilsilva(a)gmail.com> wrote: >_______________________________________________ >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

kamailio with mediaproxy-ng, 488 Not Acceptable Here
by Mihai Marin 26 Feb '14

26 Feb '14

Hello Sirs, I have a problem configuring kamailio with mediaproxy-ng and I'm asking for help. I managed to build everything, kamailio find support for mediaproxy-ng using rtpproxy-ng. When I'm trying to make a call from Web using my phone's internet provider to my computer's web I get 488 Not Acceptable Here. Swithing the caller I get no video. I have used the kamailio-advanced.cfg generated and added websocket support. The call in my network is working perfect. Can you help me with this? mediaproxy-ng log: mediaproxy-ng[14896]: Returning to SIP proxy: d7:createdi1390864117e7:streamslld3:tag10:trhh9viigs6:status34:known but unconfirmed peer address5:statsd3:rtpd8:countersd7:packetsi0e5:bytesi0e6:errorsi0ee12:peer addressd6:family4:IPv47:address13:213.233.85.554:porti48279ee23:advertised peer addressd6:family4:IPv47:address13:213.233.85.554:porti48279ee10:local porti30008ee4:rtcpd8:countersd7:packetsi0e5:bytesi0e6:errorsi0ee12:peer addressd6:family4:IPv47:address13:213.233.85.554:porti48280ee23:advertised peer addressd6:family4:IPv47:address13:213.233.85.554:porti48280ee10:local porti30009eeeed3:tag0:6:status20:unknown peer address5:statsd3:rtpd8:countersd7:packetsi0e5:bytesi0e6:errorsi0ee12:peer addressd6:family4:IPv67:address2:::4:porti0ee23:advertised peer addressd6:family4:IPv67:address2:::4:porti0ee10:local porti30010ee4:rtcpd8:countersd7:packetsi0e5:bytesi0e6:errorsi0ee12:peer addressd6:family4:IPv67:address2:::4:porti0ee23:advertised peer addressd6:family4:IPv67:address2:::4:porti0ee10:local porti30011eeeeee6:totalsd5:inputd3:rtpd7:packetsi0e5:bytesi0e6:errorsi0ee4:rtcpd7:packetsi0e5:bytesi0e6:errorsi0eee6:outputd3:rtpd7:packetsi0e5:bytesi0e6:errorsi0ee4:rtcpd7:packetsi0e5:bytesi0e6:errorsi0eeee6:result2:oke kamailio: 11(14926) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio-advanced.cfg] l=860 a=24 n=rtpproxy_manage 11(14926) DEBUG: rtpproxy-ng [rtpproxy.c:1333]: rtpp_function_call(): proxy reply: d7:createdi1390864117e7:streamslld3:tag10:trhh9viigs6:status34:known but unconfirmed peer address5:statsd3:rtpd8:countersd7:packetsi0e5:bytesi0e6:errorsi0ee12:peer addressd6:family4:IPv47:address13:213.233.85.554:porti48279ee23:advertised peer addressd6:family4:IPv47:address13:213.233.85.554:porti48279ee10:local porti30008ee4:rtcpd8:countersd7:packetsi0e5:bytesi0e6:errorsi0ee12:peer addressd6:family4:IPv47:address13:213.233.85.554:porti48280ee23:advertised peer addressd6:family4:IPv47:address13:213.233.85.554:porti48280ee10:local porti30009eeeed3:tag0:6:status20:unknown peer address5:statsd3:rtpd8:countersd7:packetsi0e5:bytesi0e6:errorsi0ee12:peer addressd6:family4:IPv67:address2:::4:porti0ee23:advertised peer addressd6:family4:IPv67:address2:::4:porti0ee10:local porti30010ee4:rtcpd8:countersd7:packetsi0e5:bytesi0e6:errorsi0ee12:peer addressd6:family4:IPv67:address2:::4:porti0ee23:advertised peer addressd6:family4:IPv67:address2:::4:porti0ee10:local porti30011eeeeee6:totalsd5:inputd3:rtpd7:packetsi0e5:bytesi0e6:errorsi0ee4:rtcpd7:packetsi0e5:bytesi0e6:errorsi0eee6:outputd3:rtpd7:packetsi0e5:bytesi0e6:errorsi0ee4:rtcpd7:packetsi0e5:bytesi0e6:errorsi0eeee6:result2:oke 11(14926) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio-advanced.cfg] l=867 a=16 n=if 11(14926) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio-advanced.cfg] l=862 a=24 n=is_request 11(14926) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio-advanced.cfg] l=866 a=16 n=if 11(14926) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio-advanced.cfg] l=863 a=24 n=has_totag 11(14926) DEBUG: siputils [checks.c:103]: has_totag(): no totag 11(14926) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio-advanced.cfg] l=864 a=25 n=add_rr_param 11(14926) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio-advanced.cfg] l=873 a=16 n=if 11(14926) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio-advanced.cfg] l=867 a=24 n=is_reply 11(14926) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio-advanced.cfg] l=873 a=2 n=return 11(14926) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio-advanced.cfg] l=1009 a=16 n=if 11(14926) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio-advanced.cfg] l=989 a=24 n=t_is_canceled 11(14926) DEBUG: tm [t_lookup.c:1071]: t_check_msg(): DEBUG: t_check_msg: msg id=2 global id=2 T start=0x7f62174df4e0 11(14926) DEBUG: tm [t_lookup.c:1143]: t_check_msg(): DEBUG: t_check_msg: T already found! 11(14926) DEBUG: tm [t_reply.c:1827]: relay_reply(): DEBUG: relay_reply: branch=0, save=0, relay=0 icode=0 11(14926) DEBUG: <core> [msg_translator.c:2007]: generate_res_buf_from_sip_res(): old size: 397, new size: 305 11(14926) DEBUG: <core> [msg_translator.c:2025]: generate_res_buf_from_sip_res(): copied size: orig:125, new: 33, rest: 272 msg= SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/WS ebhg3v0qb6fm.invalid;rport=1034;received=213.233.85.55;branch=z9hG4bK8048296 To: <sip:bob@93.187.138.214>;tag=r6dc2287g9 From: "Alice Test" <sip:alice@93.187.138.214>;tag=trhh9viigs Call-ID: fvgga4ikm8vrvuji0g0n CSeq: 6228 INVITE Content-Length: 0 11(14926) DEBUG: websocket [ws_frame.c:713]: ws_frame_transmit(): Tx message: SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/WS ebhg3v0qb6fm.invalid;rport=1034;received=213.233.85.55;branch=z9hG4bK8048296 To: <sip:bob@93.187.138.214>;tag=r6dc2287g9 From: "Alice Test" <sip:alice@93.187.138.214>;tag=trhh9viigs Call-ID: fvgga4ikm8vrvuji0g0n CSeq: 6228 INVITE Content-Length: 0 Thank you. Best regards, Mihai M

8 27

coredump on using jsonprc_request
by davy van de moere 26 Feb '14

26 Feb '14

Gents, I’ve stumbled upon a coredump when using jsonrpc_request. I might be expecting the wrong things from jsonrpc_request. I was hoping to use it in combination with CNXCC, and have jsonrpc_request get a maximum duration for every call, as a fraud avoidance mechanism. I tried this with the branch 4.1 and master. Both behave identical. So from the kamailio-basic.cfg example, I just added this line around line 350 (just above route(SIPOUT)): jsonrpc_request("bar.sendsms", "{'ru':'$rU','rb':'$rb'}", "JSONRESPONSE", "JSONERROR", "$var(result)"); Doing a ngrep on the netstrings I have: interface: venet0 (127.0.0.2/255.255.255.255) filter: (ip or ip6) and ( port 7080 ) # T 10.66.8.42:44995 -> 192.168.60.45:7080 [AP] 94:{ "id": 2, "jsonrpc": "2.0", "method": "bar.sendsms", "params": { "ru": "", "rb": "" } }, ## T 192.168.60.45:7080 -> 10.66.8.42:44995 [AP] 43:{"jsonrpc": "2.0", "result": "12", "id": 2}, ####^Cexit 7 received, 0 dropped I do get some results from the JSONRESPONSE route. But when launching a call, or when sending just a register packet, kamailio gives up with: Feb 26 06:29:11 SBC01 kamailio[18043]: WARNING: tm [t_lookup.c:1537]: t_unref(): WARNING: script writer didn't release transaction Feb 26 06:29:12 SBC01 kamailio[18033]: : [mem/q_malloc.c:468]: qm_free(): BUG: qm_free: freeing already freed pointer (0xb4c01624), called from tm: h_table.c: free_cell(157), first free tm: h_table.c: free_cell(157) - aborting Feb 26 06:29:12 SBC01 kamailio[18023]: ALERT: [main.c:775]: handle_sigs(): child process 18033 exited by a signal 6 Feb 26 06:29:12 SBC01 kamailio[18023]: ALERT: [main.c:778]: handle_sigs(): core was generated Feb 26 06:29:12 SBC01 kamailio[18023]: INFO: [main.c:790]: handle_sigs(): INFO: terminating due to SIGCHLD Feb 26 06:29:12 SBC01 kamailio[18041]: INFO: [main.c:841]: sig_usr(): INFO: signal 15 received Feb 26 06:29:12 SBC01 kamailio[18047]: INFO: [main.c:841]: sig_usr(): INFO: signal 15 received Feb 26 06:29:12 SBC01 kamailio[18043]: INFO: [main.c:841]: sig_usr(): INFO: signal 15 received Feb 26 06:29:12 SBC01 kamailio[18048]: INFO: [main.c:841]: sig_usr(): INFO: signal 15 received Feb 26 06:29:12 SBC01 kamailio[18023]: : [mem/q_malloc.c:468]: qm_free(): BUG: qm_free: freeing already freed pointer (0xb4c01624), called from tm: h_table.c: free_cell(157), first free tm: h_table.c: free_cell(157) - aborting Using the jsonrpc_notification works like a charm. I got coredumps of it, if someone is interested. Side questions are offcourse, am I aiming to use jsonrpc_request for the correct purpose? Are there any other good alternatives? (next to using plain db queries?) Grtz, Davy -- davy van de moere

1 0

Kamailio World 2014 - Agenda
by Daniel-Constantin Mierla 26 Feb '14

26 Feb '14

The first version of the schedule during Kamailio World 2014 <http://www.kamailioworld.com> [1] has been published. Over 30 sessions grouped in two days of conference and pre-conference technical tutorials, spanning from various use cases for Kamailio to presentations approaching related VoIP applications such as Jitsi, Sems, Asterisk or FreeSwitch. Among the topics: WebRTC, IMS or VoIP security and fraud detection, tools for testing and troubleshooting VoIP systems, scalability with OpenStack and asynchronous processing, real time charging. First version of agenda has been published at: - http://conference.kamailio.com/k02/schedule/ Information about speakers can be found at: - http://conference.kamailio.com/k02/speakers/ Many thanks to the sponsors [2], they are making possible this event: FhG Fokus <http://www.fokus.fraunhofer.de/en/fokus/index.html>, Asipto <http://www.asipto.com>, Sipwise <http://www.sipwise.com>, Sipgate <http://www.sipgate.de>, IT Center <http://www.itcenter.com.pt/>, Simwood <http://www.simwood.com>, and NG Voice <http://www.ng-voice.com/>. Should you consider sponsoring, contact us via email at <conference [at] kamailio.com>**and we can present you more details. If you plan to attend, do not delay the registration, the number of seats is limited – see more details in Registration page <http://conference.kamailio.com/k02/registration/> [3]. /Looking forward to meeting many of you in Berlin! / Daniel/ / [1] - http://conference.kamailio.com [2] - http://conference.kamailio.com/k02/sponsors/ [3] - http://conference.kamailio.com/k02/registration/ -- Daniel-Constantin Mierla http://www.asipto.com

1 0

debian packages from opensuse.org
by Jöran Vinzens 26 Feb '14

26 Feb '14

Hi All, we are using Kamailio 4.1.0 packages from opensuse: http://download.opensuse.org/repositories/home:/kamailio:/telephony-debs/De… During installation it seems, the packages are malformed. Installation of kamailio-mysql-modules and kamailio-xml-modules stops during unpacking with failure: "trying to overwrite '/usr/lib64/kamailio/libsrdb2.so.1.0', which is also in package kamailio 4.1.0" it seem this one file is part of all three packages. There is a way to enforce the installation using dpkg -i --force-all.. but it seems not the right way. Does anyone have an idea how to solve without building own packages? Maybe this is an bug in kamailio Project? Many thanks for all proposals. BR Jöran -- Jöran Vinzens sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois Steuernummer: 106/5724/7147, Umsatzsteuer-ID: DE219349391 www.sipgate.de - www.sipgate.at - www.sipgate.co.uk

2 1

Re: [SR-Users] RTPProxy/Mediaproxy issue
by Daniel Grotti 25 Feb '14

25 Feb '14

Hi, Yes packet loss is very high. Please investigate around that. Daniel Ravi <wingsravi777(a)gmail.com> wrote: >_______________________________________________ >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

2 5

Problems while reading from location and parsing expires
by Jöran Vinzens 25 Feb '14

25 Feb '14

Hi All, we are upgrading from kamailio 3.1 to 4.1 and we got stuck at a Problem with usrloc module. While starting up kamailio reads location from DB and try to parse expires date from DB. this is the Problem: Feb 25 11:42:11 hagi /usr/sbin/kamailio[19580]: ERROR: <core> [db_ut.c:90]: db_str2int(): Unexpected characters: [-02-25 11:27:08] Feb 25 11:42:11 hagi /usr/sbin/kamailio[19580]: ERROR: <core> [db_val.c:75]: db_str2val(): error while converting integer value from string Feb 25 11:42:11 hagi /usr/sbin/kamailio[19580]: ERROR: db_mysql[km_row.c:68]: db_ mysql_convert_row(): failed to convert value Feb 25 11:42:11 hagi /usr/sbin/kamailio[19580]: ERROR: db_mysql [km_dbase.c:375]: db_mysql_fetch_result(): error while converting row #0 Feb 25 11:42:11 hagi /usr/sbin/kamailio[19580]: ERROR: usrloc [udomain.c:420]: preload_udomain(): fetching rows failed usrloc Module is configured with db_mode "1" so it saves it's entries by itself. When i have a look on the database it looks good and the expires field is an datetime. From logs it looks like it try to parse a unixtimestamp instead of the datetime. when i truncate the location table it starts perfectly, since there are no entries stored where it could fail parsing expires date. does anybody has similar issue or any suggestions how to solve? found in Version: 4.1.0 many thanks! BR Jöran -- Jöran Vinzens sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois Steuernummer: 106/5724/7147, Umsatzsteuer-ID: DE219349391 www.sipgate.de - www.sipgate.at - www.sipgate.co.uk

2 4

Re: [SR-Users] Kamailio as redirect
by Daniel Tryba 24 Feb '14

24 Feb '14

On Monday 24 February 2014 14:45:18 Mike Claudi Pedersen wrote: > okay i somewhat get that.. but cant i use something like a lookup in a > textfile instead of a db? You can. alias_db also accepts dbtext as backend: http://kamailio.org/docs/modules/stable/modules/db_text.html But TIMTOWTDI, alias_db was just a suggestion. Please keep the mailing list included. -- POCOS B.V. - Croy 9c - 5653 LC Eindhoven Telefoon: 040 293 8661 - Fax: 040 293 8658 http://www.pocos.nl/ - http://www.sipo.nl/ K.v.K. Eindhoven 17097024

1 0

Re: [SR-Users] Configuring Kamailio as an upstream proxy for FreeSwitch and which RTP proxy to choose
by Fred Posner 24 Feb '14

24 Feb '14

On Mon Feb 24 10:50:07 CET 2014 Sean Kemball wrote: > New to Kamailio and FreeSwitch, loosely familiar with SIP mechanics, > and not a complete network idiot... but please be gentle. :) Welcome!s, > Questions: > > 1. Should the proposed topology, with Kamailio + an RTP proxy > behind a firewall, relaying to FS on an inside interface, work? > (Can't see why not) Yes, you said that your upstream is on the same private network. So it should be pretty straight forward. > 2. Does it need a local RTP proxy on the Kamailio box, particularly > if we turn off the ASA SIP inspect stuff? If you are all on the same private network, I would let FreeSWITCH handle the RTP, but you can do this a variety of ways. > 3. Can you recommend which RTP proxy to use? There seem to be at > least 3 that work with Kamailio. The box is CentOS 6.5, and it would > be nice to use known-to-work packages rather than compile from source. > (But eh, if I haveta). On your scenario, I'd just use FreeSWITCH for the media proxy. Again, many different ways to go here. > 4. Can anyone point me to some docs to explain what ports need to > be open between the Kamailio box and my upstream proxy/media server? > I can be more liberal between inside and DMZ I guess. Your upstream provider would generally tell you which rtp ports they would want opened. > 5. Is static NAT in this environment going to bite me, or should > it be OK? I've never had an upstream provider communicate with me on private nat. > 6. Is there any better documentation that we should be using to > make this easier, or should I just man up and try harder? Man up. =) Practive makes perfect. -- Fred Posner The Palner Group, Inc.

1 0

pcscf - removing registration entries from hash table
by Daniel Ciprus 24 Feb '14

24 Feb '14

Hello, Our current setup is Kamailio as IMS core and we're observing following behavior: Clients register themselves over TCP so pcscf creates hash table which keeps these information in memory. Once tcp reset occurs for any reason, registration entry stays in the memory and client re-registers himself with new pair ip:port. This kind of behavior creates problem because outbound SIP messages are being forked and requests are sent to non-existent tcp sessions which creates a lot of timeouts and confusion for AS. We're using ims_usrloc_pcscf module to keep reg entries in and as I noticed, with usrloc module, there was an option which automatically removed entry from hash table once tcp session was closed which is not the case in this module. Any suggestions how to get out of this ? thanks for any hints/suggestions. -- Daniel Ciprus Integration engineer http://www.acision.com 9954 Mayland Dr Suite 3100 Richmond, VA 23233 USA T: +1 804 762 5601 E: daniel.ciprus(a)acision.com<mailto:daniel.ciprus@acision.com> ________________________________ This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you for understanding.

1 0

Kamailio as redirect
by Mike Claudi Pedersen 24 Feb '14

24 Feb '14

i want to setup a test environment to see how kamailio handles redirection. i want to be able to reroute calls between servers, where do i insert the configuration for this? can i append a list of user and destination to make the configuration go through, to see where the call is going? im new at kamailio, and im having a really hard time finding the right documentation to support a redirect server. could somone please give me an example? lets say i have 3 user 1000 1001 1002 and they are at 3 different servers eg. 1000(a)192.168.1.10 1001(a)192.168.1.11 1002(a)192.168.1.12 then i call user 1002 from user 1000, through the kamailio redirect server, which redirects to corresponding servers. Mike Claudi Pedersen

2 1

Configuring Kamailio as an upstream proxy for FreeSwitch and which RTP proxy to choose
by Sean Kemball 24 Feb '14

24 Feb '14

Hi New to Kamailio and FreeSwitch, loosely familiar with SIP mechanics, and not a complete network idiot... but please be gentle. :) We're trying to get Kamailio set up in front of a FreeSwitch-based SIP application server, to do some simple policy controls and, using one of the RTP proxy modules, to help handle the media side of things. Our SIP trunks come from our provider on a VLAN addressed as 10.x, and they provide an upstream proxy/media server (10.y address). The link also carries Internet traffic so we're running it through a Cisco ASA which breaks out the VLANs and static NATs the SIP stuff to our DMZ (10.z address). This is where we want to put the Kamailio box, where it should receive the calls and route them back through the ASA to the internal address (10.w range - all 10.w/x/y/z are separate, non-overlapping networks). Outbound calls from the application server (there are very few) should pass back up to Kamailio which will validate the numbers against an approved list (the app should only dial a subset of numbers) and pass them to the upstream proxy if necessary. We need to ensure that inbound calls from outside trombone through our application server so that the caller doesn't get billed for any calls, but this should be the default I think as that side of things is handled by FS (the app stands up a new call and bridges it to the incoming). This topology should allow us to use Kamailio to hide details of the FS app from the outside world, and make life easier for the app developer who should just send/receive to the Kamailio box with no further thought or complexity. If we put the FreeSwitch box in the DMZ where we want to put Kamailio, and then turn on SIP packet inspection in the ASA, calls flow but quality is poor for some callers. If we turn off SIP packet inspection, we get no audio - I can't find any clear Cisco documentation for this but I think the SIP inspection stuff in the ASA seems to handle RTP NAT fixups and the like too. But with no docs it may as well be magic, and I want to remove it if possible. With Kamailio in the DMZ and FS internal, we loosely followed http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc and hacked out the voicemail and Lua stuff, but couldn't get calls to flow, whether or not we used the SIP inspection on the ASA. We were confused why this example isn't define as WITH_NAT and doesn't use a local RTP proxy. Surely that would be needed? Questions: 1. Should the proposed topology, with Kamailio + an RTP proxy behind a firewall, relaying to FS on an inside interface, work? (Can't see why not) 2. Does it need a local RTP proxy on the Kamailio box, particularly if we turn off the ASA SIP inspect stuff? 3. Can you recommend which RTP proxy to use? There seem to be at least 3 that work with Kamailio. The box is CentOS 6.5, and it would be nice to use known-to-work packages rather than compile from source. (But eh, if I haveta). 4. Can anyone point me to some docs to explain what ports need to be open between the Kamailio box and my upstream proxy/media server? I can be more liberal between inside and DMZ I guess. 5. Is static NAT in this environment going to bite me, or should it be OK? 6. Is there any better documentation that we should be using to make this easier, or should I just man up and try harder? TIA Sean Volunteer for our Red Puppy street appeal to help puppies like Gordy become guide dogs. ############################################################################# This email, including any attachments, is intended solely for the addressee(s) It is confidential and may be legally privileged. If you are not the intended recipient, you must not copy, disclose, distribute or otherwise use it or the information in it. Please notify the sender at once and delete it from your system immediately. Any views or opinions expressed are solely those of the author and do not necessarily represent those of Blind Foundation. The Blind Foundation does not accept responsibility for any viruses or other malicious code that may be transmitted with this email. #############################################################################

1 0

Change from_uri
by arun Jayaprakash 24 Feb '14

24 Feb '14

Hello, I have a problem that my calls get rejected by he pstn gateway as my gateway is expecting a DID number instead of a 4 digit extension as the 'from user". Let me explain my situation: 1. I have set callfwd_busy in my user preference table. 2. In my failure route I check to see if callfwd_busy has been set, if it has then I execute the following script: if (t_check_status("486|408")) { $avp(dst_number) = $rU; if (avp_db_load("$avp(dst_number)", "$avp(callfwd_busy)")) { xlog("LOG: avp(callfwd_busy)=$avp(callfwd_busy), avp(dst_number)=$avp(dst_number)\n"); } xlog("LOG: Failure route with $rU $avp(dst_number) \n"); $rU = $avp(callfwd_busy); $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); route(RELAY); exit; } When I call from extension 7004 to ext 7002 ( this has callfwd_busy set to a DID number) the trace looks as follows: U 172.10.30.15:5080 -> 172.10.30.15:5060 SIP/2.0 486 Busy Here. Via: SIP/2.0/UDP 54.200.xx.yy:5060;branch=z9hG4bKee75.ab791c5.0;received=172.10.30.15. Via: SIP/2.0/UDP 192.168.1.8:5060;received=71.252.219.63;branch=z9hG4bK209939211;rport=1025. From: "7004" <sip:7004@ajfmc1.myDomain.net>;tag=1830267367. To: <sip:7002@ajfmc1.myDomain.net>;tag=as17cd0b40. Call-ID: 876294364-5060-74(a)BJC.BGI.B.I. CSeq: 731 INVITE. Server: Asterisk PBX 1.8.17.0. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Content-Length: 0. U 172.10.30.15:5060 -> 172.10.30.15:5080 INVITE sip:18455981234@64.136.xx.yy:5060 SIP/2.0. Record-Route: <sip:54.200.xx.xx;lr=on;ftag=1830267367;nat=yes>. Via: SIP/2.0/UDP 54.200.xx.xx:5060;branch=z9hG4bKee75.ab791c5.1. Via: SIP/2.0/UDP 192.168.1.8:5060;received=71.252.xx.yy;branch=z9hG4bK209939211;rport=1025. From: "7004" <sip:7004@ajfmc1.myDomain.net>;tag=1830267367. To: <sip:7002@ajfmc1.myDomain.net>. Call-ID: 876294364-5060-74(a)BJC.BGI.B.I. CSeq: 731 INVITE. Contact: "7004" <sip:7004@71.252.xx.yy:1025>. Max-Forwards: 16. User-Agent: Grandstream HT701 1.0.4.8. Privacy: none. P-Preferred-Identity: "7004" <sip:7004@ajfmc1.myDomain.net>. Supported: replaces, path, timer, eventlist. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE. Content-Type: application/sdp. Accept: application/sdp, application/dtmf-relay. Content-Length: 453. . v=0. o=7004 8000 8000 IN IP4 172.10.30.1554.200.xx.xx. s=SIP Call. c=IN IP4 172.10.30.1554.200.xx.xx. t=0 0. m=audio 6248262482 RTP/AVP 0 18 4 8 2 97 101. a=sendrecv. a=rtpmap:0 PCMU/8000. a=ptime:20. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:4 G723/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:97 iLBC/8000. a=fmtp:97 mode=20. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16,32-36,54. a=nortpproxy:yes. a=nortpproxy:yes. # U 172.10.30.15:5080 -> 172.10.30.15:5060 SIP/2.0 503 Unavailable. Via: SIP/2.0/UDP 54.200.xx.xx:5060;branch=z9hG4bKee75.ab791c5.1;received=172.10.30.15. Via: SIP/2.0/UDP 192.168.1.8:5060;received=71.252.xx.yy;branch=z9hG4bK209939211;rport=1025. From: "7004" <sip:7004@ajfmc1.myDomain.net>;tag=1830267367. To: <sip:7002@ajfmc1.myDomain.net>;tag=as17cd0b40. Call-ID: 876294364-5060-74(a)BJC.BGI.B.I. CSeq: 731 INVITE. Server: Asterisk PBX 1.8.17.0. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Content-Length: 0. It looks like my gateway is expecting to see a DID number in the from header. Can someone let me know how I can dot this? Thank you. Arun

3 4

Query about RTP flow
by Nandini madhu 23 Feb '14

23 Feb '14

Dear All, I have read about SIP and RTP packet flows but ended with a question about RTP packets flow logic between client-to client when the SIP session starts. I have a running Kamailio (V 4.0.4) server on ubuntu (12.04) platform which is also configured fro DNS and DHCP servers. And following are my test bed set-ups: Kamailio server (DNS+ DHCP+Kamailio) | Router <----------------------------> Switch<----------------------------> Router | | clientA <--------------------- RTP(audio/video) ? --------------------> client B Now when the SIP session established (SIP Packets have to pass through all the routers and switch to reach Kamailio server), RTP session will start between Client A and B. But What is meant by client to client here ? what is the RTP packets path between the clients to reach their destinations ? Is the RTP packets have to pass through Routers and switch to reach its destination client ? (i.e Client A--->Router 1---->switch -----> Router 2-----> client B) ? What if i connect both client A and client B to the same router ? PS: I have not configured any media-relays (RTPproxy/Media-proxy) Please help me in clarifying all this doubts. Anybody's help will greatly appreciate. Regards, Nandini

1 0

HTABLE update from DB
by Uri Shacked 23 Feb '14

23 Feb '14

It does not work...I tried with autoexpire=0, I tried with no autoexpires define, I Tried defining a number and see if the value is updated.... this is my table definition: modparam("htable", "htable", "A=>size=8,dbtable=aaaaaa;initial=0,dbmobe=0;") I remind you i need the memory to be updated from the DB every interval, and not the DB form the memory. thanks. >Declare the table without autoexpire. Check again the module README. >Regards, >Ovidiu Sas >On Feb 23, 2014 11:23 AM, "Uri Shacked" <ushacked at gmail.com <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>> wrote: >>* Hi, *>>>>* Following my issue with reloading data. I am thinking of a way to keep *>>* data updated in memory. *>>* I read the HTABLE module again and notice the "db_expires" option. *>>* It does not work for me... *>>>>* I try to set it so an item will expire in 60 sec, and when expired, it *>>* will be reloaded from DB and not deleted. *>>>>* Any ideas? *>>>>* Uri*

2 1

RTPProxy/Mediaproxy issue
by Wingsravi R 23 Feb '14

23 Feb '14

Dear Kamailio'ns, I have configured kamailio (V 4.0) with RTPproxy server. Audio/Video call establishments are just fine. With this i have some unknown behaviour in my set-up is that When i am running kamailio server with RTPproxy instance, i am experiencing Jitter, Latency, pixelled audio/video calls. But when i run only Kamailio server without RTPproxy service running, audio/video calls are just fine (no any jitter, latency kind of issues). I also tried with Mediaproxy (AG projects) server but the results are same as like in Kamailio/RTPproxy case. What could be the problem with this set-up ? How can i solve this issue ? Please anybody help me in resolving this problem. Any help will greatly appreciate. Regards. Ravi.

4 7

HTABLE update from DB
by Uri Shacked 23 Feb '14

23 Feb '14

Hi, Following my issue with reloading data. I am thinking of a way to keep data updated in memory. I read the HTABLE module again and notice the "db_expires" option. It does not work for me... I try to set it so an item will expire in 60 sec, and when expired, it will be reloaded from DB and not deleted. Any ideas? Uri

2 1

Using Kamailio as a websocket LB
by Luis Silva 22 Feb '14

22 Feb '14

Hi guys, Is it possible to use Kamailio as a Websocket LoadBalancer (transparent to the Websocket content, rather it's SIP or other protocol)? Many thanks, Luís

3 3

parallel forking and 30x
by Andreas Granig 21 Feb '14

21 Feb '14

Hi, What is the intended behaviour from a general point of view if you get a 1xx on one branch and a 30x on another branch for a parallel fork? >From my understanding, the 30x is a final reply, so it's kept in tm until all other branches are finished, then it enters failure-route with the highest error code (e.g. a 487 if you cancel the other branch). What I actually want is intercepting the 30x and send another INVITE to the Contact of the 30x in the failure route. This works fine if you only have one branch, but it doesn't enter failure route for 30x if you have another active branch (e.g. in ringing state). Would it be possible to create a new branch directly from the reply route, or force to enter failure route for a specific reply/branch? Andreas

3 5

Kamailio not starting
by Owais ul Haq 21 Feb '14

21 Feb '14

Hello, I have deployed Kamailio-3.1 on a fedora machine. And my database is placed on another windows 2008 server situated on the local network. Problem is when I run kamailio. I get the following error from logs. <core> [cfg.y:3416] : parse error in config gile /usr/local/kamailio-3.1/etc/kamailio//kamailio.cfg, line 311, column 31-35 : syntax error <core> [cfg.y:3416] : parse error in config gile /usr/local/kamailio-3.1/etc/kamailio//kamailio.cfg, line 311, column 31-35 : Invalid arguments Line 311 is kamailio.cfg is "modparam ("usrloc", "db_url", DBURL)" And 'DBURL' is defined as #! define DBURL "mysql://openser:5mil5My5ql@192.168.1.20/openser" Can some one please help me out in resolving this error. Secondly I am deploying my database on seperate server for the first time. I have deployed kamailio-3.1 already but with database placed on same machine. Can someone please guide what changings do I need to make in kamailio.cfg and kamctlrc file to make it work. I have made some as per my understanding. Maybe they are a cause of this error. Thanks, Regards, Owais

2 1

Information
by hfermi 21 Feb '14

21 Feb '14

Good morning I would like to receive more information about your platform web based voip. Many Thanks and Regards Henry Landi

1 0

Re: [SR-Users] FW: Regd. t_suspend() and t_continue()
by Shankar 21 Feb '14

21 Feb '14

Hi, We have found the root cause for the problem that was reported (refer below mail for details) in async module. Below is the brief description, - async_route("Resume", "1") - At time t, the async records are stored at slot 't+1' of async list. - Every second, async records from the async list are processed. - In function 'async_timer_exec', slot = ticks % ASYNC_RING_SIZE; since slot is based on the ticks and if the previous invocation of 'async_timer_exec' did not finish in time, the next tick is missed. Subsequent call to 'async_timer_exec' will be with t+2. Here slot t+1 will be missed and the same will be processed only during next cycle. That means the async records will be stored for the next 100 seconds before they are actually processed. TM module will drop those timed out transactions and hence the error "t_continue: transaction not found" Now we have modified the code in such a way that slot is incremented sequentially irrespective of the ticks that is sent to the function 'async_timer_exec' function. This way we do not see any call failures and all our load runs are successful. If someone is interested we can share the code as well. Regards, Shankar From: Shankar [mailto:shankar.rk@plintron.com] Sent: Thursday, January 23, 2014 12:18 PM To: 'Jason Penton'; 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List' Subject: RE: [SR-Users] FW: Regd. t_suspend() and t_continue() Hi, >From our repeated load tests what we can conclude irrespective of the number of simultaneous calls, there is always this error "t_continue: transaction not found" occurring. If I run say 20 cps, then after running 5000 calls, we observe exactly 20 calls failing with the above error. We doubt that there is something happening during a particular point in time (for a second) which impacts the saving of those new transactions into shared memory. For 10 cps run, we observe exactly 10 call failures. We repeated with different cps and found that error is exactly equal to the cps being run. Any configuration we are missing. Anyone can help? Regards, Shankar From: Shankar [mailto:shankar.rk@plintron.com] Sent: Tuesday, January 21, 2014 3:09 PM To: 'Jason Penton'; 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List' Subject: FW: [SR-Users] FW: Regd. t_suspend() and t_continue() Hi Jason, Below is our config, route[LOCATION] { if(is_method("INVITE")) { if(!route(FROMCSCF)) { setflag(FLT_ACC); setflag(FLT_ACCFAILED); dlg_manage(); dlg_setflag("4"); async_route("RESUME", "1"); exit; } } } route[RESUME] { route(TO_LOCATION); // here t_relay to REGISTRAR is done for user lookup. exit; } Regards, Shankar Date: Tue, 21 Jan 2014 11:14:21 +0200 From: Jason Penton <jason.penton(a)smilecoms.com> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] FW: Regd. t_suspend() and t_continue() Message-ID: <CAE=KcrghqJHgnGDxqS1fYvUzM=HqRAcKWfEAsNJjm8xUDCq68w(a)mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" We use it heavily, but not using the async module - we use it directly from the IMS code. Can you please provide your config (or a relevant snippet) file so I can see what exactly you are testing/trying to do Cheers jason From: Shankar [mailto:shankar.rk@plintron.com] Sent: Tuesday, January 21, 2014 2:25 PM To: 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List' Subject: RE: [SR-Users] FW: Regd. t_suspend() and t_continue() Hi, Anyone who had used t_suspend() and t_continue() can share the performance details? I tried async module with one sec sleep time. I tried only 5 calls per second but still it was not successful. After sometime I see below logs, Jan 21 13:51:55 PLT-RA-RD-W167A PCscf[16520]: ERROR: tm [t_suspend.c:128]: t_continue(): ERROR: t_continue: transaction not found Jan 21 13:52:49 PLT-RA-RD-W167A last message repeated 15 times Jan 21 13:59:38 PLT-RA-RD-W167A last message repeated 12 times Jan 21 14:13:03 PLT-RA-RD-W167A last message repeated 5 times Any configuration changes can help here? Regards, Shankar From: Shankar [mailto:shankar.rk@plintron.com] Sent: Wednesday, January 15, 2014 1:26 PM To: 'Jason Penton' Cc: 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List' Subject: RE: [SR-Users] FW: Regd. t_suspend() and t_continue() Hi Jason, I am using 4.0.2 Regards, Shankar From: Jason Penton [mailto:jason.penton@smilecoms.com] Sent: Wednesday, January 15, 2014 1:21 PM To: Shankar Cc: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List Subject: Re: [SR-Users] FW: Regd. t_suspend() and t_continue() Hi Shankar, What version of Kamailio are you running? Kamailio -V Cheers Jason On Wed, Jan 15, 2014 at 6:58 AM, Shankar <shankar.rk(a)plintron.com> wrote: Hi Jason, Please find below my response inline, I have some questions for you as we have used suspend/continue quite a lot in the IMS code and don't have any leaks. Firstly, why are you using pkg_mem for your hash_id and label? Remember that you will be in 2 different processes in the suspend and continue portions of the code... so pkg_mem will not work - you should use shm_mem instead. [Shankar] We use pkg_mem because we are invoking t_continue from the same process ( using thread ). Secondly, how are you using top to tell that you have a leak? Kamailio's memory is internally managed. [Shankar] After running for say 20minutes, we get out of shared memory error. Also in top output we observed incremental increase in the shared usage of shared memory for the process. Cheers Jason On Mon, Jan 13, 2014 at 1:29 PM, Shankar <shankar.rk(a)plintron.com> wrote: > Re-sending without the attachment. > > > > *From:* Shankar [mailto:shankar.rk@plintron.com] > *Sent:* Monday, January 13, 2014 4:57 PM > *To:* 'sr-users(a)lists.sip-router.org' > *Subject:* Regd. t_suspend() and t_continue() > > > > Hi, > > > > We are trying out the t_suspend() and t_continue() in our test setup. > We are facing memory leak ( both shm and pkg as per top command results). > > > > Please find below the scenario, > > > > 1) Do a t_newtran() > > 2) Allocate pkg memory for hashid and label. > > 3) Call t_suspend() > > 4) Do t_continue() when async result is available > > 5) De-allocate pkg memory reserved for hashid and label > > 6) Do a t_relay() which forwards the sip message to another sip node. > > > > In the step (6) above, we see t_newtran() allocates one more time > shared memory for the same transaction. > > > > We tried t_release() after step (4) to release the transaction as > t_relay() anyways allocates new shared memory. Nothing helped. > > > > Please let me know what are the logs you would require to debug the same. > I am attaching syslog for this run. > > > > Regards, > > Shankar -- Jason Penton Senior Manager: Applications and Services Smile Communications Pty (Ltd) Mobile: +27 (0) 83 283 7000 Skype: jason.barry.penton <mailto:name.surname@smilecoms.com> jason.penton(a)smilecoms.com <http://www.smilecoms.com/> www.smilecoms.com <http://196.33.227.129/~smlcoms/sigs/pty/images/smile_signature_07_09.jpg> This email is subject to the disclaimer of Smile Communications at http://www.smilecoms.com/disclaimer

1 0

Call Control for Credit and Rate
by Raj Roy Ghandhi 21 Feb '14

21 Feb '14

Hi, I want to implement the call control by 1. Validating the CLI and get the credit from DB 2. Check the Rate for the dialed number 3. set the call timeout for the call duration. ( credit/rate) 4. pass the call to remote PSTN GW Please guide me on this. Regards, Roy.

3 2

Debian packages not updated since august 2013?
by Johan Wilfer 21 Feb '14

21 Feb '14

Hi, Looking at http://www.kamailio.org/wiki/packages/debs the suggestion is to use http://deb.kamailio.org/ for debian packages. But the last version of kamailio 4.0 here is 4.0.3, and 4.1 doesn't exist. The most files are last modified in 2013-08.. Here for example: http://deb.kamailio.org/kamailio/dists/wheezy/main/binary-amd64/ Is there a new server to get the deb-packages from or should I build from source from now on? /Johan

3 3

Performance impact with AVP and VAR's
by Jijo 21 Feb '14

21 Feb '14

Hi All, I have around 300 AVP's and quite amount of VAR's are used in the config file? Does that impact performance?, If so how can i improve it? Thanks Jijo

4 5

Re: [SR-Users] Issue with Async module
by Daniel-Constantin Mierla 20 Feb '14

20 Feb '14

Hello, thanks for reporting, I pushed a slightly different patch in master, 4.1 and 4.0 branches -- free is done even when ai->act is null. Cheers, Daniel On 20/02/14 15:13, Surendra wrote: > Hi Daniel, > > > I am using async module to continue the transaction in > asynchronously. But I am observing a leak in shared memory. Then i checked > the code, we are not freeing the async_item_t after calling t_continue. > Same this has checked with MEMDBG=1. > > Original code: > while(1) { > lock_get(&_async_list_head->ring[slot].lock); > ai = _async_list_head->ring[slot].lstart; > if(ai!=NULL) > _async_list_head->ring[slot].lstart = ai->next; > lock_release(&_async_list_head->ring[slot].lock); > > if(ai==NULL) > break; > if(ai->act!=NULL) > tmb.t_continue(ai->tindex, ai->tlabel, ai->act); > } > > > Changed code: > > while(1) { > lock_get(&_async_list_head->ring[slot].lock); > ai = _async_list_head->ring[slot].lstart; > if(ai!=NULL) > _async_list_head->ring[slot].lstart = ai->next; > lock_release(&_async_list_head->ring[slot].lock); > > if(ai==NULL) > break; > if(ai->act!=NULL) > { > tmb.t_continue(ai->tindex, ai->tlabel, ai->act); > shm_free(ai); /* Added by surendra */ > } > } > > Please correct this if I am wrong. > > > Regards > @surendra > > > > > _______________________________________________ > 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

1 0

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