- sr-users - kamailio.org

where does iceserver configuration get set
by jaflong jaflong 17 Mar '14

17 Mar '14

Hi List Is ice settings initiaised on the browser side or Kamailio websocket side? Where are the setting parameters created for a=candidate In the debug I get such logs as a=candidate:3164634682 1 tcp 1509957375 xxx.xxx.xxx.xxx 0 typ host generation 0 a=candidate:3164634682 2 tcp 1509957375 xxx.xxx.xxx.xxx 0 typ host generation 0 a=ice-options:google-ice I am connecting with with jssip like the snippet below. There is nothing set for stun/turn. So how then does it use a=ice-options:google-ice. Does it set google as default var configuration = { .... .... stun_servers: '', turn_servers: null, .... .... }; // User Agent instance var coolPhone = new JsSIP.UA(configuration); Does anyone know are these the only options is relation to ice onfig, stun_servers: turn_servers: null,

1 0

evapi module question
by Alex Balashov 17 Mar '14

17 Mar '14

Hi, Does evapi_relay() have to emit JSON blobs? Or can it be used to emit generic strings? Thanks! -- 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/

2 1

Endpoint registration tracking and reporting
by Nick Raper 17 Mar '14

17 Mar '14

Hi all, What is the best way to monitor endpoint registrations? When one of our clients has a link failure or for some reason they lose their SIP registration with Kamailio, we'd like to be able to see this so that we can respond and get the trunk re-registered. We do monitor the physical links but endpoints, (usually Asterisk based), often don't re-register after connectivity is restored. I was thinking of a script that dumps the 'location' table every 10 minutes and compares it to the previous 'location' table dump and reports on the changes. Is there a better way of doing this? Regards, [Description: cid:image001.jpg@01C8D7A0.651BD080] Nick Raper 4th Dimension Technology | South Africa Office: +27 21 506 6260 | 24 hour helpline: +27 21 506 6260 | Fax: +27 86 610 5792 Email: nick(a)4dt.co.za<mailto:nick@4dt.co.za> | Web: www.4dt.co.za<http://www.4dt.co.za/> P Please consider the environment before printing this email.

3 4

Softphone cant connect
by aawaise 17 Mar '14

17 Mar '14

Hello, I have deployed a kamailio server. Now I am trying to set up a voip call between two soft phones (EyeBeam) located on local network. I am using "Outbound proxy option" on softphone. It shows "peer to peer mode enabled". Settings are shown in figure attached <http://sip-router.1086192.n5.nabble.com/file/n125931/1.jpg> But when I make a call. It doesnot connect. Using wireshark on kamailio server, I found that in response to "SIP : Request" packet, it sends ICMP packet "Destination Unreachable: Port Unreachable". Firewall is disabled on all concerned systems. What can be the possible reason. Any one will help please. Regards, Awais -- View this message in context: http://sip-router.1086192.n5.nabble.com/Softphone-cant-connect-tp125931.html Sent from the Users mailing list archive at Nabble.com.

1 0

alias behaving differently than droute
by Pete Ashdown 15 Mar '14

15 Mar '14

I presumed that aliases could be used for quickie destination routing, but for some reason my RTP isn't connecting up after. If I put the same alias into droute, it works. Am I understanding aliases wrong?

3 3

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

2 7

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

1 1

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.

5 6

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.

4 3

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.

2 1

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.....

2 2

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/

3 8

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

2 6

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

3 3

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

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