sr-users May 2013

sr-users@lists.kamailio.org

95 participants
156 discussions

by Daniel W. Graham

While I have seen other posts I have not found anything that describes my scenario. Kamailio sets in front of a group of asterisk servers which are used in a round robin dispatch group. All asterisk servers share a common database in which voicemail boxes are defined and messages are stored. I'm looking for a way to make message waiting work in this example. I have tried creating a view with basic peer information for asterisk to use for message notification, however asterisk does not 'see' these peers until a call is made to the specific user and the peer info is retrieved from the database, which can delay notifications severely rendering them useless. I also want to avoid sending a copy of the registrations to asterisk servers, as the purpose of the distributed system is to eliminate the large number of sip registrations asterisk needs to manage. I have also seen a couple of script methods that look at the voicemail directory structure for file changes and trigger an application like sipsak to generate the notifications and let kamailio relay them. From what I see, the consensus is not to go this route. In my case the script would need to look at the database level, and maybe some sort of database trigger could be used. What are others doing / what do others recommend as the best way to handle message waiting with a situation where kamailio sets in front of a voicemail server? Any input is appreciated. Dan-

11 years

Call Forwarding

by Lucas Girard

Hi guys, I need to implement a call forwarding (blind call forward) in a kamailio. Do you know if this is possible? There is a way that allow the subscriber to configure it's own forwarding ? Thanks in advance, Lucas

11 years, 1 month

[PATCH] Memory corruption using s.substr transformation

by Martin Mikkelsen

Hello, I have a kamailio 3.3.4 server running on x86-64 Linux, and I have a script which looks something like this: $var(gwruri) = $rU; if ($(var(gwruri){s.substr,0,1}) == "+") { $var(gwruri) = $(var(gwruri){s.substr,1,0}); } When this script is run with say $rU = "+0000009724" (real number removed). $var(gwruri) should contain the same but without the +, this is not the case, $var(gwruri) ends up being "0000007724". This smells memory corruption. When run under valgrind, valgrind generates this error when that code is run: ==1206== Source and destination overlap in strncpy(0x55e3b2a, 0x55e3b2b, 10) ==1206== at 0x4C25ACF: strncpy (mc_replace_strmem.c:339) ==1206== by 0x2A28BB92: set_var_value (pv_svar.c:122) ==1206== by 0x2A2800D1: pv_set_scriptvar (pv_core.c:1683) ==1206== by 0x45E8F8: lval_assign (lvalue.c:353) ==1206== by 0x416A78: do_action (action.c:1524) ==1206== by 0x41E465: run_actions (action.c:1644) ==1206== by 0x4177FD: do_action (action.c:1136) ==1206== by 0x41E465: run_actions (action.c:1644) ==1206== by 0x4177FD: do_action (action.c:1136) ==1206== by 0x41E465: run_actions (action.c:1644) ==1206== by 0x419C67: do_action (action.c:1140) ==1206== by 0x41E465: run_actions (action.c:1644) ==1206== I am not very familiar with the kamailio source code, but as far as I can tell this happens because the strncpy() in set_var_value() in pv_svar.c copies the string directly within the same overlapping memory area. The same memory area is used because the tr_eval_string() function and the TR_S_SUBSTR-code in pv_trans.c reuses the input buffer and just increments the pointer and since the source pvar is the same as the destination pvar the strncpy() ends up copying between overlapping memory area. I am not sure what the best fix would be for that, but I have attached a patch which copies the string in the TR_S_SUBST code to _tr_buffer and returns that buffer instead like a lot of the other transformations in that function does. If this fix is correct some other transformation functions probably needs to be corrected as well. -- Martin Mikkelsen, Zisson AS

11 years, 1 month

Problems with Siremis Register Page

by Alexander Albert

Hi there, i hope you can help me, i am having trouble with the register page of siremis, strange thing is, i can open the register page, but only after i have logged my self in, if not, it redirects me directly to the login page.. how can i setup the register page to be accessible without the need of logging in ? Also i am getting the info after trying to register, that "Public registration is not enabled!" but i have set siremis/modules/ser/config/common.Main.php to $cfg_siremis_public_registrations = true; What is the problem, please help. Greetings, Alex

11 years, 1 month

if (t_check_status("486|408"))

by hiro

i'm trying to use the example kamailio.cfg to route to voicemail server on busy or decline. Only thing I did was adding decline code to t_check_status("486|408"), enabling the preprocessor variable for voicemail and changing the voicemail host and port to my voicemail server. No requests arrive on my voicemail server and the dial tone keeps ringing even if phone is busy and when I decline on the receiving phone there's a new INVITE sent to the phone directly afterwards. Am I doing something wrong here?

11 years, 1 month

Re: [SR-Users] Freepbx 2.11.0rc1 with Asterisk 11.3.0 and Kamailio 4.0.1

by Michael Leuker

Thank you very much for sharing your insights, Barry! I am facing the same problem that Trevor described: Things are working just fine on their own, but as soon as FreePBX comes into play, calling extensions becomes impossible because of the different tables used. Removing the password from FreePBX (and setting the Kamailio IP in the ACL field) seems to mitigate the issue somewhat, but even though the extension shows as registered in FreePBX, it always shows as busy: chan_sip.c:23237 handle_response_invite: Failed to authenticate on INVITE to '"xxxxxxxx" <sip:xxxxxxxx@198.23.139.21>;tag=as72a4117a' -- SIP/1001-00000006 is circuit-busy I doubt that I can make the necessary modifications even with your hints, but would be willing to pay for your (or anybody's) time solving the matter in a way that (a) leaves the FreePBX installation untouched as far as possible and (b) is easy to apply to subsequent versions. Ideally, it would be a slight modification of Daniel's excellent tutorial at http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb, allowing to create extensions in FreePBX. Anybody willing to give it a shot? Just let me know what you would charge and we can talk about it.

11 years, 1 month

Kamailio and Siremis

by Isaac A. McDonald

Hello All, I'm a bit confused as to how Siremis interacts with Kamailio. From what I understand, the dial-plan is done in the kamailio.cfg with regex statements and requires a restart of the Kamailio process to take effect. How then does Siremis change the dial-plan in Kamailio without writing re-writing the config? I'm assuming Siremis is merely writing to a MySQL database and some additional code is required in the kamailio.cfg in order to be able to use the dial-plan functionality of Siremis? What must I change in the kamailio.cfg to be able to use Siremis for dial-plan modification?

11 years, 1 month

is_uri_host_local() maybe not considering IP:PORT

by DanB

Guys, I was wondering if I am doing something wrong or domain module not considering properly local ips within is_uri_host_local(). Although I have domain loaded with register_myself on 1, the request going to my IP is simply not matching local domain (listening on 127.0.0.1 port 5070). Bellow the trace of such request: """ # U 2013/05/30 12:35:56.436485 10.10.10.21:5060 -> 127.0.0.1:5070 OPTIONS sip:127.0.0.1:5070 SIP/2.0. Via: SIP/2.0/UDP 10.10.10.21;branch=z9hG4bK4866.b4f45983000000000000000000000000.0. To: <sip:127.0.0.1:5070>. From: <sip:ep@iec.itsyscom.com>;tag=ae9b2706b606c3acb0ebe4f1c8f81cee-f20d. CSeq: 10 OPTIONS. Call-ID: 467de807489c4482-3002(a)10.10.10.21. Max-Forwards: 70. Content-Length: 0. User-Agent: iClass4-EP 4.0.0. . # U 2013/05/30 12:35:56.440410 127.0.0.1:5070 -> 10.10.10.21:5060 SIP/2.0 484 Address Incomplete. Via: SIP/2.0/UDP 10.10.10.21;branch=z9hG4bK4866.b4f45983000000000000000000000000.0. To: <sip:127.0.0.1:5070>;tag=46a6e639fa023622ac1ba4fea686e961.d61e. From: <sip:ep@iec.itsyscom.com>;tag=ae9b2706b606c3acb0ebe4f1c8f81cee-f20d. CSeq: 10 OPTIONS. Call-ID: 467de807489c4482-3002(a)10.10.10.21. Server: iClass4-AP 4.0.0. Content-Length: 0. """ My script looks something like bellow, so Address Incomplete should never be reached: """ #!define LISTEN_IP 127.0.0.1 #!define LISTEN_PORT 5070 ... listen=LISTEN_IP port=LISTEN_PORT auto_aliases=yes ... # ----- domain params ----- modparam("domain", "db_url", DBURL) modparam("domain", "register_myself", 1) ... if (is_method("OPTIONS") && is_uri_host_local()) { options_reply(); exit; } ... if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } """ I am running on git master with the test patch Daniel did few days back for me. Thanks in advance for any tip! DanB

11 years, 1 month

Faild to add Subscription via WebRTC2SIP websocket GW

by Raj Roy Ghandhi

Hi Friends, I am using Kamailio 4.0 SIP Proxy GW + WebRTC2SIP 2 web socket GW to maintain the presence status of SIPMP5 clients. When I subscriber for presence statue of contact I get the following error and no record get inserted into the "active_wathcers" table. 5(7435) WARNING: <core> [msg_translator.c:2499]: TCP/TLS connection (id: 0) for WebSocket could not be found 5(7435) ERROR: tm [t_msgbuilder.c:1367]: assemble_via: via building failed 5(7435) ERROR: tm [t_msgbuilder.c:1540]: build_uac_req(): Error while assembling Via 5(7435) ERROR: tm [uac.c:338]: t_uac: Error while building message 5(7435) ERROR: presence [notify.c:1591]: in function tmb.t_request_within 5(7435) ERROR: presence [subscribe.c:1613]: send Notify not successful 5(7435) ERROR: presence [subscribe.c:2129]: in function handle_expired_record 5(7435) ERROR: presence [subscribe.c:464]: Failed to delete subscription from memory Here is the SIPML5 console log for that perticular area State machine: tsip_dialog_generic_Any_2_InProgress_X_oSubscribe tsk_utils.js?svn=20:110 SEND: SUBSCRIBE sip:1005@sip.mysip.com SIP/2.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKLmNGlDNlChMuoGqF9qFzMS7dOk5h0E5T;rport From: <sip:1004@sip.mysip.com>;tag=FRInEZSEyF7xFFuCHMK3 To: <sip:1005@sip.mysip.com> Contact: "undefined"<sip:1004@df7jal23ls0d.invalid ;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;+audio;language="en,fr" Call-ID: 1267da78-fbd1-cdcd-ee45-5cde2d2650bf CSeq: 27396 SUBSCRIBE Expires: 200 Content-Length: 0 Max-Forwards: 70 Event: presence Accept: application/pidf+xml tsk_utils.js?svn=20:110 session event = type = connecting - description = Connecting... test_api.html:51 __tsip_transport_ws_onmessage tsk_utils.js?svn=20:110 recv=SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/WS df7jal23ls0d.invalid;rport=20124;received=192.168.1.55;branch=z9hG4bKLmNGlDNlChMuoGqF9qFzMS7dOk5h0E5T From: <sip:1004@sip.mysip.com>;tag=FRInEZSEyF7xFFuCHMK3 To: <sip:1005@sip.mysip.com>;tag=b01221cac0ed2edd8409481b9ce2d72c.ac8c Call-ID: 1267da78-fbd1-cdcd-ee45-5cde2d2650bf CSeq: 27396 SUBSCRIBE Content-Length: 0 Proxy-Authenticate: Digest realm="sip.mysip.com ",nonce="UaaBx1GmgJtqCoZzmu9Tskw+AmewKCiC",stale=FALSE Server: kamailio (4.0.1 (x86_64/linux)) tsk_utils.js?svn=20:110 State machine: tsip_dialog_generic_InProgress_2_InProgress_X_401_407_421_494 tsk_utils.js?svn=20:110 SEND: SUBSCRIBE sip:1005@sip.mysip.com SIP/2.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKpPobVdaUhPHunxtfAuQFOlkfEeGs2ksE;rport From: <sip:1004@sip.mysip.com>;tag=FRInEZSEyF7xFFuCHMK3 To: <sip:1005@sip.mysip.com> Contact: "undefined"<sip:1004@df7jal23ls0d.invalid ;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;+audio;language="en,fr" Call-ID: 1267da78-fbd1-cdcd-ee45-5cde2d2650bf CSeq: 27397 SUBSCRIBE Expires: 200 Content-Length: 0 Max-Forwards: 70 Proxy-Authorization: Digest username="1004",realm="sip.mysip.com ",nonce="UaaBx1GmgJtqCoZzmu9Tskw+AmewKCiC",uri="sip:1005@sip.mysip.com ",response="223e983b8e2de2b04dbcd2ac9fc0b17d",algorithm=MD5 Event: presence Accept: application/pidf+xml tsk_utils.js?svn=20:110 __tsip_transport_ws_onmessage tsk_utils.js?svn=20:110 recv=SIP/2.0 202 OK Via: SIP/2.0/WS df7jal23ls0d.invalid;rport=20124;received=192.168.1.55;branch=z9hG4bKpPobVdaUhPHunxtfAuQFOlkfEeGs2ksE From: <sip:1004@sip.mysip.com>;tag=FRInEZSEyF7xFFuCHMK3 To: <sip:1005@sip.mysip.com>;tag=2a2b07acfe8894f890540e2e919abb3e-1594 Contact: <sip:192.168.1.15:8080;transport=ws> Call-ID: 1267da78-fbd1-cdcd-ee45-5cde2d2650bf CSeq: 27397 SUBSCRIBE Expires: 200 Content-Length: 0 Server: kamailio (4.0.1 (x86_64/linux)) tsk_utils.js?svn=20:110 State machine: tsip_dialog_generic_InProgress_2_Connected_X_2xx tsk_utils.js?svn=20:110 __tsip_transport_ws_onmessage tsk_utils.js?svn=20:110 recv=NOTIFY sip:1004@192.168.1.55:20124;rtcweb-breaker=no;click2call=no;transport=ws SIP/2.0 Via: SIP/2.0/WS 192.168.1.15:8080 ;branch=z9hG4bK4b8a.ab27a051000000000000000000000000.0 From: <sip:1005@sip.mysip.com>;tag=2a2b07acfe8894f890540e2e919abb3e-1594 To: <sip:1004@sip.mysip.com>;tag=FRInEZSEyF7xFFuCHMK3 Contact: <sip:192.168.1.15:8080;transport=ws> Call-ID: 1267da78-fbd1-cdcd-ee45-5cde2d2650bf CSeq: 2 NOTIFY Content-Length: 0 User-Agent: kamailio (4.0.1 (x86_64/linux)) Max-Forwards: 70 Event: presence Subscription-State: active;expires=200 tsk_utils.js?svn=20:110 State machine: tsip_dialog_generic_Any_2_Any_X_iNotify tsk_utils.js?svn=20:110 session event = type = i_ao_request - description = OK test_api.html:51 session event = type = connected - description = Connected test_api.html:51 SEND: SIP/2.0 200 OK Via: SIP/2.0/WS 192.168.1.15:8080 ;branch=z9hG4bK4b8a.ab27a051000000000000000000000000.0 From: <sip:1005@sip.mysip.com>;tag=2a2b07acfe8894f890540e2e919abb3e-1594 To: <sip:1004@sip.mysip.com>;tag=FRInEZSEyF7xFFuCHMK3 Contact: <sip:1004@df7jal23ls0d.invalid;transport=ws> Call-ID: 1267da78-fbd1-cdcd-ee45-5cde2d2650bf CSeq: 2 NOTIFY Content-Length: 0 tsk_utils.js?svn=20:110 session event = type = i_notify - description = Incoming NOTIFY test_api.html:51 NOTIFY content = test_api.html:72 NOTIFY content-type = null Please guide me on this. Best Regards, Roy.

11 years, 1 month

Does Kamailio check for CSeq, Call-ID when unregister ?

by Khoa Pham

Hi, When I want to unregister, I have 2 ways 1. Append ;expires=0 in Contact field 2. Add another Expires : 0 header with CSeq, Call-ID not the same with when i register Does Kamailio check for the saem Cseq, Call-ID to allow unregistration ? -- Khoa Pham HCMC University of Science www.fantageek.com

11 years, 1 month

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sr-users May 2013