sr-users November 2013

sr-users@lists.kamailio.org

100 participants
160 discussions

Hash Table capacity
by Ricardo Martinez 21 Oct '14

21 Oct '14

Hello list. I have a question about the HTABLE module. Suppose that I have a mysql database with 2 million of records. Is possible to load this table as a hash table compatible with Kamailio!? Hope someone can help me. Thanks, Ricardo Martinez.-

6 15

kamailio and app_python
by Victor Seva 11 Jun '14

11 Jun '14

Hi, is there any way to return values from a call to python_exec function orther than the default returned integer? Following code generates this error: Oct 1 09:08:23 zskio /usr/sbin/kamailio[21789]: ERROR: app_python [python_support.c:38]: python_exec2: Unhandled exception in the Python code: Oct 1 09:08:23 zskio /usr/sbin/kamailio[21789]: ERROR: app_python [python_support.c:78]: #011RuntimeError: no such function if (!python_exec("get_provider", $(var(clean_uri){s.prefixes}))) { xlog("L_ERR","Can not choose a provider for $rU\n"); sl_send_reply("503","Can not choose a provider for $rU"); exit; } def get_provider(self, msg, param): # snipped code... result = self.util.get_provider(prefixes, position, mode) if result is not None: LM_ERR("mode: %s position: %d result: %s" % (mode, position, result)) # this raises error msg.call_function('rewritehost', result['provider']['ip']) return 1 return -1 I tried to use headers as transport but this simple code does not work: Oct 1 09:45:31 zskio /usr/sbin/kamailio[22303]: ERROR: <script>: No header if (!is_present_hf("X-Test-header")) { xlog("L_ERROR", "No header\n"); } else { xlog("L_INFO", "Header added\n"); } def test_set_header(self, msg): msg.call_function('append_hf', 'X-Test-header: test\r\n') return 1 Thanks in advance.

4 6

TLS and SIP
by Fabian Borot 23 May '14

23 May '14

Hi I am using Kamailio 4.0.1 in front of an asterisk servers farm to handle TLS with our clients and providers. The idea is to have kamailio "talking" SIP/UDP/5060 and TLS/TCP/5061 with the customers and providers and regular SIP/UDP/5060 with our internal asterisk servers. So far at least for the customers it looks like it can work. But I have a problem, when the call is established and the called person hangs up, the BYE from the called person to the calling person is ignored. Only when the calling person hangs up first the call is terminated properly. This is what I have been able to see: 1- Customer starts the TLS handshake/connection. 2- Kamailio authenticate it, then routes the call to the asterisk server using regular SIP/UDP/5060 but I see that it is inserting 2 Record Routes in the INVITE: Record-Route: <sip:192.168.1.58;r2=on;lr=on> Record-Route: <sip:192.168.1.58:5061;transport=tls;r2=on;lr=on> 3- The Contact on that INVITE to the asterisk also comes like this: Contact: <sip:94167032@172.31.196.21:53325;transport=tls> 4- The ACK sent to the asterisk once it accepts the call (200 OK) also has those 2 Record-Routes: Record-Route: <sip:192.168.1.58;r2=on;lr=on> Record-Route: <sip:192.168.1.58:5061;transport=tls;r2=on;lr=on> 5- The call is established, once the called person decides to hang up the BYE looks like this: BYE sip:94167032@172.31.196.21:53325;transport=tls SIP/2.0 Via: SIP/2.0/UDP 192.168.1.59:5060;branch=z9hG4bK40fa1c23;rport Route: <sip:192.168.1.58;r2=on;lr=on>,<sip:192.168.1.58:5061;transport=tls;r2=on;lr=on> Max-Forwards: 70 From: <sip:3030500@1.2.3.4>;tag=as37953869 To: "kamailio" <sip:kamailio@1.2.3.4>;tag=788cd7c892df40f3b1967112395e2ca4 Call-ID: f9fe65daf1074219be26cb0c224339f1 CSeq: 102 BYE User-Agent: Asterisk PBX 11.3.0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 My kamailio TLS config is shown below: enable_tls=yes loadmodule "tls.so" # ----- tls params ----- modparam("tls", "config", "/usr/local/kamailio-4.1//etc/kamailio/tls.cfg") modparam("tls", "private_key", "./privkey.pem") modparam("tls", "certificate", "./kamailio1_cert.pem") modparam("tls", "ca_list", "./calist.pem") modparam("tls", "verify_certificate", 1) modparam("tls", "require_certificate", 1) The TLS client that I am using is called Blink.At this point I don't know whether kamailio is sending the BYE using TLS to the customer and waiting for the 200 OK from the customer or whether kamailio does not like something in the BYE and that is why is ignoring it. I see some encrypted packets from kamailio to the client but I don't know what is inside. Any help would be very appreciated. thank you fabian

7 14

Double attempt to record-route
by Juha Heinanen 27 Apr '14

27 Apr '14

I tried to call record_route_preset() in branch route in order to add contact specific RR headers when request is forked to more than one contact. What happened was that I got error message: Sep 26 17:03:47 wheezy2 /usr/sbin/sip-proxy[2436]: ERROR: rr [rr_mod.c:256]: Duble attempt to record-route Is it really so that record routing is not a branch specific thing? Is there a way to solve this problem? -- Juha

3 4

PERMISSIONS module issue
by Samuel Ware 15 Apr '14

15 Apr '14

I having issue updating my allow list for the PERMISSIONS module. I added an address to the ADDRESS table. I have tried to do a service restart, kamctl address reload, and kamcmd permissions.addressReload. The kamctl address show displays the new address; however kamcmd permissions.addressDump does not neither does kamcmd permissions.subnetDump. The messages from this new address return a false to the !allow_source_address("1”) command in my routing logic. I am wondering if this is a bug or I am doing something wrong. I am on the most recent GIT version to the best of my knowledge. Sam

6 6

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

Error with ACK from SIPML5
by Jayesh Nambiar 14 Feb '14

14 Feb '14

Hello All, I have a setup which is SIPML5 -> Kamailio(Websocket) -> Freeswitch. All I am trying to do is initiate a call from SIPML5 and Play a Music file from freeswitch. This works, but the call hangs up after 30 seconds due to ACK timeout from from the SIPML5. What I can see is the SIPML5 from Chrome does send an ACK on websocket, but I see the following error in kamailio syslog: *via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found* *ERROR: <core> [msg_translator.c:1725]: build_req_buf_from_sip_req(): could not create Via header* *ERROR: <core> [forward.c:607]: forward_request(): ERROR: forward_request: building failed* *ERROR: sl [sl_funcs.c:371]: sl_reply_error(): ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL)* Can someone help me understand what I am missing here and why kamailio is not able to create a VIA header to forward this request towards freeswitch. I have built a simple config file by following https://gist.github.com/jesusprubio/4066845 as reference. Any help in understanding this is really appreciated. Thanks, --- Jayesh

4 11

Asterisk re-INVITE race condition, error 500.
by David K 04 Feb '14

04 Feb '14

Hello all, So I have three machines, we don't care about audio for this problem, so everything I mention here is SIP related. Freeswitch <--> Kamailio 3.3.2 <--> Asterisk 1. Asterisk sends an INVITE to Freeswitch through the Kamailio proxy. 2. Kamailio replies 100 Trying and forwards to Freeswitch 3. Freeswitch replies 100 Trying 4. Freeswitch replies 180 Ringing to Kamailio 5. Kamailio routes the answer to Asterisk 6. Freeswitch replies 200 OK to Kamailio 7. Kamailio replies 200 OK to Asterisk 8. Asterisk replies ACK to Kamailio 9. Asterisk sends a re-INVITE to Freeswitch through Kamailio 10. Kamailio routes the re-INVITE to freeswitch 11. Kamailio routes the ACK to freeswitch. 12. Freeswitch replies 500 Server error because it got a re-INVITE before the ACK. So, my problem is that Kamailio seems to process my re-INVITE more quickly than the ACK. So Freeswitch replies an error because it got the re-INVITE before the ACK. So my "solution" is to add a usleep(20); for re-INVITEs on Kamailio, but I think this is a lousy solution. Has anyone here had to deal with problems where Kamailio routes a re-INVITE faster than an ACK causing endpoints to return error messages? Has anyone had to deal with a similar issue? Thanks, David

6 11

Implementing the Presence module(Kamailio) to work with websockets (sipML5)
by אורן אברהם 15 Jan '14

15 Jan '14

Dear List, I have some strange error when using sipML5 client (over websockets) with Kamailio 4.0 server when sending a SUBSCRIBE request. here is my log print: Apr 18 02:55:21 oren-ubuntu /usr/sbin/kamailio[26476]: WARNING: <core> [msg_translator.c:2499]: TCP/TLS connection (id: 0) for WebSocket could not be found (tcp:10.0.0.4:8080) Apr 18 02:55:21 oren-ubuntu /usr/sbin/kamailio[26476]: ERROR: tm [t_msgbuilder.c:1367]: assemble_via: via building failed Apr 18 02:55:21 oren-ubuntu /usr/sbin/kamailio[26476]: ERROR: tm [t_msgbuilder.c:1540]: build_uac_req(): Error while assembling Via Apr 18 02:55:21 oren-ubuntu /usr/sbin/kamailio[26476]: ERROR: tm [uac.c:338]: t_uac: Error while building message Apr 18 02:55:21 oren-ubuntu /usr/sbin/kamailio[26476]: ERROR: presence [notify.c:1591]: in function tmb.t_request_within Apr 18 02:55:21 oren-ubuntu /usr/sbin/kamailio[26476]: ERROR: presence [notify.c:1678]: sending Notify not successful Apr 18 02:55:21 oren-ubuntu /usr/sbin/kamailio[26476]: ERROR: presence [subscribe.c:678]: Could not send notify Apr 18 02:55:21 oren-ubuntu /usr/sbin/kamailio[26476]: ERROR: presence [subscribe.c:713]: occured Apr 18 02:55:21 oren-ubuntu /usr/sbin/kamailio[26476]: ERROR: presence [subscribe.c:994]: in update_subscription Could anyone help here ? I think it is only some configuration problem, but i'm not sure how to solve it.

3 6

t_reply in failure route with dialog module
by Efelin Novak 03 Jan '14

03 Jan '14

Hi, when I use t_reply("505", "Error"); in my failure route, the response is not forwarded and following is written into a log: kamailio[26216]: WARNING: tm [t_lookup.c:1559]: t_unref(): WARNING: script writer didn't release transaction plus next line is written exactly 416000 times into a log afterwards: kamailio[32685]: CRITICAL: dialog [dlg_hash.c:794]: log_next_state_dlg(): bogus event 4 in state 5 for dlg 0xb4af6588 [2575:7017] with clid '121d44f0-6555f4c8' and tags 'd12546d053aadc68o2' '' My point is to change the incoming code from users and append a Q.850 reason code. Is there any other way how to do this or a way how to fix this? I'm using Kamilio 4.0.4 on Debian 7.1 The code is as follows: failure_route[MANAGE_FAILURE] { if (t_is_canceled()) { exit; } if($T_reply_code == 408 && isflagset(10)) { xlog("Ringing timeout"); append_to_reply("Reason: Q.850;cause=28\r\n"); t_reply("505", "Error"); } }

3 10

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sr-users November 2013