March 2014 - sr-users - kamailio.org

Kamailio: MI Command: 'trusted_dump' and 'trusted_reload'
by José Luis Millán 22 Apr '15

22 Apr '15

Hi, I'm using 'allow_trusted()' from Permissions module without a problem in a Kamailio 4.0 routing logic. It does just work, but the MI command 'trusted_dump' tells me that the module is not in use: ''' server:/home/jmillan# kamctl fifo trusted_dump 500 Trusted-module not in use ''' Doing a 'trusted_reload' does neither make any query to database to retrieve the data. Other MI commands of same module do work correctly. ie: 'address_reload' or 'address_dump' Note: I have tested in a Kamailio 1.5 and the result is the same. Regards. -- José Luis Millán

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File transferring issue
by Wingsravi R 25 Mar '15

25 Mar '15

Dear Kamailio'ns, I am working on kamailio Server (V 4.0) installed on Ubuntu 12.04 system. Currently i am trying file transfer feature using IMSDroid SIP clients. file transferring is fine between two IMSDroid clients but at the receiving end (file recipient) i cont see the sender identity/ sender name. it is just displaying as 'null' (Screen shot is attached). But the full file is transferring to destination address without any problem. I have configured MSRP module related script in my kamailio configuration file (find the attachment). but i guess it is not invoking anyway when file transferring and the file transferring is negotiating client-to-client basis. And also find Sip capture of file transfer (ngrep based) for better understanding. What could be the problem? Why at the file recipient side it is not displaying sender's identity (instead it is 'null')? how can i resolve this issue? Anyone please help me in resolving this issue. Any help will greatly appreciate. Regards, Ravi.

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Fwd: Need a support for Siremis and kamailio server
by Gaurav Rai 14 Jan '15

14 Jan '15

Hi I am using a siremis and *kamailio * hosted server with lylix.net and they give a link for getting a support about Siremis. I need a support to how to make OUT call on PSTN with registered subscriber on Softphone. Currently we have registered user on subscriber table with their location and can make a PEER to PEER audio call and text messaging also. To make OUT call on PSTN we had create a LCR gateway and LCR rule with Prefix but still can not make a OUT call getting a error 404 can u please help me out to make a PSTN call Waiting for your reply Thanks Gaurav Rai +919716238804

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Re: [Kamailio-Users] /usr/local/sbin/kamdbctl create error
by vivi 01 Nov '14

01 Nov '14

Hi all.. I have solve the problem . Cheers, vivi From: vivi [mailto:vivi.hilton@gmail.com] Sent: Saturday, June 06, 2009 11:29 PM To: 'users(a)lists.kamailio.org' Subject: /usr/local/sbin/kamdbctl create error Hi all I have specify the wanted db type (DBENGINE=MYSQL) in the /usr/local/etc/kamailio/kamctlrc then I using "/usr/local/sbin/kamdbctl create" to create MySQL database, but I got this error: ERROR: database engine not specified, please setup one in the config script root@acer:/usr/local/src/kamailio-1.5.0/sip-server# vim /usr/local/etc/kamailio/kamctlrc root@acer:/usr/local/src/kamailio-1.5.0/sip-server# /usr/local/sbin/kamdbctl create MySQL password for root: INFO: test server charset ERROR 2002 (HY000): Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2) ERROR 2002 (HY000): Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2) Usage: grep [OPTION]... PATTERN [FILE]... Try `grep --help' for more information. /usr/local/lib/kamailio/kamctl/kamdbctl.mysql: line 112: [: =: unary operator expected INFO: creating database openser ... ERROR 2002 (HY000): Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2) ERROR: Creating core database and grant privileges failed! Cheers, vivi

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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.-

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XMLRPC problem
by Kelvin Chua 18 Aug '14

18 Aug '14

i am currently having problems on xmlrpc i am using node-xmlrpc and it fails when htable.dump returns more than 1 row. i raised this issue on the node-xmlrpc group and this is the explanation i got. Each <value> node inside <param> is supposed to only have one child node, so this looks like an invalid response. The<value> should (probably, i don't know your use case) contain an <array> with <data> wrapped around the structs. Seehttp://xmlrpc.scripting.com/spec.html does this mean kamailio is not following standards? here is the working response: <methodResponse> <params> <param> <value><struct><member><name>entry</name><value><int>6</int></value></member><member><name>size</name><value><int>1</int></value></member><member><name>slot</name><value><struct><member><name>item</name><value><struct><member><name>name</name><value><string>4::num</string></value></member><member><name>value</name><value><int>1</int></value></member></struct></value></member></struct></value></member></struct> </value> </param> </params> </methodResponse> here is the non-working response: <methodResponse> <params> <param> <value><struct><member><name>entry</name><value><int>6</int></value></member><member><name>size</name><value><int>1</int></value></member><member><name>slot</name><value><struct><member><name>item</name><value><struct><member><name>name</name><value><string>4::num</string></value></member><member><name>value</name><value><int>1</int></value></member></struct></value></member></struct></value></member></struct> <struct><member><name>entry</name><value><int>11</int></value></member><member><name>size</name><value><int>1</int></value></member><member><name>slot</name><value><struct><member><name>item</name><value><struct><member><name>name</name><value><string>3::num</string></value></member><member><name>value</name><value><int>1</int></value></member></struct></value></member></struct></value></member></struct> </value> </param> </params> </methodResponse> Kelvin Chua

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FW: Module TOPOH: Does not unmask CALL-ID on request BYE in one direction
by Samuel F. 04 Jul '14

04 Jul '14

Hi, We are setting up a Kamailio Proxy with the Topoh module in place masking everything including CALL-ID. It seems to be masking everything just fine except for CALL-ID in BYE requests (verified with PCAPs) when received from downstream. Upstream works fine. Looking at the source code for the module 4.1.2 we see the unmasking being done like this: if(dialog) { /* dialog request */ th_unmask_ruri(&msg); th_unmask_route(&msg); th_unmask_refer_to(&msg); if(direction==1) { th_unmask_callid(&msg); } } It only seems to be unmasking for one direction. From my understanding BYE can be initated from both directions? Might be something I don’t understand here how the source code works but I wanted to ask for any pointers of help here.. been troubleshooting this for a good while now. Kind regards, Samuel

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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.

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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

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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

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