- sr-users - kamailio.org

Calls per second
by Olle E. Johansson 03 Jul '14

03 Jul '14

Hi! Have you implemented a per-customer rate limit in Calls per second? If so - how? I've played with ratelimit/pipelimit and it seems like I can define a database with one pipe per customer - but have to restart Kamailio to add customers. There are warnings for low timer settings, like 1 second, but I don't know how up-to-date those warnings are. I guess I could play with hash tables and implement something pike-like there, but it seems like a workaround for something pretty common. So the question remains - how are you limiting on a cps per customer? /O

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sipcapture / raw_interface
by András Stefler 03 Jul '14

03 Jul '14

Hello all, I am new in using Kamailio. I have installed the version 4.0.1 and using it as SIP capture node. My server has lots of physical interfaces where I want to enforce capturing on one of the only. This is called 'csm1'. This is the corresponding part from kamailio.cfg modparam("sipcapture", "capture_node", "sipserv") modparam("sipcapture", "db_url", "mysql://****) modparam("sipcapture", "capture_on", 1) modparam("sipcapture", "hep_capture_on", 0) #modparam("sipcapture", "raw_socket_listen", "192.168.0.1:5060") modparam("sipcapture", "raw_interface", "csm1") modparam("sipcapture", "raw_ipip_capture_on", 0) modparam("sipcapture", "table_name", "sip_capture") modparam("sipcapture", "raw_sock_children", 4) modparam("sipcapture", "raw_moni_capture_on", 1) modparam("sipcapture", "promiscious_on", 1) modparam("sipcapture", "raw_moni_bpf_on", 1) ******************************** stefler@sipserv:~$ ifconfig csm1 csm1 Link encap:Ethernet HWaddr ac:16:2d:96:3f:e0 inet addr:1.0.8.1 Bcast:1.0.8.3 Mask:255.255.255.252 inet6 addr: fe80::ae16:2dff:fe96:3fe0/64 Scope:Link UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 My problem is that capturing is activated on all of the interfaces and only on the selected one. Have you seen this problem before? Regards Andras Stefler

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How to check message queue (buffer) in Kamailio
by AliReza Khoshgoftar Monfared 03 Jul '14

03 Jul '14

Hi, I had another simple question: In a kamailio server (proxy), how do I check the number of messages currently waiting for processing? Is there a variable that I can monitor, say, if I want to make a routing decision in my config based on the number of messages in the queue? Also, is it possible to get a head count by method? or is it only possible after fully parsing the message? I see that "ratelimit" module uses similar information, but I am not sure how to get the status of these queues that the module uses. Thanks

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Server generated 408 time out goes to onreply route?
by Allen Zhang 03 Jul '14

03 Jul '14

Hi, Does local server generated 408 time out reply go to onreply_route? If so, can I change the status code to a 200 class code by doing: $rs = "202"? If I can change it, does it still go to failure_route? (It's not a 400 class response anymore.) The scenario I want to test it's easy to produce. Otherwise I would just try it...... Regards, Allen

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Replys though onsend_route?
by Jöran Vinzens 03 Jul '14

03 Jul '14

Hi all, is there a possibility to pass reply and failure though onsend_route as well? We try to capture via siptrace the incoming and outgoing packet. So far this is only possible with Requests. I've found this: http://lists.sip-router.org/pipermail/sr-users/2012-May/073224.html Can anybody confirm this is still the behavior? Does anybody have a solution passing reply though onsend_route or any other way to capture the outgoing reply via siptrace? Thanks a lot for your help. BR Jöran Vinzens -- Jöran Vinzens - vinzens[at]sipgate.de <vinzens(a)sipgate.de> 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

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how to remove non-digits using dp_translate
by Henry Fernandes 03 Jul '14

03 Jul '14

I'd like to use the dialplan module to remove all non-digits from a string. For example, "1-306-555-1234" becomes "13065551234" and "(306) 555-1234" becomes "3065551234". Unfortunately, I can't figure out the regular expressions for subst_exp and repl_exp. I can easily replace one instance of a non-digit, however I can't figure out how to replace all non-digits. -H

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Making RLS presence work with Blink and Kamailio 4.1.4
by Alex Villacís Lasso 02 Jul '14

02 Jul '14

I am having trouble making all of the supposed features of Blink work with Kamailio 4.1.4. My kamailio.cfg file is attached. Specifically, what I am having trouble is with presence (the way Blink wants to implement it), and MSRP. Ordinary voice calls work correctly. With presence, I have managed to make XCAP storing and retrieval work, in the sense that a document that is being stored with PUT can be later retrieved with GET on the same URL. However, I cannot make Blink use this support in order to actually monitor presence. As far as I understand, when user A subscribes to the presence state of user B, and user B is registered, user B is supposed to receive a notification that user A requests authorization for presence subscription. However, this is not working so far when both A and B use Blink. From what I can see through a packet capture, neither adding a new contact on the Blink window or having existing contacts result in any SUBSCRIBE packets for the contact users being sent to the Kamailio instance. If the subscription is meant to be performed some other way, I am not seeing it being done either. Of course, this results in presence changes not being exchanged between contacts. The only SUBSCRIBE packet I see from Blink is the one for the SIP account used for Blink. With MSRP, I have copied the example from the msrp.so documentation, but I have replaced the authentication shown with a static password, with an actual query using auth_db.so, the very same query used to authenticate other SIP packets. Through the packet capture, when user A tries starting a MSRP chat with user B, an INVITE is sent from user A to Kamailio, which has a special check to NOT send this invite to Asterisk, but rather route it itself, and it gets "correctly" routed to user B. Then user B attempts to start a MSRP session with Kamailio (not user A) without authentication credentials, Kamailio challenges user B for the credentials, and... that's it. User B makes no attempt to repeat the MSRP session with the required credentials, and the communication eventually times out. The presence failure might have something to do with the following exchange I see in the packet capture: GET /xcap-root/pidf-manipulation/users/sip:avillacisIM@pbx.elastix.com/index HTTP/1.1 Accept-Encoding: identity Authorization: Digest username="avillacisIM", realm="pbx.elastix.com", nonce="U6xFz1OsRKNuU11dgljLCgGppv5lK6Eu", uri="/xcap-root/pidf-manipulation/users/sip:avillacisIM@pbx.elastix.com/index", response="3dfdf556c9812f4bebb8b0cc82578ef9", algorithm="MD5" Host: pbx.elastix.com:5060 Connection: close Accept: application/pidf+xml User-Agent: Blink 0.8.2 (Linux) HTTP/1.1 404 Not found Via: SIP/2.0/TCP 192.168.3.2:55567 Server: kamailio (4.1.4 (x86_64/linux)) Content-Length: 0 That is, the Blink client tries to fetch the pidf-manipulation document, and Kamailio answers with a 404. In other cases Blink has responded with a PUT of a minimal document at the required URI (which gets stored to the database), but in this case (pidf-manipulation) this is not done. I am not sure of what is supposed to happen here. This 404 is the same regardless of whether there are other (Blink or otherwise) SIP clients connected. I think this has been asked before, but is there a working implementation of MSRP+presence for Kamailio that works with Blink? Or failing that, a complete kamailio.cfg that has been shown to work with at least one MSRP implementation?

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UAC remote registration - refreshing database
by Dan Christian Bogos 02 Jul '14

02 Jul '14

Hey Guys, Anybody aware if it is possible to refresh the list of remote registrations from the database without restarting the whole server? Ta, DanB

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TCP rest by peer
by Rob Moore 02 Jul '14

02 Jul '14

Hi All, Since we've been using sipPhones as a proxy for registrations and calls to phones over TCP. I've been seeing a couple of these crop up in the logs. /usr/sbin/kamailio[4255]: ERROR: <core> [tcp_read.c:1131]: ERROR: tcp_read_req: error reading /usr/sbin/kamailio[4255]: ERROR: <core> [tcp_read.c:293]: error reading: Connection reset by peer (104) We haven't had any complaints of problems with the phones so if these are not a problem I'd like to know how to clean them up. Any suggestions on cause or solution would be fantastic. Thanks in advance. Rob Kind regards, Robert Moore Telephony System Architect [Description: Aeriandi] m. +44 (0)7766 838 040 t. +44 (0)845 108 0308 [cid:image002.png@01CF8C82.823C49E0]<http://www.retailweektechnologyawards.com/winners-2014> Aeriandi Ltd / Prama House, 267 Banbury Road, Oxford OX2 7HT / +44 (0)845 108 0308 / info(a)aeriandi.com<mailto:info@aeriandi.com> / aeriandi.com<http://www.aeriandi.com/> This transmission is intended solely for the addressee and may be confidential. If you are not the named addressee, or if the message has been addressed to you in error, you must not read, disclose, reproduce, distribute or use this transmission. Please delete the message or contact the sender. Delivery of this message to any person other than the named addressee is not intended in any way to waive confidentiality. Please note that neither Aeriandi Ltd nor the sender accepts any responsibility for any viruses, which may be transmitted, and it is your responsibility to scan attachments (if any).

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msilo possible bug
by Kelvin Chua 02 Jul '14

02 Jul '14

Hi guys, i think i have stumbled on an msilo bug. m_dump() is not working. kamctl fifo get_statistics all | grep mess msilo:dumped_messages = 0 msilo:failed_messages = 0 msilo:stored_messages = 1 so here we can see there is a message to be delivered. i turned on debug mode, then the intended destination of message logs in. then this is what i see on logs DEBUG: msilo [msilo.c:1082]: m_dump(): dumping [1] messages for <sip:XX85X80X74@1X8.1X9.1X7.1X3>!!! DEBUG: msilo [ms_msg_list.c:169]: msg_list_check_msg(): checking msgid=405 DEBUG: msilo [ms_msg_list.c:207]: msg_list_check_msg(): msg already in sent list. DEBUG: msilo [msilo.c:1089]: m_dump(): message[0] mid=405 already sent. using ngrep, no message was ever seen on the wire. then, kamctl fifo get_statistics all | grep mess msilo:dumped_messages = 0 msilo:failed_messages = 0 msilo:stored_messages = 1 still the same. anybody experienced this? Kelvin Chua

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Fedora 18-20 GeoIP missing?
by Roberto Fichera 02 Jul '14

02 Jul '14

Hi All, is there any reason why the GeoIP module is missing in the Fedora repo? Thanks in advance. Roberto Fichera.

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Possible GPL violation by third party (4PSA)
by Matthew Costa 02 Jul '14

02 Jul '14

4PSA develop and sell a software suite called VoipNow Professional (version 2.5) which, amongst other things, includes it's own packaged copy of Kamailio. This version is derived from Kamailio version 1.5.4. The software installs a binary release from a 4PSA repository. The RPM information contains the following: # rpm -qi voipnow-kamailio-1.5.4-130312.41.rhel5 Name : voipnow-kamailio Relocations: (not relocatable) Version : 1.5.4 Vendor: Rack-Soft, Inc <devel(a)4psa.com> Release : 130312.41.rhel5 Build Date: Tue 12 Mar 2013 18:31:30 EST Install Date: Sun 02 Feb 2014 00:15:14 EST Build Host: rhel5-1.64b.build.4psa.ro Group : System Environment/Daemons Source RPM: voipnow-kamailio-1.5.4-130312.41.rhel5.src.rpm Size : 6599403 License: GPL Signature : RSA/SHA1, Tue 12 Mar 2013 18:35:28 EST, Key ID 42ba8c472f75de11 Packager : Rack-Soft, Inc <devel(a)4psa.com> URL : http://kamailio.org/ Summary : Kamailio, a very fast and flexible SIP Proxy I've made a direct request to 4PSA for the source code for this package, only to be told they don't release their sources. The request, and subsequent response, can be found at http://my.4psa.com/4psa/topics/where_can_i_find_the_source_code_for_the_voi pnow_2_5_x_modified_version_of_kamailio. For those interested, the following packages are included with VoipNow which are derived from GPL software. There is no source available for any of these from the vendor. # rpm -qa |grep voipnow voipnow-spandsp-0.0.6-100708.10.rhel5 voipnow-asterisk-1.6.1.20-130312.07.rhel5 voipnow-asterisk-addons-1.6.1-130131.30.rhel5 voipnow-php-2.5.5-130312.11.rhel5 voipnow-sox-14.3.1-100723.30.rhel5 voipnow-asterisk-sounds-1.4.22-100723.25.rhel5 voipnow-kamailio-1.5.4-130312.41.rhel5 voipnow-ejabberd-2.1.4-100723.29.rhel5 voipnow-asterisk-debuginfo-1.6.1.20-120417.09.rhel5 voipnow-asterisk-extra-2.5.5-130312.11.rhel5 My requests are falling on deaf ears. Does anybody know how to make them take this matter seriously? Cheers, Matthew Costa Infrastructure Manager <http://www.greenlight-itc.com/> Suite 703, Level 7 815 Pacific Highway Chatswood 2067 tel: 02 8412 0000 fax: 02 8412 0001

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Re: [SR-Users] Kamailio Freepbx Integration Dropping Calls
by Carlos Rangel 01 Jul '14

01 Jul '14

Hello List Hopefully someone can help. This is the problem when the call is hug up 20-30 seconds after it initiates. The call is only hung on when the remote extension initiates the call. If the remote extension receives the call there is no problem the call is not hung on. I changed the remote cisco phone for a yealink and it is the same behavior. It thought it was the phone. This is what I am using in kamailio.cfg #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_ASTERISK #!define WITH_USRLOCDB #!define WITH_ANTIFLOOD Remote User Internet Internal network Yealink IP TG28P ----DSL router ---|------Internet --------|-----Cisco ASA 5500 FW--------------Kamailio/Freepbx (Same Box)------IAX Trunk----------Freepbx Production Server --------|------ PSTN Thanks Carlos Rangel De: Carlos Rangel [mailto:crangel@globaltelesourcing.com] Enviado el: jueves, 26 de junio de 2014 01:27 p.m. Para: miconda(a)gmail.com; 'Kamailio (SER) - Users Mailing List' Asunto: RE: [SR-Users] Kamailio Freepbx Integration Dropping Calls Hi Daniel Thank you so much for your response. Here is the SIP trace of one of the calls, I am not sure where the call initiates but you can see at the end of the file in bold X-Asterisk-HangupCause: No user responding. I am not sure why is it sending this message though. The variables are Extension/Username=XXXXX Ext_IP= Public IP Internal_IP= Asterisk/Kamailio internal IP Sorry for the long file but again I am not sure where the call initiates This is the part where that call is hung on. U 2014/06/26 13:36:11.831965 Kamailio_IP:5080 -> Kamailio_IP:5060 BYE sip:XXXXX@65.190.71.203:5060;user=phone;transport=udp SIP/2.0. Via: SIP/2.0/UDP Kamailio_IP:5080;branch=z9hG4bK7ce1be47. Route: <sip:Kamailio_IP;lr=on;ftag=000653dc394000970f227678-1fafb4e2>. Max-Forwards: 70. From: <sip:919707249077@Kamailio_IP:5060>;tag=as2e670ea4. To: "User" <sip:XXXXX@Kamailio_IP:5060>;tag=000653dc394000970f227678-1fafb4e2. Call-ID: 000653dc-3940000b-33caf1b2-20ccd185(a)192.168.0.22. CSeq: 102 BYE. User-Agent: FPBX-2.11.0(11.10.2). X-Asterisk-HangupCause: No user responding. X-Asterisk-HangupCauseCode: 18. Content-Length: 0. . U 2014/06/26 13:36:11.832260 Kamailio_IP:5060 -> 65.190.71.203:5060 BYE sip:XXXXX@65.190.71.203:5060;user=phone;transport=udp SIP/2.0. Via: SIP/2.0/UDP Kamailio_IP;branch=z9hG4bKcf68.d6ef5aa9cc5bd0fb0ab13a563b7cf284.0. Via: SIP/2.0/UDP Kamailio_IP:5080;branch=z9hG4bK7ce1be47. Max-Forwards: 69. From: <sip:919707249077@Kamailio_IP:5060>;tag=as2e670ea4. To: "User" <sip:XXXXX@Kamailio_IP:5060>;tag=000653dc394000970f227678-1fafb4e2. Call-ID: 000653dc-3940000b-33caf1b2-20ccd185(a)192.168.0.22. CSeq: 102 BYE. User-Agent: FPBX-2.11.0(11.10.2). X-Asterisk-HangupCause: No user responding. X-Asterisk-HangupCauseCode: 18. Content-Length: 0. Descripción: Description: Description: DLR-Logo-No-TextCARLOS RANGEL | INFORMATION TECHNOLOGY DIRECTOR Global Telesourcing México, S. de R.L. de C.V. | Aarón Sáenz #1891-1 | Monterrey, N.L., México Direct 703 894 1667 | Mobile US 703 894 1667 | Mobile MX +52 1 812 000 7362 | crangel(a)globaltelesourcing.com ________________________________________________________________________________________________________________ The information contained in this e-mail and any attached documents may contain information that is confidential or otherwise protected from disclosure. If you are not the intended recipient of this message, or if this message has been sent to you in error, please immediately alert the sender by reply e-mail and then delete this message, including any attachments. Any dissemination, distribution or other use of the contents of this message by anyone other than the intended recipient is strictly prohibited. De: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] En nombre de Daniel-Constantin Mierla Enviado el: jueves, 26 de junio de 2014 03:12 a.m. Para: Kamailio (SER) - Users Mailing List Asunto: Re: [SR-Users] Kamailio Freepbx Integration Dropping Calls Hello, can you gran the SIP trace on kamailio server for such case? You can use ngrep, like: ngrep -d any -qt -W byline port 5060 and send the output to the mailing list. You can replace any sensitive information (e.g., ip address) before sending to mailing list. The typical call drop after 30-40 secs is when ACK is not routed properly, but we have to see that in the sip trace. Cheers, Daniel On 25/06/14 18:50, Carlos Rangel wrote: Hello I have successfully (I believe) implemented Kamailio 4.1.4 integration with Freepbx 5.2.11 taking as a guide Daniel’s tutorial http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb. I just did not create the voicemail tables because voice mail is handled by Freepbx. I installed the system in a separate box for testing and connected to the Freepbx Production server via IAX trunk. The system is behind a Cisco Firewall and looks like this Remote User Internet Internal network Cisco 7960 ----DSL router ---|------Internet --------|-----Cisco ASA 5500 FW--------------Kamailio/Freepbx (Same Box)------IAX Trunk----------Freepbx Production Server --------|------ PSTN I have configured the FW to allow UDP and TCP traffic from the corresponding IP as well as tfpt that is needed for the Ciscos to pick up the configuration from the server. I have a few remotes Cisco 7960 phones that can register remotely in Kamailio as long as the user is added with kamctl add user password and as long as the extension is created in Freepbx. The problem that I have is when try to make a call from the remote Ciscos the call is dropped after 30 or 40 seconds. I can see from the logs that the problem appears to be that the server is not receiving responses from the phone 06-25 10:57:30] WARNING[1814] chan_sip.c: Retransmission timeout reached on transmission 000653dc-39400006-2579bbcd-13d9adcb(a)192.168.0.22 for seqno 102 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 32001ms with no response [2014-06-25 10:57:30] WARNING[1814] chan_sip.c: Hanging up call 000653dc-39400006-2579bbcd-13d9adcb(a)192.168.0.22 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). Is this something that we can adjust in kamailio or could it be related to the FW configuration?? Sorry but I am very new to kamailio and sip. Thanks Carlos _______________________________________________ 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

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installation siremis error
by Hossein Aghaie 01 Jul '14

01 Jul '14

hi after install kamailio 4.1.0 on ubuntu server,I try to install siremis-4.1.0 of this Doc : http://kb.asipto.com/siremis:install41x:main but to end setup,show this error : http://paste.fedoraproject.org/114029/ how to solved this problem? tnx

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Kamailio Development Workshop, July 9-10, 2014, Paris, France
by Daniel-Constantin Mierla 01 Jul '14

01 Jul '14

Hello, I want to announce the next Kamailio Development Workshop, to take place in Paris, France, during July 9-10, 2014. This event is the next in its series targeting to show the internals of Kamailio and enable more people to become developers as well as let users of Kamailio to get more knowledge about the design of the application which can have relevant impact in operating deployment. Be aware that it is not a workshop about Kamailio installation and administration. Its content is about writing C code to extend Kamailio. More details can be found at: - http://www.kamailio.org/w/2014/06/kamailio-development-workshop-july-9-10-2… Similar to the past events, we are considering to have a social networking event in the evening of July 9 (most probably a dinner meetup at a nice place in Paris), which is open for everyone, each participant taking care of own expenses. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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how to compile module db_oracle on Centos 6.5
by Konstantin Khatskevich 01 Jul '14

01 Jul '14

Hello! 5. Installation Because it dependes on an external library, the oracle module is not compiled and installed by default. You can use one of the next options. * - edit the "Makefile" and remove "db_oracle" from "excluded_modules" list. Then follow the standard procedure to install Kamailio: "make all; make install". * - from command line use: 'make all include_modules="db_oracle"; make install include_modules="db_oracle"'. But "db_oracle: requires non-free Oracle client SDK" (http://lists.sip-router.org/pipermail/sr-dev/2013-October/021803.html) What can I do? -- Best regards, Konstantin Khatskevich

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Changes in registrar module source code
by aawaise 01 Jul '14

01 Jul '14

Hello, I am a user of kamailio server. I am in need to pick domain of particular online users from location list. I found out that usrloc module is used for Location Table items. And we cant directly export parameters from usrloc module. I decided to use registrar module to approach domain parameter of usrloc module. I want to understand complete flow control in registrar module source code, so that I can add my commands to pick domain parameter from usrloc module. How can it be done ? Secondly, if I make a change in registrar module, to see its reaction, I will have to recompile kamailio server or is there any shortcut to immediately see effect of changes made in registrar module ?? Any help will be appreciated. -- View this message in context: http://sip-router.1086192.n5.nabble.com/Changes-in-registrar-module-source-… Sent from the Users mailing list archive at Nabble.com.

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Siremis 4.1.0 menu adjustment for different users
by Fred 01 Jul '14

01 Jul '14

Hi, I'm setting up a Registrar server using Kamailio 4.1.3 with Siremis 4.1.0 as WebGUI. On Siremis I struggle to set up different users which only see limited menu items. What I want to do is e.g. an user called "CustomerService" which only sees the "Subscriber_List" menu item and a second user called "SystemAdmin" which only sees the "Dispatcher_List" menu item. I tried different things with new roles, groups and adjustments in the menu administration but didn't manage to get the desired result. Does anyone have a hint for me if that is possible and how? Thanks Fred

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Child Processes Stopped kamailio 4.0.6
by Jonathan Hunter 30 Jun '14

30 Jun '14

Hi Guys, Sorry thought I better raise this in new stream to avoid confusion. We have another crash today, not sure if the below is of much use without debug? Thanks Jon Jun 26 09:38:09 pbx /usr/sbin/kamailio[18529]: ALERT: <core> [main.c:788]: handle_sigs(): child process 18532 exited by a signal 11Jun 26 09:38:09 pbx /usr/sbin/kamailio[18529]: ALERT: <core> [main.c:791]: handle_sigs(): core was not generatedJun 26 09:38:09 pbx /usr/sbin/kamailio[18529]: INFO: <core> [main.c:803]: handle_sigs(): INFO: terminating due to SIGCHLDJun 26 09:38:09 pbx /usr/sbin/kamailio[18531]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:09 pbx9 /usr/sbin/kamailio[18529]: ALERT: <core> [main.c:788]: handle_sigs(): child process 18532 exited by a signal 11Jun 26 09:38:09 pbx9 /usr/sbin/kamailio[18529]: ALERT: <core> [main.c:791]: handle_sigs(): core was not generatedJun 26 09:38:09 pbx9 /usr/sbin/kamailio[18529]: INFO: <core> [main.c:803]: handle_sigs(): INFO: terminating due to SIGCHLDJun 26 09:38:09 pbx9 /usr/sbin/kamailio[18531]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:09 pbx9 /usr/sbin/kamailio[18533]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:09 pbx9 /usr/sbin/kamailio[18537]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:09 pbx9 /usr/sbin/kamailio[18534]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:09 pbx9 /usr/sbin/kamailio[18535]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:09 pbx9 /usr/sbin/kamailio[18538]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:09 pbx9 /usr/sbin/kamailio[18530]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:09 pbx9 /usr/sbin/kamailio[18539]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:09 pbx9 /usr/sbin/kamailio[18536]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:09 pbx9 kamailio[2076]: ERROR: <script>: Failed Reg Block Traffic from 194.72.34.18Jun 26 09:38:09 pbx kamailio[2075]: ERROR: <script>: Failed Reg Block Traffic from 194.72.34.18Jun 26 09:38:09 pbx kamailio[2074]: ERROR: <script>: Routing Call To Asterisk serverJun 26 09:38:09 pbx kamailio[2074]: INFO: <script>: Routing Call to Dispatcher group 1 -Asterisk Media ServersJun 26 09:38:10 pbx kamailio[2072]: ALERT: <core> [main.c:788]: handle_sigs(): child process 2075 exited by a signal 11Jun 26 09:38:10 pbx kamailio[2072]: ALERT: <core> [main.c:791]: handle_sigs(): core was generatedJun 26 09:38:10 pbx kamailio[2072]: INFO: <core> [main.c:803]: handle_sigs(): INFO: terminating due to SIGCHLDJun 26 09:38:10 pbx kamailio[2074]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:10 pbx kamailio[2078]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:10 pbx kamailio[2076]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:10 pbx kamailio[2077]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:10 pbx kamailio[2079]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:10 pbx kamailio[2073]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:10 pbx kamailio[2080]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:10 pbx kamailio[2081]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:10 pbx kamailio[2082]: INFO: <core> [main.c:854]: sig_usr(): INFO: signal 15 receivedJun 26 09:38:10 pbx kamailio[2072]: ERROR: ctl [ctl.c:379]: mod_destroy(): ERROR: ctl: could not delete unix socket /tmp/kamailio_ctl: No such file or directory (2)Jun 26 09:43:33 pbx kamailio: INFO: <core> [tcp_main.c:4846]: init_tcp(): init_tcp: using epoll_lt as the io watch method (auto detected)

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How to extract information from location table
by aawaise 30 Jun '14

30 Jun '14

I want to know is there any indirect method to extract information from location table that is kept by Usrloc module ?? I want to know about the username and domain name of users from Location Table on arrival of INVITE packet. How can it be done ? -- View this message in context: http://sip-router.1086192.n5.nabble.com/How-to-extract-information-from-loc… Sent from the Users mailing list archive at Nabble.com.

2 1

route [Location] Documentation
by aawaise 30 Jun '14

30 Jun '14

I want to know that how route [Location] works when an INVITE packet approaches kamailio server. Basically route [Location] gathers information whether callee is a valid user or not. Secondly if it is online or not. I want to understand this whole procedure. Any help will be appreciated. Thanks. -- View this message in context: http://sip-router.1086192.n5.nabble.com/route-Location-Documentation-tp1286… Sent from the Users mailing list archive at Nabble.com.

2 1

Voice quality issues when registering to Kamailio
by Travis Dillon 30 Jun '14

30 Jun '14

Hello all, I have an environment setup as follows: VoIP routers -> Kamailio 4.0 -> Asterisk 11.6 When I register the phones to Asterisk the call quality is excellent. So to be clear, the call comes in the VoIP router, routes through Kamailio to the phone registered to Asterisk. When I register the phone to Kamailio the call quality is significantly reduced. So clarity here, the call comes in the VoIP router, routes through Kamailio to the phone registered to Kamailio. Why would the call quality diminish when registered to Kamailio? Where can I start looking for this issue? Thank you, Travis R. Dillon

2 1

UAC module refresh registration, could not reuse callid/fromtag !!!
by nevious 30 Jun '14

30 Jun '14

Hello all! I make kamailio1 remotely register to kamailio2. Here I set K1 modparam("uac", "reg_timer_interval", 15), and expires in uacreg as 30s. Then K1 should refresh registration after 15s. However the callid and from tag of every registration are different, and the Cseq just 10 or 11,never increased.In RFC3261, when UAC want refresh bindings, A UA SHOULD use the same Call-ID for all registrations during a single boot cycle. Is there any way to reuse the callid and fromtag during refresh registration ? Many thanks! And how to make the Cseq become 12 or more?

4 4

Strange application hang querying DB
by Tim Chubb 27 Jun '14

27 Jun '14

Hi, I'm currently wracking my brains trying to deduce where an application hang can be coming from. I have ported the PostGres schema to be compatible with MS Sql Server and have had no issues getting a working dev environment working. However when I try and use SQL Azure as the backend db as opposed to SQL Server 2012 (running locally on my dev machine) Kamailio will seem to hang after about 10 - 15 mins of flawless operation. I have recreated the SQL Azure schema from the working schema running on my dev machine and copied all data in a 1:1 manner so I can be certain it's not a data quality issue. I am using the same FreeTDS/UnixODBC config that I am using for Asterisk realtime (which doesn't exhibit this behaviour). The really strange thing is that I can see the queries being sent by Kmailio in the freetds trace log, and the response from SQL Azure, however its as if the db response handler is deadlocked/hung. The kamailio process continues to work fine, and as long as its not required to commiunicate with the DB all appears well i.e. no error messages in the log file. This behaviour only happens when im using SQL Azure as opposed to SQL Server, so there must be something causing the issue, that's unique to SQL azure as opposed to SQL Server. As far as I can tell this is not something I can create a backtrace for as no core dumps are produced, as far as I can tell there is no easy way debug what the lock is. If anyone has any suggestions I would be very grateful to hear them. One thing that occurs to me is that the db_unixodbc module doesn't have half as many parameters as its equilent in Asterisk, and that connection pooling may be in use, this could be a bit of a show stopper AFAIK you have to disable connection pooling with SQL server, certainly any of the guides I have come across configuring Asterisk to work with SQL server suggests that. Tim Chubb Developer [VS-logo-email] The Coach House, Heywood House, Park Lane, Heywood, Westbury, BA13 4NA Direct: +44 (0)333 0110164 Email: tim.chubb(a)voicesimplified.com<mailto:tim.chubb@voicesimplified.com> Web: www.VoiceSimplified.com<http://www.voicesimplified.com/> Voice Simplified Ltd (registered in England & Wales: 07171825) registered office: Curzon House, 2nd Floor, 24 High Street, Banstead, Surrey. SM7 2LJ. The information in this email is confidential and may be legally privileged. It is intended solely for the addressee. Access to this email by anyone else is unauthorised. If you are not the intended recipient, any disclosure, copying, distribution or any action taken or omitted to be taken in reliance on it, is prohibited and may be unlawful. If you are not the intended recipient, please return the message to the sender and delete it from your records.

2 3

Error when install Kamailio 4.1.4 on Mac OS 10.9.
by Son Truong 27 Jun '14

27 Jun '14

Hi Kamailio, - I meet an error when I am trying install Kamailio 4.1.4 on Mac OS 10.9. The error: touch /usr/local/lib64/kamailio/modules/usrloc.so install -m 755 usrloc.so /usr/local/lib64/kamailio/modules touch /usr/local/lib64/kamailio/modules/xhttp.so install -m 755 xhttp.so /usr/local/lib64/kamailio/modules touch /usr/local/lib64/kamailio/modules/xhttp_rpc.so install -m 755 xhttp_rpc.so /usr/local/lib64/kamailio/modules touch /usr/local/lib64/kamailio/modules/xlog.so install -m 755 xlog.so /usr/local/lib64/kamailio/modules touch /usr/local/lib64/kamailio/modules/xprint.so install -m 755 xprint.so /usr/local/lib64/kamailio/modules /bin/sh: -c: line 1: syntax error: unexpected end of file make: *** [install-cfg] Error 2 How can I fix it? I am new in Kamailio, I want to learn Kamailio. -- Thanks and best regards, Phạm Sơn Trường Android, iOS Team Lead at ME Corp. Email1: truonguit2010(a)gmail.com Email2: truongps(a)mecorp.vn Mobile: 01218816208

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