sr-users July 2013

sr-users@lists.kamailio.org

92 participants
148 discussions

Kamailio CDR issue
by Nandini madhu 06 Aug '13

06 Aug '13

Dear All, Greetings, I am working Kamailio (4.0.1) server with GUI package Siremis (4.0) of it. I was successful in get through Calls and SMS features in our VoIP phones using this kamailio sever. And now i am trying to get GUI for my kamailio server with SIREMIS (4.0.0) package. Installation was good as i am getting subscribers list in GUI which are in MySQL database. As it comes accounting section. i am Failed to get the CDR list when the call session between two clients have been done. I gone through Acc module's documentation and followed the following links to set up the accounting in GUI : http://kb.asipto.com/siremis:install40x:accounting. And i read this is description in accounting module documentation : 'Note that CDR generation does not involve any kind of database storage (yet). In order to persist the CDRs into a database you will have to set up an exterior process (i.e., a script living outside of Kamailio) and implement the storage task yourself'. So what does it mean ? should i have to install external process like CDR tool and Freeradius? Or is there any other way to get the CDR related information in GUI using database backend. Please find below my kamailio configuration file. By the way CDR data is logged to syslog fine, but i dont know where is wrong for database backend. Any help will greatly appreciate. Regards, Nandini

3 2

Asterisk and Kamailio as a sip router
by Rafael Carvallo 02 Aug '13

02 Aug '13

Hey everyone, currently i have this configuration set on kamailio: modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list") modparam("dispatcher", "force_dst", 0) #forzado de la reescritura direccion de destino modparam("dispatcher", "flags", 2) #banderas de funcionamiento, 2 significa "soporte para failover" modparam("dispatcher", "dst_avp", "$avp(dsdst)") modparam("dispatcher", "grp_avp", "$avp(dsgrp)") modparam("dispatcher", "cnt_avp", "$avp(dscnt)") modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_ping_interval", 5) #tiempo que transcurre antes de verificar nuevamente una salida inactiva modparam("dispatcher", "ds_probing_threshhold", 5) #Numero de intentos antes de marcar una salida como inactiva modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=403;code=488;class=3") modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_hash_expire", 3600) modparam("dispatcher", "ds_hash_initexpire", 60) #loadmodule "dispatcher.so" ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { if (is_method("SUBSCRIBE")){ if (src_ip == 192.168.2.1 | src_ip == 192.168.2.2){ t_relay(); } route(REGISTRAR); } if (src_ip == 192.168.2.1 | src_ip == 192.168.2.2){ t_relay(); } else{ route(ASTERISK); } } route[ASTERISK]{ ds_select_dst("1", "8"); t_relay(); exit(); } Everything else is left with the same basic configuration it had when i installed the software. I want to use it kinda of a sip router, so what Kamailio does is just forward the packets betwen my Asterisk boxes and the Sip Phones. Currently it seems to work (partially) but i have doubts about if this is correctly done (it's the first time using Kamailio and i need this working withing a week at most). As you can see i use the module dispatcher for failover/failback (this is the purpose of using Kamailio, a failover/failback setup). One major problem i've found with this setup is, if the phones are currently connected and working with one of my 2 asterisk boxes and if that box fails, Kamailio starts sending the traffic to the second box (as intended), but the phones don't try to subscribe to the new asterisk box, rather they just keep sending traffic (and obviously kamailio forwarding it). Sometimes one of the phones subscribe to the new box, but that's not always the case, the packets reach the new Asterisk box, but since the phones aren't registered to it, they can't make calls. Other times the behavior of the setup is rather weird, i can call some extensions and some others i can not (even though they are registered within the asterisk box) The traffic gets to the Asterisk box (as shown in the asterisk logs) but the call is shown as "service unavailable". I've checked a lot of times the Asterisk setup and it seems to be fine so i think it has something to do with Kamailio. I know this might be a really bad configuration file, but it's been at most 3 days since i started using Kamailio which i sometimes find kinda hard to understand and i really need this working within a week. To summarize all i want is kamailio forwarding packets between the currently active server and the phones so if it fails, then the packets go to the second one, the phones must re-subscribe to the new active server.

2 1

kamailio and oracle dbms/database
by Barthel Marco (CI/AFU1) 01 Aug '13

01 Aug '13

Hello, anyone out there using kamailio and oracle database in production system? How's your experience on that? We may be forced to migrate from mysql to oracle dbms. Mit freundlichen Grüßen / Best regards Marco Barthel Robert Bosch GmbH (CI/AFU1) Postfach 30 02 20 70442 Stuttgart GERMANY www.bosch.com<http://www.bosch.com> Marco.Barthel(a)de.bosch.com<mailto:Marco.Barthel@de.bosch.com> Sitz: Stuttgart, Registergericht: Amtsgericht Stuttgart, HRB 14000; Aufsichtsratsvorsitzender: Franz Fehrenbach; Geschäftsführung: Dr. Volkmar Denner, Dr. Stefan Asenkerschbaumer, Dr. Rolf Bulander, Dr. Stefan Hartung, Dr. Dirk Hoheisel, Christoph Kübel, Uwe Raschke, Wolf-Henning Scheider, Dr. Werner Struth, Peter Tyroller

2 1

Kamailio as a sip/rtp proxy
by cormac 31 Jul '13

31 Jul '13

Hi there, I am trying to implement a sip/rtp proxy. I have one public ip address with a domain name. example (whatever.com) On the lan side i have multiple asterisk instances all listening on 5060 for sip requests. These instances will be assigned to different customers I am hoping to run kamailio in front of these instances and route sip and rtp traffic to the correct asterisk instance. So in this use case asterisk instances will be used as application servers and kamailio will be used as a sip/rtpproxy with nathelper So my question is what would be the best way and using which modules can i route the requests for customerone.whatever.com to the first asterisk instance and customertwo.whatever.com to the second. is this achieved with the dispatcher module? and if so, where can it be defined that requests for cusomertone.whatever.com is routed to the first asterisk instance. Kind regards, Cormac O'Loughlin

2 1

websocket double record_route()
by Vitaliy Aleksandrov 31 Jul '13

31 Jul '13

Hello. I'm trying to configure kamailio as a gateway between Websocket and TCP/TLS transports. When I call record_route() for an initial INVITE that comes via WS and will be forwarded via TCP to a registered UA kamailio inserts only a one record-route header with its IP and transport=ws instead of two record-route headers with both incoming/outgoing transports. This behaviour breaks in-dialog requests routing. rr module parameters are: modparam("rr", "enable_full_lr", 1) modparam("rr", "append_fromtag", 0) modparam("rr", "enable_double_rr", 1) I use: - kamailio-4.0.2 - sipml5 as sip client Can anybody point me in the right direction to understand why it happens ?

3 10

Re: [SR-Users] kamailio call problem
by Jason Sia 31 Jul '13

31 Jul '13

Sorry, this test was done on a public server that's why it was showing public IPs. I was trying out opensips and kamailio and same response for both. Do you think this is a network firewall issue? On Tue, Jul 30, 2013 at 2:31 PM, Gertjan Wolzak <g.wolzak(a)foize.com> wrote: > Hello Jason,**** > > ** ** > > Well, as you can see now the error is:**** > > ** ** > > U 2013/07/29 18:24:03.328456 198.23.160.81:5060 -> 49.144.184.97:41998 > SIP/2.0 477 Send failed (477/TM). > Via: SIP/2.0/UDP 192.168.0.107:41998 > ;received=49.144.184.97;branch=z9hG4bK-d8754z-1cfc096228010333-1---d8754z-;rport=41998. > To: <sip:639195015475@198.23.160.81 > >;tag=eb0b3ccde0b882baee99bc071578cb61-d8d9. > From: "639178864952"<sip:639178864952@198.23.160.81>;tag=1b3b463c. > Call-ID: NjhmYzk0ODZhNWVmOGE5MTU4ZWExM2Q5NjI1NDVhODg. > CSeq: 2 INVITE. > Server: OpenSIPS (1.8.3-notls (x86_64/linux)). > Content-Length: 0.**** > > ** ** > > But as I am not a guru at all and this is a Kamailio users mailing list > and not an opensips list….**** > > ** ** > > I just googled the error, found that for Opensips and Kamailio it has > something to do with not being able to make a tcp connection.**** > > ** ** > > So, check the port translations, make a tcpdump of wireshark dump and see > if what is going wrong.**** > > ** ** > > You said that the clients and Kamailio were on the same lan, but from the > logging you can see that you are on the internet…….**** > > ** ** > > Hope this helps a bit.**** > > ** ** > > Good luck.**** > > ** ** > > ** ** > > *From:* sr-users-bounces(a)lists.sip-router.org [mailto: > sr-users-bounces(a)lists.sip-router.org] *On Behalf Of *Jason Sia > *Sent:* maandag 29 juli 2013 18:16 > *To:* Kamailio (SER) - Users Mailing List > *Subject:* Re: [SR-Users] kamailio call problem**** > > ** ** > > Hi,**** > > Thanks for the advice. Here is the ngrep logs: > > U 2013/07/29 18:23:52.994331 49.144.184.97:41998 -> 198.23.160.81:5060 > INVITE sip:639195015475@198.23.160.81 SIP/2.0. > Via: SIP/2.0/UDP 192.168.0.107:41998;branch=**** > > z9hG4bK-d8754z-2999840441d3d05a-1---d8754z-;rport. > Max-Forwards: 70. > Contact: <sip:639178864952@49.144.184.97:41998>. > To: <sip:639195015475@198.23.160.81>. > From: "639178864952"<sip:639178864952@198.23.160.81>;tag=1b3b463c. > Call-ID: NjhmYzk0ODZhNWVmOGE5MTU4ZWExM2Q5NjI1NDVhODg. > CSeq: 1 INVITE. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, > SUBSCRIBE, INFO. > Content-Type: application/sdp. > Supported: replaces. > User-Agent: X-Lite release 4.5.3 stamp 70576. > Content-Length: 357. > . > v=0. > o=- 1375107832761441 1 IN IP4 192.168.0.107. > s=X-Lite 4 release 4.5.3 stamp 70576. > c=IN IP4 192.168.0.107. > t=0 0. > m=audio 53084 RTP/AVP 123 9 0 8 97 100 98 101. > a=rtpmap:123 opus/48000/2. > a=fmtp:123 useinbandfec=1. > a=rtpmap:97 speex/8000. > a=rtpmap:100 speex/16000. > a=rtpmap:98 ILBC/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-15. > a=sendrecv. > > > U 2013/07/29 18:23:52.994422 198.23.160.81:5060 -> 49.144.184.97:41998 > SIP/2.0 407 Proxy Authentication Required. > Via: SIP/2.0/UDP 192.168.0.107:41998 > ;received=49.144.184.97;branch=z9hG4bK-d8754z-2999840441d3d05a-1---d8754z-;rport=41998. > To: <sip:639195015475@198.23.160.81 > >;tag=80403214130c49515b8f7d7842a4c119.ef65. > From: "639178864952"<sip:639178864952@198.23.160.81>;tag=1b3b463c. > Call-ID: NjhmYzk0ODZhNWVmOGE5MTU4ZWExM2Q5NjI1NDVhODg. > CSeq: 1 INVITE. > Proxy-Authenticate: Digest realm="198.23.160.81", > nonce="51f67b16000000fae8c70592af760c78fa39cd43f84c1a6b". > Server: OpenSIPS (1.8.3-notls (x86_64/linux)). > Content-Length: 0. > . > > > U 2013/07/29 18:23:53.299025 49.144.184.97:41998 -> 198.23.160.81:5060 > ACK sip:639195015475@198.23.160.81 SIP/2.0. > Via: SIP/2.0/UDP 192.168.0.107:41998 > ;branch=z9hG4bK-d8754z-2999840441d3d05a-1---d8754z-;rport. > Max-Forwards: 70. > To: <sip:639195015475@198.23.160.81 > >;tag=80403214130c49515b8f7d7842a4c119.ef65. > From: "639178864952"<sip:639178864952@198.23.160.81>;tag=1b3b463c. > Call-ID: NjhmYzk0ODZhNWVmOGE5MTU4ZWExM2Q5NjI1NDVhODg. > CSeq: 1 ACK. > Content-Length: 0. > . > > > U 2013/07/29 18:23:53.317842 49.144.184.97:41998 -> 198.23.160.81:5060 > INVITE sip:639195015475@198.23.160.81 SIP/2.0. > Via: SIP/2.0/UDP 192.168.0.107:41998 > ;branch=z9hG4bK-d8754z-1cfc096228010333-1---d8754z-;rport. > Max-Forwards: 70. > Contact: <sip:639178864952@49.144.184.97:41998>. > To: <sip:639195015475@198.23.160.81>. > From: "639178864952"<sip:639178864952@198.23.160.81>;tag=1b3b463c. > Call-ID: NjhmYzk0ODZhNWVmOGE5MTU4ZWExM2Q5NjI1NDVhODg. > CSeq: 2 INVITE. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, > SUBSCRIBE, INFO. > Content-Type: application/sdp. > Proxy-Authorization: Digest > username="639178864952",realm="198.23.160.81",nonce="51f67b16000000fae8c70592af760c78fa39cd43f84c1a6b",uri=" > sip:639195015475@198.23.160.81 > ",response="1605e4e6b20dbbd6be5917c94dd3961e",algorithm=MD5. > Supported: replaces. > User-Agent: X-Lite release 4.5.3 stamp 70576. > Content-Length: 357. > . > v=0. > o=- 1375107832761441 1 IN IP4 192.168.0.107. > s=X-Lite 4 release 4.5.3 stamp 70576. > c=IN IP4 192.168.0.107. > t=0 0. > m=audio 53084 RTP/AVP 123 9 0 8 97 100 98 101. > a=rtpmap:123 opus/48000/2. > a=fmtp:123 useinbandfec=1. > a=rtpmap:97 speex/8000. > a=rtpmap:100 speex/16000. > a=rtpmap:98 ILBC/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-15. > a=sendrecv. > > > U 2013/07/29 18:23:53.318191 198.23.160.81:5060 -> 49.144.184.97:41998 > SIP/2.0 100 Giving a try. > Via: SIP/2.0/UDP 192.168.0.107:41998 > ;received=49.144.184.97;branch=z9hG4bK-d8754z-1cfc096228010333-1---d8754z-;rport=41998. > To: <sip:639195015475@198.23.160.81>. > From: "639178864952"<sip:639178864952@198.23.160.81>;tag=1b3b463c. > Call-ID: NjhmYzk0ODZhNWVmOGE5MTU4ZWExM2Q5NjI1NDVhODg. > CSeq: 2 INVITE. > Server: OpenSIPS (1.8.3-notls (x86_64/linux)). > Content-Length: 0. > . > > > U 2013/07/29 18:24:03.328456 198.23.160.81:5060 -> 49.144.184.97:41998 > SIP/2.0 477 Send failed (477/TM). > Via: SIP/2.0/UDP 192.168.0.107:41998 > ;received=49.144.184.97;branch=z9hG4bK-d8754z-1cfc096228010333-1---d8754z-;rport=41998. > To: <sip:639195015475@198.23.160.81 > >;tag=eb0b3ccde0b882baee99bc071578cb61-d8d9. > From: "639178864952"<sip:639178864952@198.23.160.81>;tag=1b3b463c. > Call-ID: NjhmYzk0ODZhNWVmOGE5MTU4ZWExM2Q5NjI1NDVhODg. > CSeq: 2 INVITE. > Server: OpenSIPS (1.8.3-notls (x86_64/linux)). > Content-Length: 0. > . > > > U 2013/07/29 18:24:03.582094 49.144.184.97:41998 -> 198.23.160.81:5060 > ACK sip:639195015475@198.23.160.81 SIP/2.0. > Via: SIP/2.0/UDP 192.168.0.107:41998 > ;branch=z9hG4bK-d8754z-1cfc096228010333-1---d8754z-;rport. > Max-Forwards: 70. > To: <sip:639195015475@198.23.160.81 > >;tag=eb0b3ccde0b882baee99bc071578cb61-d8d9. > From: "639178864952"<sip:639178864952@198.23.160.81>;tag=1b3b463c. > Call-ID: NjhmYzk0ODZhNWVmOGE5MTU4ZWExM2Q5NjI1NDVhODg. > CSeq: 2 ACK. > Content-Length: 0.**** > > **** > > ** ** > > > Thanks, > Jason**** > > ** ** > > On Mon, Jul 29, 2013 at 3:33 PM, Gertjan Wolzak <g.wolzak(a)foize.com> > wrote:**** > > Hello Jason,**** > > **** > > What you should do is on the Kamailio server do an ngrep, that > information will make it possible for the people on the list to take a look > and help you.**** > > **** > > By only telling your experiences it is hard to help you.**** > > **** > > Just give the command: ngrep –qt –W byline port 5060 and then do the > test calls, so from android to xlite and vice versa.**** > > **** > > Save the info and paste it in the case.**** > > **** > > Good luck.**** > > **** > > Gertjan**** > > **** > > *From:* sr-users-bounces(a)lists.sip-router.org [mailto: > sr-users-bounces(a)lists.sip-router.org] *On Behalf Of *Jason Sia > *Sent:* zondag 28 juli 2013 10:52 > *To:* sr-users(a)lists.sip-router.org > *Subject:* [SR-Users] kamailio call problem**** > > **** > > Hi,**** > > I installed kamailio v 3.1.6 rpm. I used the default configuration. > Kamailio started. I have two clients one is an android phone using native > sip client, and the other one x-lite. I can call x-lite to phone but not > the other way around. When I restarted kamailio, I can call phone to > x-lite but not the other way around. The first one I did when I restart > kamailio will be the only leg that will be working, then I have to restart > again if I want to do the reverse. Has anyone encountered this problem? > How do I fix it? I'm on a LAN, kamailio server and the two clients are on > the same LAN.**** > > **** > > Thanks, > Jason**** > > > _______________________________________________ > 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**** > > ** ** > > _______________________________________________ > 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 > >

2 1

My Account is Disabled and I never logged on once
by Ammon Sarver 31 Jul '13

31 Jul '13

I just made an account but for some reason it doesn't let me sign in. Also your ssl certifcate is expired. Ammon

2 1

Kamailio4.0 + Siremis4.0
by Premchandiran 31 Jul '13

31 Jul '13

Hi Lucky Santiago, May I know how installed php with pdo-mysql. I kindly request to share steps to install php with pdo-mysql. Regards, Prem Chandiran M -----Original Message----- From: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of sr-users-request(a)lists.sip-router.org Sent: Wednesday, July 31, 2013 03:30 PM To: sr-users(a)lists.sip-router.org Subject: sr-users Digest, Vol 98, Issue 128 Send sr-users mailing list submissions to sr-users(a)lists.sip-router.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users or, via email, send a message with subject or body 'help' to sr-users-request(a)lists.sip-router.org You can reach the person managing the list at sr-users-owner(a)lists.sip-router.org When replying, please edit your Subject line so it is more specific than "Re: Contents of sr-users digest..." Today's Topics: 1. Kamailio4.0 + Siremis4.0 (Lucky Santiago) 2. siremis login issue (Premchandiran) 3. websocket double record_route() (Vitaliy Aleksandrov) ---------------------------------------------------------------------- Message: 1 Date: Wed, 31 Jul 2013 14:57:58 +0800 From: Lucky Santiago <lucky(a)luckysantiago.com> To: sr-users(a)lists.sip-router.org Subject: [SR-Users] Kamailio4.0 + Siremis4.0 Message-ID: <51F8B576.8060306(a)luckysantiago.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Sorry for being a newbie, I have newly installed Kamailio 4.0 + Siremis 4.0 and i have setup until properly registering extension/subscriber, what i want to attain for now is a simple exten -> kamailio/siremis -> carrier/telco that includes dialplan and trunking, it will really help if there is some sort of documentation managing it via the web interface or siremis. Thanks. ------------------------------ Message: 2 Date: Wed, 31 Jul 2013 12:46:39 +0530 From: "Premchandiran" <premchandiran.marimuthu(a)plintron.com> To: <sr-users(a)lists.sip-router.org> Subject: [SR-Users] siremis login issue Message-ID: <001901ce8dbd$e67dc2e0$b37948a0$@marimuthu@plintron.com> Content-Type: text/plain; charset="us-ascii" Hi All, After installing in siremis when I try to login using admin and admin as username and pwd a pop window opens in that "SQLSTATE[HY000] [2002] No such file or directory" . and in error log of apache2 I am getting below error " [client 192.168.137.7:4758] AH01630: client denied by server configuration: /opt/siremis-4.0.0/siremis/.htaccess" I kindly request you all to help me in resolving this issue. Thanks and Regards, Prem Chandiran M

1 0

siremis login issue
by Premchandiran 31 Jul '13

31 Jul '13

Hi All, After installing in siremis when I try to login using admin and admin as username and pwd a pop window opens in that "SQLSTATE[HY000] [2002] No such file or directory" . and in error log of apache2 I am getting below error " [client 192.168.137.7:4758] AH01630: client denied by server configuration: /opt/siremis-4.0.0/siremis/.htaccess" I kindly request you all to help me in resolving this issue. Thanks and Regards, Prem Chandiran M

1 0

Kamailio4.0 + Siremis4.0
by Lucky Santiago 31 Jul '13

31 Jul '13

Sorry for being a newbie, I have newly installed Kamailio 4.0 + Siremis 4.0 and i have setup until properly registering extension/subscriber, what i want to attain for now is a simple exten -> kamailio/siremis -> carrier/telco that includes dialplan and trunking, it will really help if there is some sort of documentation managing it via the web interface or siremis. Thanks.

1 0

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sr-users July 2013