- sr-users - kamailio.org

Deleting the expired contacts using the Rtimer module
by kiran bhosale 06 Feb '13

06 Feb '13

Hi we have developed the custom module which stores the registered users in a file.now we are trying to remove the expired contacts using the rtimer module.while saving the registered users to the file we also store the expires values in static array.but when we try to decrement the these values in a function called with the help of rtimer module. the values used by this periodic function are not modified ones but the initial which are zero. is it that we cant pass the modified values of the global variables to the timed functions. to get around this problem, we also registered the timer in our module but got same results !!!!!!! Please Help. Thanks and regards .

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Fwd:
by jose luis 05 Feb '13

05 Feb '13

http://www.lacucinalibanese.altervista.org/rjugo4.php?s=ot

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Syslog
by Vitaliy Aleksandrov 05 Feb '13

05 Feb '13

Hi list. I have a strange problem with syslog. Kamailio doesn't send any messages to syslog from UDP workers that are bound to sock=127.0.0.1:5060. Neither LM_DBG() from a source code nor xlog("any level", "") works. I use kamailio-3.2.4 with this opions: log_stderror=no debug=3. With log_stderror=yes kamailio writes all debug messages to stderr without any problems.

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modifying global data-structures using Rtimer module
by Kiran Bhosale 05 Feb '13

05 Feb '13

Hi we have developed the custom module which stores the registered users in a file.now we are trying to remove the expired contacts using the rtimer module.while saving the registered users to the file we also store the expires values in static array.but when we try to decrement the these values in a function called with the help of rtimer module. the values used by this periodic function are not modified ones but the initial which are zero. is it that we cant pass the modified values of the global variables to the timed functions. to get around this problem, we also registered the timer in our module but got same results !!!!!!! Please Help. Thanks and regards .

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Refreshing the list of submitted patches
by Daniel-Constantin Mierla 05 Feb '13

05 Feb '13

Hello, following some discussion over the weekend at Fosdem, it seems that some of submitted patches got forgotten in the mailing list archives and even on tracker. If any of you submitted patches was not accepted or rejected for a persistent reason, please write again to sr-dev mailing list referring to it. It is a short period still good for getting them in v4.0. In general, if there is no more activity regarding a patch for one or two weeks, come back and ask for an update. We don't mind reminders (especially because many of us do have a heavy loaded traveling schedule). Cheers and happy testing upcoming v4.0, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

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Best way to 'detect' direction
by Paul Belanger 04 Feb '13

04 Feb '13

Greetings, I'm in the process of adding logic to 'detect' the direction of my message, I have to set: rtpproxy_manage("ei") or rtpproxy_manage("ie") respectively. Rather then hard-coding IP addresses in my configuration, I was curious how other people over come this? Perhaps tagging their call control in some other method? -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belanger(a)polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger

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50000+ subscribers on Kamailio?
by Алексей Ёлохов 04 Feb '13

04 Feb '13

Hi! I am choosing a softswitch to serve more than 50000 (up to 100000) active subscribers (each subscriber is using SIP registration). Please tell me , I want to understand can kamailio work with such load? --- Alex Elokhov

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Remove dash (-) from called number
by Ali Jawad 04 Feb '13

04 Feb '13

Dear All I need to remove any dashes wherever they do occur in a called number, any guidance please ? Regards * *

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Re: [SR-Users] sr-users Digest, Vol 92, Issue 80
by Ian French 04 Feb '13

04 Feb '13

Thank you very much Sammy that worked Ian On 29 January 2013 18:14, <sr-users-request(a)lists.sip-router.org> wrote: > 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. Re: Asterisk and dispatcher (Ian French) > 2. Re: Asterisk and dispatcher (SamyGo) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Tue, 29 Jan 2013 17:55:40 +0000 > From: Ian French <fretec(a)gmail.com> > Subject: Re: [SR-Users] Asterisk and dispatcher > To: sr-users(a)lists.sip-router.org > Message-ID: > <CAOAGGZSr_24VDL8508XJo9LjkvR= > uo67JoM_MPPyd3bVbWxC7Q(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > On 24 January 2013 08:16, <sr-users-request(a)lists.sip-router.org> wrote: > > > 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. Re: Asterisk and dispatcher (SamyGo) > > 2. Re: Asterisk and dispatcher (SamyGo) > > 3. Kamailio Multi tenant as registrar server ony (andrea mucci) > > 4. help me about siremis. (???) > > > > > > ---------------------------------------------------------------------- > > > > Message: 1 > > Date: Wed, 23 Jan 2013 22:26:26 -0800 > > From: SamyGo <govoiper(a)gmail.com> > > Subject: Re: [SR-Users] Asterisk and dispatcher > > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > > Users Mailing List" <sr-users(a)lists.sip-router.org> > > Message-ID: > > < > > CAJUJwtjjg+iGH6+QuDtRkEH+mn+dkiObUFcPyNMKTv-0xHnYMQ(a)mail.gmail.com> > > Content-Type: text/plain; charset="iso-8859-1" > > > > Hi, > > > > You've metioned this already in your code: > > > > if(ds_is_from_list("1")) > > > > Is this not working for you? > > > > Regards, > > Sammy > > > > > > > > On Wed, Jan 23, 2013 at 10:44 AM, Ian French <fretec(a)gmail.com> wrote: > > > > > Hi, > > >> I've been working my way through this tutorial ( > > >> > > > http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb > > ) > > >> and all it working well thanks. In the other benifits section you > > mention > > >> that this configuration can be used with multiple instances of > Asterisk. > > >> I've configured my [TOASTERISK] route > > > > > > > > > route[TOASTERISK] { > > > ds_select_dst("1","4"); > > > $sht(forw=>$ft)=$du; > > > sl_send_reply("100","Trying"); > > > xlog("L_INFO","INVITE: INVITE [$tU] Passed to [$du] Asterisk > > > Cluster [$rm:$au].\n"); > > > route(RELAY); > > > exit; > > > > > > How can i achieve the same for my [FROMASTERISK] route as now with > > > > > > asterisk.bindip = "x.x.x.x" desc "Asterisk IP Address" > > > asterisk.bindport = "5060" desc "Asterisk Port" > > > > > > and > > > > > > route[FROMASTERISK] { > > > if($si==$sel(cfg_get.asterisk.bindip) > > > && $sp==$sel(cfg_get.asterisk.bindport)) > > > xlog("L_INFO","INVITE: INVITE Passed From [$fu] > Asterisk > > > Cluster To [$rm:$au].\n"); > > > return 1; > > > return -1; > > > } > > > > > > This allows calls only from cfg_get.asterisk.bindip (single gateway) > > where > > > I would like gateways in the dispatcher list to be selected and marked > > as a > > > positive match > > > > > > if(ds_is_from_list("1")) > > > $sht(forw=>$ft)=$si; > > > xlog("L_INFO","INVITE: INVITE Passed From [$fu] > Asterisk > > > Cluster To [$rm:$au].\n"); > > > return 1; > > > return -1; > > > } > > > > > > Can you help? > > > > > > Thanks in advance > > > Ian > > > _______________________________________________ > > > 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 > > > > > > > >

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Voip Issue
by BrianSuper7 04 Feb '13

04 Feb '13

Spam detection software, running on the system "mail.iptel.org", has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: Hi I have been using your excellent Voip facility for sometime. Recently I have encountered that when I called from 'phone A' to 'Phone B' connects and can hear Phone B's voice but can't hear phone A's voice when dialed from phone A. But when Phone B made the call to phone A, can hear both parties voices. This problem started very recently. I haven't done any configuration changes. This problem incurred nearly a week or two ago. I experienced same problem long time ago and it resolved after few days time. But this time it didn't resolved. My SIP no 87415661. I would appreciate if you could kindly help me to rectify the problem. [...] Content analysis details: (5.2 points, 5.0 required) pts rule name description ---- ---------------------- -------------------------------------------------- -0.0 RCVD_IN_DNSWL_NONE RBL: Sender listed at http://www.dnswl.org/, no trust [220.233.0.17 listed in list.dnswl.org] 0.0 FREEMAIL_FROM Sender email is commonly abused enduser mail provider (briansuper7[at]yahoo.com.au) 0.2 FREEMAIL_REPLYTO_END_DIGIT Reply-To freemail username ends in digit ("briansuper7" <briansuper7[at]yahoo.com.au> ) 0.2 FREEMAIL_ENVFROM_END_DIGIT Envelope-from freemail username ends in digit (briansuper7[at]yahoo.com.au) 0.0 HTML_MESSAGE BODY: HTML included in message 0.8 BAYES_50 BODY: Bayes spam probability is 40 to 60% [score: 0.4955] 0.6 REPTO_QUOTE_YAHOO Yahoo! doesn't do quoting like this 3.3 AXB_XMAILER_MIMEOLE_OL_4379D AXB_XMAILER_MIMEOLE_OL_4379D The original message was not completely plain text, and may be unsafe to open with some email clients; in particular, it may contain a virus, or confirm that your address can receive spam. If you wish to view it, it may be safer to save it to a file and open it with an editor.

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issue with #<number> calls
by Ivan Milivojevic 03 Feb '13

03 Feb '13

Hi Andrew, Here is one call with special characters: call-id = 100c8a98-c0a8016b-13c4-50029-203fa-68e78324-203fa How we can block such attempts, by rewrite rules? Regards Ivan

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Build 3.3 on ubuntu 12.10
by Gareth Rylance 01 Feb '13

01 Feb '13

Hi I am trying to build the kamailio deb packages on ubuntu 12.04. when I run make deb I get dpkg-checkbuilddeps: Unmet build dependencies: libmono-dev but libmono-dev is not available on 12.04 how do I 1. Fix the dependancy 2. Ignore it as I am not using app_mono (although I have an idea that may in the future) Thanks Gareth

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loose_route causing loop
by Renan Capaverde 31 Jan '13

31 Jan '13

Hello, I have a problem with my configuration file. When I receive a in-dialog request (REINVITE) with topmost route being my kamailio server, it enters in loop and then the call is dropped. What is the easiest way to prevent this to happen? Best Regards, -- *Renan Capaverde* /Estagiário Nível Superior/ /DDT - STE - Soluções em Tecnologia e Embarcados/ *DÍGITRO TECNOLOGIA* *E-mail:* renan.capaverde(a)digitro.com.br <mailto:renan.capaverde@digitro.com.br> *Fone:* +55 48 3281-7000*Ramal:* 8132 *Fax:* +55 48 3281-7299 *Site:* www.digitro.com <http://www.digitro.com> /"Antes de imprimir, pense na sua responsabilidade e no seu compromisso com o meio ambiente"/ Esta mensagem, incluindo seus anexos, é reservada somente à Dígitro e ao destinatário da mensagem. Caso você tenha recebido esta mensagem por engano, queira por favor, retorná-la ao remetente e apagá-la de seus arquivos.

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Re: [SR-Users] Error searching $keepalive.socket pvar in NAT Traversal module
by vito 31 Jan '13

31 Jan '13

Hello, I have: # Kamailio (OpenSER) SIP Server v3.3 # Kamailio (OpenSER) SIP Server v3.0 Defining pv in this format works well: $fs = $(keepalive.socket($du)); Thanks a lot !! Vito

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BasicKamailio DB question
by Jason 31 Jan '13

31 Jan '13

Hi all, excuse the basic nature of my question, I'm just starting out. I've read in a tutorial that it is possible to connect from asterisk to Kamailio and for asterisk to use the Kamailio db for authentication. However, I also read that it is possible to connect to astereisk from Kamailio for retrieving auth data. My question is this... If Kamailio uses the db on a remote asterisk server to retrieve usernames and password info how do you set Kamailio to use multiple databases as you may ultimately have multiple asterisk servers? As far as I could see there is only 1 db typically in the config file. Also, which is the most common configuration and why would one choose one over the other? Thanks in advance.

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REGISTER OK but on send "407 Proxy Authentication Required" with no ACK
by Carsten Maass 30 Jan '13

30 Jan '13

Hi all, I am trying to set up a fax gateway in the following way: PSTN-GW (10.1.1.150) <--> Kamailio (10.1.1.148:5123) <--> t38modem (10.1.1.148:6050) <--> Hylafax PSTN-GW is a standalone Berofix appliance and Kamailio 3.3.3, t38modem 1.2 and Hylafax 6.0.3 running on the same host under Redhat EL6. I used the default kamailio.cfg and just adjusted the PSTN route pattern to "if(!($rU=~"^(\+|00|0)[1-9][0-9]{3,20}$"))", to route all calls starting with 0 to the PSTN-GW. Both PSTN-GW and t38modem Successfully register to Kamailio but when I try to send out a fax from t38modem, Kamailio doesn't ACK the Proxy-Authorization request and t38modem terminates the call: <siptrace> 05:49:11.158157 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 986) 10.1.1.148.6050 > 10.1.1.148.5123: UDP, length 958 E.....@.@.Y. .d. .d.........INVITE sip:030987654321@10.1.1.148:5123 SIP/2.0 Route: <sip:10.1.1.148:5123;lr> Date: Tue, 29 Jan 2013 04:49:11 GMT CSeq: 1 INVITE Via: SIP/2.0/UDP 10.1.1.148:6050;branch=z9hG4bK020144e4-3c68-e211-98e4-005056bd002b;rport User-Agent: T38Modem/1.2.0 From: "root" <sip:030123456789@10.1.1.148:5123>;tag=169d43e4-3c68-e211-98e4-005056bd002b Call-ID: 16a243e4-3c68-e211-98e4-005056bd002b(a)myhost.mydomain.local Organization: Vyacheslav Frolov To: <sip:030987654321@10.1.1.148:5123> Contact: <sip:030123456789@10.1.1.148:6050> Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING Content-Type: application/sdp Content-Length: 262 Max-Forwards: 70 v=0 o=- 1359434951 1 IN IP4 10.1.1.148 s=Opal SIP Session c=IN IP4 10.1.1.148 t=0 0 m=audio 5008 RTP/AVP 8 101 100 a=sendrecv a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16,32,36 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 05:49:11.158927 IP (tos 0x10, ttl 64, id 28825, offset 0, flags [none], proto UDP (17), length 584) 10.1.1.148.5123 > 10.1.1.148.6050: UDP, length 556 E..Hp...@.*. .d. .d......4.oSIP/2.0 407 Proxy Authentication Required CSeq: 1 INVITE Via: SIP/2.0/UDP 10.1.1.148:6050;branch=z9hG4bK020144e4-3c68-e211-98e4-005056bd002b;rport=6050 From: "root" <sip:030123456789@10.1.1.148:5123>;tag=169d43e4-3c68-e211-98e4-005056bd002b Call-ID: 16a243e4-3c68-e211-98e4-005056bd002b(a)myhost.mydomain.local To: <sip:030987654321@10.1.1.148:5123>;tag=75c88a497c1ef184e8ac0d2e60c130e4.0580 Proxy-Authenticate: Digest realm="10.1.1.148", nonce="UQdV81EHVMeoOdE8gs8P4V1EFJAb79/W" Server: kamailio (3.3.3 (x86_64/linux)) Content-Length: 0 05:49:11.160273 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 501) 10.1.1.148.6050 > 10.1.1.148.5123: UDP, length 473 E.....@.@.[. .d. .d.........ACK sip:030987654321@10.1.1.148:5123 SIP/2.0 Route: <sip:10.1.1.148:5123;lr> CSeq: 1 ACK Via: SIP/2.0/UDP 10.1.1.148:6050;branch=z9hG4bK020144e4-3c68-e211-98e4-005056bd002b;rport From: "root" <sip:030123456789@10.1.1.148:5123>;tag=169d43e4-3c68-e211-98e4-005056bd002b Call-ID: 16a243e4-3c68-e211-98e4-005056bd002b(a)myhost.mydomain.local To: <sip:030987654321@10.1.1.148:5123>;tag=75c88a497c1ef184e8ac0d2e60c130e4.0580 Content-Length: 0 Max-Forwards: 70 05:49:11.166542 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 1195) 10.1.1.148.6050 > 10.1.1.148.5123: UDP, length 1167 E.....@.@.Y. .d. .d.........INVITE sip:030987654321@10.1.1.148:5123 SIP/2.0 Route: <sip:10.1.1.148:5123;lr> Date: Tue, 29 Jan 2013 04:49:11 GMT CSeq: 2 INVITE Via: SIP/2.0/UDP 10.1.1.148:6050;branch=z9hG4bK146145e4-3c68-e211-98e4-005056bd002b;rport User-Agent: T38Modem/1.2.0 From: "root" <sip:030123456789@10.1.1.148:5123>;tag=169d43e4-3c68-e211-98e4-005056bd002b Call-ID: 16a243e4-3c68-e211-98e4-005056bd002b(a)myhost.mydomain.local Organization: Vyacheslav Frolov To: <sip:030987654321@10.1.1.148:5123> Contact: <sip:030123456789@10.1.1.148:6050> Proxy-Authorization: Digest username="979", realm="10.1.1.148", nonce="UQdV81EHVMeoOdE8gs8P4V1EFJAb79/W", uri="sip:030987654321@10.1.1.148:5123", algorithm=MD5, response="48d896ef41a71720b29c56230858582f" Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING Content-Type: application/sdp Content-Length: 262 Max-Forwards: 70 v=0 o=- 1359434951 1 IN IP4 10.1.1.148 s=Opal SIP Session c=IN IP4 10.1.1.148 t=0 0 m=audio 5008 RTP/AVP 8 101 100 a=sendrecv a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16,32,36 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 05:49:11.167573 IP (tos 0x10, ttl 64, id 28826, offset 0, flags [none], proto UDP (17), length 584) 10.1.1.148.5123 > 10.1.1.148.6050: UDP, length 556 E..Hp...@.*. .d. .d......4.oSIP/2.0 407 Proxy Authentication Required CSeq: 2 INVITE Via: SIP/2.0/UDP 10.1.1.148:6050;branch=z9hG4bK146145e4-3c68-e211-98e4-005056bd002b;rport=6050 From: "root" <sip:030123456789@10.1.1.148:5123>;tag=169d43e4-3c68-e211-98e4-005056bd002b Call-ID: 16a243e4-3c68-e211-98e4-005056bd002b(a)myhost.mydomain.local To: <sip:030987654321@10.1.1.148:5123>;tag=75c88a497c1ef184e8ac0d2e60c130e4.263a Proxy-Authenticate: Digest realm="10.1.1.148", nonce="UQdV81EHVMeoOdE8gs8P4V1EFJAb79/W" Server: kamailio (3.3.3 (x86_64/linux)) Content-Length: 0 05:49:11.169220 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 710) 10.1.1.148.6050 > 10.1.1.148.5123: UDP, length 682 E.....@.@.Z. .d. .d.........ACK sip:030987654321@10.1.1.148:5123 SIP/2.0 Route: <sip:10.1.1.148:5123;lr> CSeq: 2 ACK Via: SIP/2.0/UDP 10.1.1.148:6050;branch=z9hG4bK146145e4-3c68-e211-98e4-005056bd002b;rport From: "root" <sip:030123456789@10.1.1.148:5123>;tag=169d43e4-3c68-e211-98e4-005056bd002b Call-ID: 16a243e4-3c68-e211-98e4-005056bd002b(a)myhost.mydomain.local To: <sip:030987654321@10.1.1.148:5123>;tag=75c88a497c1ef184e8ac0d2e60c130e4.263a Proxy-Authorization: Digest username="979", realm="10.1.1.148", nonce="UQdV81EHVMeoOdE8gs8P4V1EFJAb79/W", uri="sip:030987654321@10.1.1.148:5123", algorithm=MD5, response="3056b266b15a92f5cc8cc8e1c073fbcb" Content-Length: 0 Max-Forwards: 70 </siptrace> What do I have to adjust to make this work? Any help and pointers highly appreciated. Thanks in advance and greetings, Carsten. -- Blinkenlichten (Maass & Sacha GbR) - Open Source Solutions Weigandufer 45 - 12059 Berlin - http://www.blinkenlichten.de FON: ++49 +30 13896247 - MAIL: cm(a)blinkenlichten.de FAX: ++49 +30 13896249 - PGP: Key Id 0x2CBCA806 St.Nr. 16/274/61636

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Re: [SR-Users] Asterisk and dispatcher
by Ian French 29 Jan '13

29 Jan '13

On 24 January 2013 08:16, <sr-users-request(a)lists.sip-router.org> wrote: > 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. Re: Asterisk and dispatcher (SamyGo) > 2. Re: Asterisk and dispatcher (SamyGo) > 3. Kamailio Multi tenant as registrar server ony (andrea mucci) > 4. help me about siremis. (???) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 23 Jan 2013 22:26:26 -0800 > From: SamyGo <govoiper(a)gmail.com> > Subject: Re: [SR-Users] Asterisk and dispatcher > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: > < > CAJUJwtjjg+iGH6+QuDtRkEH+mn+dkiObUFcPyNMKTv-0xHnYMQ(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Hi, > > You've metioned this already in your code: > > if(ds_is_from_list("1")) > > Is this not working for you? > > Regards, > Sammy > > > > On Wed, Jan 23, 2013 at 10:44 AM, Ian French <fretec(a)gmail.com> wrote: > > > Hi, > >> I've been working my way through this tutorial ( > >> > http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb > ) > >> and all it working well thanks. In the other benifits section you > mention > >> that this configuration can be used with multiple instances of Asterisk. > >> I've configured my [TOASTERISK] route > > > > > > route[TOASTERISK] { > > ds_select_dst("1","4"); > > $sht(forw=>$ft)=$du; > > sl_send_reply("100","Trying"); > > xlog("L_INFO","INVITE: INVITE [$tU] Passed to [$du] Asterisk > > Cluster [$rm:$au].\n"); > > route(RELAY); > > exit; > > > > How can i achieve the same for my [FROMASTERISK] route as now with > > > > asterisk.bindip = "x.x.x.x" desc "Asterisk IP Address" > > asterisk.bindport = "5060" desc "Asterisk Port" > > > > and > > > > route[FROMASTERISK] { > > if($si==$sel(cfg_get.asterisk.bindip) > > && $sp==$sel(cfg_get.asterisk.bindport)) > > xlog("L_INFO","INVITE: INVITE Passed From [$fu] Asterisk > > Cluster To [$rm:$au].\n"); > > return 1; > > return -1; > > } > > > > This allows calls only from cfg_get.asterisk.bindip (single gateway) > where > > I would like gateways in the dispatcher list to be selected and marked > as a > > positive match > > > > if(ds_is_from_list("1")) > > $sht(forw=>$ft)=$si; > > xlog("L_INFO","INVITE: INVITE Passed From [$fu] Asterisk > > Cluster To [$rm:$au].\n"); > > return 1; > > return -1; > > } > > > > Can you help? > > > > Thanks in advance > > Ian > > _______________________________________________ > > 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

Re: [SR-Users] Default 3.3 config and dispatcher
by Daniel Tryba 29 Jan '13

29 Jan '13

On Tuesday 29 January 2013 16:58:08 Paul Belanger wrote: > > [kamailio] > > host=192.168.1.1 > > context=incoming > > type=friend > > allow=all > > qualify=yes > > insecure=port,invite > > nat=no > > canreinvite=yes > > > > Nothing more to it. > > Thanks for pointing me in the right direction. Will provide feedback > shortly. One more thing I might have forgotten. Asterisk needs to responde with a 200 OK to the OPTIONS requests in order for Kamailio to think the dispatcher route is up and runnnig. So add an s exten in the asterisk context: [incoming] exten => s,1,NoOp() -- POCOS B.V. - Croy 9c - 5653 LC Eindhoven Telefoon: 040 293 8661 - Fax: 040 293 8658 http://www.pocos.nl/ - http://www.sipo.nl/ K.v.K. Eindhoven 17097024

1 0

Subscribe and PUA database
by Misha Ketslah 29 Jan '13

29 Jan '13

Hello, i'm trying to set BLF on my snom 320 phone to get notification on function key for certain extension as well registered against kamailio. Any other info i can provide except the following one and as well i want to exuse me, as i'm pretty noob to all this, and my question might be silly, but after hours in front of google i found no solution :) I always get that: after pua_set_publish() ** 3(7825) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=791 a=16 n=if 3(7825) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=780 a=41 n=isflagset 3(7825) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=799 a=16 n=if 3(7825) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=791 a=25 n=save 3(7825) ERROR: pua [send_publish.c:578]: New PUBLISH and no body found- invalid request 3(7825) ERROR: pua_usrloc [ul_publish.c:326]: while sending publish 3(7825) ERROR: pua_usrloc [ul_publish.c:327]: TEST: type & UL_CONTACT_UPDATE is 2, and error is -1 (^^^ This line was added by me into module as i saw related thread and the patch solution, though it never meets condition and fails to insert from there as well.) 3(7825) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=799 a=2 n=exit my configs are : in BLF i set <sip:666@SERV_IP;user=phone> in cfg i have alias="SERV_IP" listen=udp:SERV_IP port=5060 loadmodule "db_mysql.so" loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "mi_rpc.so" loadmodule "acc.so" loadmodule "auth.so" loadmodule "auth_db.so" loadmodule "alias_db.so" loadmodule "speeddial.so" ... modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 3) modparam("usrloc", "use_domain", MULTIDOMAIN) ... modparam("presence", "db_url", DBURL) modparam("presence", "subs_db_mode" , 3 ) #modparam("presence", "publ_cache", 0) modparam("presence", "expires_offset", 300) modparam("presence", "max_expires", 3600) #modparam("presence", "force_active", 1) modparam("presence", "db_update_period", 10) modparam("presence", "server_address", "sip:SERV_IP:5060") modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #modparam("presence_dialoginfo", "force_single_dialog", 1) modparam("pua", "db_url" ,DBURL ) modparam("pua", "db_mode", 2) modparam("pua", "db_table", "pua") #modparam("pua", "outbound_proxy", "sip:SERV_IP") modparam("pua", "min_expires", 10) modparam("pua", "default_expires", 3600) modparam("pua", "update_period", 60) modparam("pua_usrloc", "default_domain", "SERV_IP") request_route { if(method=="NOTIFY") { xlog(" pua_update_contact $rm from $fu (IP:$si:$sp) \n"); if(!pua_update_contact()) xlog("pua update failed \n"); } # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) t_check_trans(); # authentication route(AUTH); route(SPEEDDIAL); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); route(DISPATCH); # user location service route(LOCATION); route(RELAY); } route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|SUBSCRIBE")) { t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { xlog("pua_set_publish()\n"); if(!pua_set_publish()) xlog("set publish failed \n"); xlog(" after pua_set_publish()\n"); if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } #pua_set_publish(); if (!save("location")) { sl_reply_error(); xlog("save location failed \n"); } exit; } } # USER location service route[LOCATION] { #!ifdef WITH_SPEEDIAL if(uri=~"sip:[0-9]{2}@.*") xlog("speeddials\n"); # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") if(sd_lookup("speed_dial")) route(SIPOUT); #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) route(SIPOUT); #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") sd_lookup("speed_dial"); route(SPEEDDIAL); xlog("PUBLISH|SUBSCRIBE"); sl_send_reply("100","trying"); if (!t_newtran()) { sl_reply_error(); exit; }; append_to_reply("Contact: <sip:82.80.18.100:5060>\r\n"); if(is_method("PUBLISH")) { xlog("handle publish here\n"); handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle subscribe here \n"); handle_subscribe(); t_release(); } exit; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { add_contact_alias(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage(); if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } } if (is_reply()) { if(isbflagset(FLB_NATB)) { add_contact_alias(); } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { ....... return; } # XMLRPC routing route[XMLRPC] { .... } #!endif # route to voicemail server route[TOVOICEMAIL] { ...... return; } # manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); } # manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif } route[SPEEDDIAL] { ..... return; } ....

1 0

Web softphone using websockets against Kamailio
by Pirjo Ahvenainen 29 Jan '13

29 Jan '13

Greetings gurus! I'm playing with an idea to create a web based softphone (html5 + no installations for the end user) and use Kamailio's websocket module for backend. I'd love to hear about your comments, challenges and successes using such configuration. Is it a feasible way to construct a softphone even today when even IE9 does not support websockets, as such? I'm sure IE9 will end up in specs as a must-support platform. A collegue tried using sipml5 with webrtc against a SnomONE pbx (I know... ;)), and said there's no way it can work, but I'm not convinced the idea itself wouldn't work. It would help me lots if I could make a simple example using Kamailio with SIP over websockets, can you comment on how much effort do I need on Kamailio side to make this work? Do I need off-default config scripting, or is it enough to just set up the module and set the parameters? And even with the risk of stepping a little off topic, if anyone has worked on web based softphones, I'd love to hear if you can recommend on how to approach this. Cheers, Pirjo --

2 3

Default 3.3 config and dispatcher
by Paul Belanger 29 Jan '13

29 Jan '13

Greetings List, First post of many. I've managed to get up a running a very simple VoIP network with Kamailio as the core, and asterisk on the edge. Most of this has been accomplished using the default kamailio.cfg file shipped in 3.3. For my next adventure, I'd like to start using the dispatch module to route calls from Kamailio to asterisk, however having some trouble getting my syntax correct. I was hoping somebody could point me to an existing post / URL describing what I was asking, if it is based on the default configuration file, the better. I should note, if I have Asterisk register to kamailio, I can route calls fine however, I believe the dispatcher is the better way to do this. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belanger(a)polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger

2 1

Major problem with setup NAT + MULTIDOMAIN
by Rumen Mihailov 29 Jan '13

29 Jan '13

Hi guys, I've baning my head for the last 6 7 days with something that have to be universally simple but obviously for me it is not, so please do help. I have the following setup. xx.xx.xx.xx (real IP 1:1 NAT) -> 10.2.47.201 Kamailio listens on 10.2.47.201 I have setup a domain xxx.itradebg.com and I've made the appropriate records in the DNS. I have inserted the domain xxx.itradebg.com in the domain table in the database and kamctl domain show shows the correct thing. Now when I try to register 999(a)xxx.itradebg.com I get Request timed out I have this in my config: if (is_method("REGISTER") || from_uri==myself) { if (!auth_check("$fd", "sip", "1")) { auth_challenge("$fd", "0"); exit; } } if (from_uri!=myself && uri!=myself) { if(lookup_domain("$fd", "@from.uri.host")){ xlog("LOCAL DOMAIN -> REGISTER"); exit; } else{ xlog("L_WARN","FROM NOT LOCAL"); sl_send_reply("403","Sorry mate not relaying"); exit; } } By the time of writing this I realised that this should be all rewritten to this only, as the second check is useless: if (is_method("REGISTER") || from_uri==myself) { if (!auth_check("$fd", "sip", "1")) { auth_challenge("$fd", "0"); exit; } } However I get 401 Unauthorized. The previous setup was giving me 408 - Request Timed out... I am pretty sure I have the passwords correct. What I cannot understand is why authentication fails ?! Also, how do I set kamailio to print debug output ? debug=9 log_stderror=no This is what I have in the .conf file. /var/log/messages seems to only have what I have printed with xlog... (think) I am really out of ideas... perhaps I need some sleep. Anyway any input is helpful guys. Thanks, Rumen http://itradebg.com

2 4

Error searching $keepalive.socket pvar in NAT Traversal module
by vito 28 Jan '13

28 Jan '13

Hello, I started to use Kamailio few months ago. Maybe this is a stupid issue but I can´t start Kamailio with NAT TRAVERSAL module and $keepalive.socket pseudo-variable. My script configuration is the default template adding this lines (extracted from nat traversal doc): ... loadmodule "nat_traversal.so" ... ... $fs = $keepalive.socket($du); ... I have this error: -e 0(17455) ERROR: <core> [pvapi.c:501]: error searching pvar "keepalive" 0(17455) ERROR: <core> [pvapi.c:705]: wrong char [e/101] in [$keepalive] at [9 (0)] 0(17455) : <core> [cfg.y:3330]: parse error in config file /etc/kamailio/kamailio.cfg, line 410, column 8-17: unknown script pseudo variable $keepalive ERROR: bad config file (1 errors) I tried in this environments: Ubuntu 10.04.4 LTS x86_64 Kamailio 3.3.3+git20130128+lucid1 Ubuntu 10.04.4 LTS i686 Kamailio 3.0.4+lucid1 Ubuntu 12.04.1 LTS x86_64 Kamailio 3.3.2+precise1 All packages instaled from deb repositories (http://deb.kamailio.org/) I think there are problem parsing "$keepalive.socket" pseudovariable, because Kamailio doesn´t scape this character "." and it doesn´t recognise the pseudovar. Does anyone have this problem? Regards

2 1

Sip express live Cd router
by Shoukat, Kashif 28 Jan '13

28 Jan '13

Dear , Can you please provide me link of sip express live Cd router .. I want to use it... waiting your kind response Regards, Kashif Shoukat | Analyst Software Engineer | Work : + 1.202.289.9898 x 9077 Mobile : + 92.323.431.8548 Skype: kashif.shoukat.dgs www.dgsworld.com<http://www.dgsworld.com/> dgs deliberate by design This message contains information which may be confidential and privileged. Unless you are the addressee (or authorized to receive for the addressee), you may not use, copy or disclose to anyone the message or any information contained in the message. If you have received the message in error, please advise the sender by replying to kashif.shoukat(a)dgsworld.com<mailto:kashif.shoukat@dgsworld.com> and delete the message.

1 0

TLS certificate chain Android verification
by Nord7 28 Jan '13

28 Jan '13

Hello all. Our company buy Thawte SSL Web server certificate. I'm enabled in Kamailio 3.3 TLS support and configured him. Adding to "myhost.pem" certificate, two "intermediate" certificates (like in chain) "cat myhost.pem SSL_SecondaryCA.pem SSL_PrimaryCA.pem > chain-cert.pem". Remark: I use this chain on apache2. And Android trusted to this certificate chain. After this i use chain-cert.pem with KAMAILIO 3.3 and SIP client "BRIA Android". "Bria Android" says certificate is not trusted. Next step "BRIA IPHONE" and certificate chain works (trusted)! BRIA support says me - BRIA don't trusts certificate if system don't trusts. Why Android don't trusts to my chain-certificates with KAMAILIO? If his trust to same certificate with Apache? I'm understand this is not Kamailio problem, but maybe any one who have expirience in this problem help me? I know about bouncycastle, but this is not best decision for me. Maybe something else can help? I tryed this with Android 2.3.7, 4.0, 4.1. Best regards Kirill.

1 0

event_route for pre-dialog start
by Mino Haluz 28 Jan '13

28 Jan '13

Hi, I think that it is more than likely not implemented, but is there any event-route that is triggered just before event_route[dialog:start] ? I need to check some security flags before the dialog is created. But it is too early too check them in relay route.. Mino

1 0

Is kamailio creating multiple connections to db?
by Mino Haluz 28 Jan '13

28 Jan '13

Hi, I made some DB test on our HA cluster, and I got that for 1 process I have 1000 queries/sec and for 11 processes 3000 queries/sec. That's why I would to ask if every kamailio process that is spawned has its own database connection or queries are put in the queue and executed by some main process ? I would like to know this, so that I could assess maximum call count/sec that db can handle. Thank you, Mino

3 4

Generating logging based on SIP messages
by Grant Bagdasarian 28 Jan '13

28 Jan '13

Hello, A while ago I came across the siptrace and sipcapture modules and thought this would be a good way to generate logging out of all the SIP messages that are passed through Kamailio and stored in the siptrace/sipcapture table. My idea was to extract certain values from SIP messages and store these in our database. Also each SIP Request or Response would be mapped to a certain status so we would be able to monitor which call is in which state. I still have a lot of work to do to think this all out, since it's still only an idea. I'm just wondering if anyone could give me some pointers into what to keep in mind when doing this. Regards, Grant

2 2

Pseudo variable to get instance in the contact....
by Krishna Kurapati 28 Jan '13

28 Jan '13

Hi, I added in my local branch "$ct.inst" pseudo variable to retrieve instance. Since instance could be an important attribute to make routing decisions, having quick access to it in the message may be useful. Anyone else need this functionality? Krish Kura

3 2

iptel x media5-fone problem
by Kasper Hämäläinen 28 Jan '13

28 Jan '13

hi guys, hope u got time to help me. Im first timer testing VoIP and SIP. I created account to ur iptel.org site. I downloaded "Jitsi"-software. First when I opened it, I login with my username and password. Then I go to "add new account", give same login username and password, but Jitsi says that my password is wrong..wtf. And I installed Media5-Fone to my iPhone 4S. When adding SIP-account it's says again that I got wrong password. And YES, I didn't mistype it. Thanks - Kasper

2 1

CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg - Kamailio 3.3.x
by Rinor Hoxha 28 Jan '13

28 Jan '13

Hi list, I'm having some issues with dialog module. From time to time I get the following error: ERROR: dialog [dlg_req_within.c:231]: failed to update dialog lifetime CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg tl=0x2aeeea1c6840 tl->next=(nil) tl->prev=(nil) [root@proxy ~]# cat /var/log/kamailio.log | grep "Jan 17" | grep "tl=" | sort | uniq -c 1 Jan 17 13:27:00 proxy /usr/local/kamailio/sbin/kamailio[4100]: CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg tl=0x2aeeea09bf38 tl->next=(nil) tl->prev=(nil) 1 Jan 17 14:39:30 proxy /usr/local/kamailio/sbin/kamailio[4100]: CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg tl=0x2aeeea07c518 tl->next=(nil) tl->prev=(nil) 1 Jan 17 15:06:21 proxy /usr/local/kamailio/sbin/kamailio[4100]: CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg tl=0x2aeeea2da900 tl->next=(nil) tl->prev=(nil) 1 Jan 17 19:31:02 proxy /usr/local/kamailio/sbin/kamailio[4100]: CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg tl=0x2aeeea278cd0 tl->next=(nil) tl->prev=(nil) 1 Jan 17 19:31:42 proxy /usr/local/kamailio/sbin/kamailio[4100]: CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg tl=0x2aeee9f527a8 tl->next=(nil) tl->prev=(nil) 1 Jan 17 20:05:32 proxy /usr/local/kamailio/sbin/kamailio[4100]: CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg tl=0x2aeeea21fa30 tl->next=(nil) tl->prev=(nil) This one was flooding my log file: tl=0x2aeeea1c6840 24322 Jan 17 21:53:02 proxy /usr/local/kamailio/sbin/kamailio[4100]: CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg tl=0x2aeeea1c6840 tl->next=(nil) tl->prev=(nil) 5709 Jan 17 21:53:23 proxy /usr/local/kamailio/sbin/kamailio[4100]: CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg tl=0x2aeeea1c6840 tl->next=(nil) tl->prev=(nil) 61314 Jan 17 21:53:23 proxy /usr/local/kamailio/sbin/kamailio[4102]: CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg tl=0x2aeeea1c6840 tl->next=(nil) tl->prev=(nil) 22812 Jan 17 21:54:01 proxy /usr/local/kamailio/sbin/kamailio[4102]: CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg tl=0x2aeeea1c6840 tl->next=(nil) tl->prev=(nil) 21137 Jan 17 21:54:02 proxy /usr/local/kamailio/sbin/kamailio[4102]: CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg tl=0x2aeeea1c6840 tl->next=(nil) tl->prev=(nil) where pid 4100 is slow timer and pid 4102 is timer. Any idea why is this happening and how to fix it. (I suspect that may be some issues with our server memory...checking now...and will inform). However wanted to raise this issue in case anyone else is having the same. Thanks, Rinor

2 1

Pseudo variable to access SIP instance in contact
by Krishna Kurapati 28 Jan '13

28 Jan '13

Hi, Is there a way to directly access the SIP instance present in message from the script? Thanks Krish Kura

2 1

Kamailio behind NAT - best practice
by Klaus Darilion 27 Jan '13

27 Jan '13

Hi! Up to now I could avoid Kamailio setups with Kamailio behind NAt. But this time I have to deal with it. That's why I want to ask what others did as best practice. The scenario is: Asterisk 1----\ Kamailio+RTPPROXY \ |10.10.0.2 Asterisk n------\ | --------|------------- FW --SIP-trunk--- ITSP Freeswitch 1----/ 10.10.0.1 public-IP / Freeswitch n--/ 10.10.0.x Kamailio and rtpproxy have a private IP. Internal communication uses private IPs, external communication uses a public IP which is NATed 1:1 to Kamailio's IP address. No registrations, just forwarding of messages. Using the global "advertised_address" setting with the public IP does not work, as there is also internal communication. Using set_advertised_address() is also cumbersome. So it seems, the easiest solution would be to use 2 sockets on Kamailio, e.g. port 5050 and port 5060. Then I could use the listen with dedicated advertised addresses: listen=udp:10.10.0.2:5050 listen=udp:10.10.0.2:5060 advertise pu.bl.ic.ip:5060 If I understand it correctly, this should solve all issues with Record-Routing and Via-headers. For RTP-Proxy it seems necessary to detect the direction of each message and set the IP address in rtpproxy_manage("","ip.add.re.ss") manually. Thus, it seems straight forward - or do I miss something? Any comments and practical experience? Thanks Klaus

5 8

Pseudo variables and uid_domain module compatibility
by Philippe Sultan 25 Jan '13

25 Jan '13

Hi, According to docs of "uid_domain.so", $fd and $td are attribute groups which are set when calling the lookup_domain function, to retrieve domain attributes stored in an DB ($fd.did or $td.did) : lookup_domain("$fd", "@from.uri.host"); lookup_domain("$td", "@ruri.host"); if (strempty($fd.did) && strempty($td.did)) { # Neither the calling nor the called domain is local # This is a relaying attempt which should be forbidden sl_reply("403", "Relaying Forbidden"); drop; } But then, starting Kamailio fails with a configuration file error : 0(2098) ERROR: <core> [pvapi.c:761]: error searching pvar "fd" 0(2098) ERROR: <core> [pvapi.c:965]: wrong char [d/100] in [$fd] at [2 (0)] 0(2098) : <core> [cfg.y:3567]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 332, column 15-17: unknown script pseudo variable $fd ERROR: bad config file (1 errors) Line 332 is : if (strempty($fd.did) && strempty($td.did)) { I thought I could start Kamailio without "pv.so" (and use selects instead), but the error still shows up, as it is reported by the core pvapi.c file. So I'm kind of stuck, and asking help from the community to fix this problem :D Philippe

2 8

uac_replace_from recovery with modified header
by Andreas Granig 25 Jan '13

25 Jan '13

Hi, This is a question for kamailio 3.0.x and I'm still trying to reproduce this issue somehow for latest stable, but maybe it was a known issue anyways which got fixed in newer versions, so I don't have to dig deeper into this: There is an INVITE "A -> kamailio -> B", and kamailio does uac_replace_from() to replace the display- and user-part (leaving the domain-part untouched). When there's a BYE "B -> kamailio -> A", then From/To are switched, so the replaced From of the INVITE becomes the replaced To of the BYE. Kamailio gets that one right and tries to recover (in auto-mode) the To. However, in this BYE, B changes the domain-part of the To, which should be valid, as elements are only supposed to inspect the tags of From/To, right? The problem though is that when recovering the To, kamailio inserts garbage into the domain part. The vsf-Parameter in the Route of the BYE is definitely ok though, so that doesn't seem to be the issue. Here's an example: In the INVITE, From: "ab12345"<sip:ab12345@192.168.0.12;user=phone>;tag=948a3340 gets rewritten to From: "12345"<sip:12345@192.168.0.12;user=phone>;tag=948a3340 And in the BYE, To: "12345"<sip:12345@1.2.3.4>;tag=948a3340 gets rewritten to To: "12345"<sip:ab12345<@-%:68*0.0%12;user=phone>;tag=948a3340 Note that the received To in the BYE has a different host-part than what got sent out in the From of the INVITE, and when kamailio recovers the To, it doesn't recover the Display-part (not sure though if this is intended behaviour anyway), but it only party recovers the domain-part. You can still see some artefacts of the original domain (like the "68" from the second "168" octet, and the "12" from the last octet). Has anyone encountered this issue before? Andreas

4 12

help me about siremis.
by 문종민 25 Jan '13

25 Jan '13

I'm developers who trying to install siremis. will go in there and install siremis website. However, server-services -> domain list by pressing will get the following error. What can I do to fix this?

2 1

problem with db access
by MASebastian 25 Jan '13

25 Jan '13

Hi everyone, Recently I work on kamailio's snmpstats module. I'm trying to retrieve some data from database, the code pass the compilation but when starting kamailio always get this error. Jan 25 10:57:05 localhost kamailio: ERROR: <core> [sr_module.c:572]: ERROR: load_module: could not open module </usr/local/lib/kamailio/modules_k/snmpstats.so>: /usr/local/lib/kamailio/modules_k/snmpstats.so: undefined symbol: db_bind_mod I take the code from http://www.asipto.com/pub/kamailio-devel-guide/#c09database and ursloc module for example. not sure if I still miss something to make db access work. Any suggestion? Thanks, Sebastian

2 1

Websockets WSS problem with NOTIFY
by Pete Kelly 24 Jan '13

24 Jan '13

Hi, I am having an issue at the moment with SIP NOTIFY messages being sent from Kamailio (latest git master) over wss transport I am getting reports from the receiving end saying "Compressed bit must be 0 if no negotiated deflate-frame extension" The only reference I can find to it is at the following URL... where the problem was caused by the server miscalculating the size of the msg: http://stackoverflow.com/questions/12308728/compressed-bit-must-be-0-when-s… Does anyone have any suggestions as to how I could debug this within Kamailio? It sounds like Kamailio may be sending some incorrect packet information but I am unsure at this point.

3 14

sipcapture + db_postgres gives lots of warnings on PostgreSQL server
by Øyvind Kolbu 24 Jan '13

24 Jan '13

Hi, while trying to use sipcapture together with PostgreSQL I've found that it stores the "\r\n" in SIP messages as "\015\012", though not without a lot of noise in the PostgreSQL log files. For every line inserted in PostgreSQL I get: [0] WARNING: nonstandard use of \\ in a string literal at character 763 [0] HINT: Use the escape string syntax for backslashes, e.g., E'\\'. [0] STATEMENT: insert into sip_capture (date,micro_ts,method,reply_reason,ruri,ruri_user,from_user,from_tag,to_user,to_tag,pid_user,contact_user,auth_user,callid,callid_aleg,via_1,via_1_branch,cseq,reason,content_type,auth,user_agent,source_ip,source_port,destination_ip,destination_port,contact_ip,contact_port,originator_ip,originator_port,proto,family,rtp_stat,type,node,msg) values ('2013-01-24 15:27:43',1359037663218925,'OPTIONS','','sip:129.240.253.134','','keepalive','533cb9e91f4b999cf76861cbb9ed54ed-149d','','','','','','5001ed5d-6979(a)127.0.0.1','','SIP/2.0/UDP 129.240.253.197;branch=z9hG4bKae6f.d4a02ac7.0','z9hG4bKae6f.d4a02ac7.0','10 OPTIONS','','','','kamailio (3.3.3 (i386/linux))','129.240.253.197',5060,'129.240.253.134',5060,'',0,'',0,1,2,'',1,'voipt1:21','OPTIONS sip:129.240.253.134 SIP/2.0\\015\\012Via: SIP/2.0/UDP 129.240.253.197;branch=z9hG4bKae6f.d4a02ac7.0\\015\\012To: sip:129.240.253.134\\015\\012From: sip:keepalive@edgetest.voip.uio.no;tag=533cb9e91f4b999cf76861cbb9ed54ed-149d\\015\\012CSeq: 10 OPTIONS\\015\\012Call-ID: 5001ed5d-6979(a)127.0.0.1\\015\\012Content-Length: 0\\015\\012User-Agent: kamailio (3.3.3 (i386/linux))\\015\\012\\015\\012') Kamailio as of latest git on the 3.3 branc + 3e580e1ee24f9b9ec4b1fa151b23ebc84f3db1cb. Same warnings when linked with both PostgreSQL 8.4.13 and 9.1.1. According to Alexandr Dubovikov MySQL stores '\r\n' as '\r\n'. -- Øyvind Kolbu

1 0

Decode contact with user paramer
by Gareth Rylance 24 Jan '13

24 Jan '13

When using the siputils function encode_contact() on a message that contains a user parameter e.g. <sip:XXXXX00015;phone-context=national@10.YYY.1.92;user=phone>;tag=88930 it encodes fine e.g. <sip:natted_ua*sip:XXXXX00015;phone-context=national**192.168.242.102*5080*udp@10.YYY.70.100 > but decode_contact() errors due to the ";" (there is an explicit check for it in the code) I am currently working round this using $rU = $(rU{s.replace,;,:}); to remove the ; as the user part is not needed for my application. Would it be safe to remove the check for ";" ? Gareth

3 6

Kamailio Multi tenant as registrar server ony
by andrea mucci 24 Jan '13

24 Jan '13

Hi Gurus, i have a basic question. i have a SIP PBX server that don't have the Multi Tenant Feature or Multi Company Feature. i now that Kamailio support this feature, so is possible to use the kamailio as registrar server and redirect all the services, SUBSCRIBE, NOTIFY, and RTP to the SIP PBX with corrects headers? Thanks for your help

3 2

Asterisk and dispatcher
by Ian French 24 Jan '13

24 Jan '13

> > Hi, > I've been working my way through this tutorial ( > http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb) > and all it working well thanks. In the other benifits section you mention > that this configuration can be used with multiple instances of Asterisk. > I've configured my [TOASTERISK] route route[TOASTERISK] { ds_select_dst("1","4"); $sht(forw=>$ft)=$du; sl_send_reply("100","Trying"); xlog("L_INFO","INVITE: INVITE [$tU] Passed to [$du] Asterisk Cluster [$rm:$au].\n"); route(RELAY); exit; How can i achieve the same for my [FROMASTERISK] route as now with asterisk.bindip = "x.x.x.x" desc "Asterisk IP Address" asterisk.bindport = "5060" desc "Asterisk Port" and route[FROMASTERISK] { if($si==$sel(cfg_get.asterisk.bindip) && $sp==$sel(cfg_get.asterisk.bindport)) xlog("L_INFO","INVITE: INVITE Passed From [$fu] Asterisk Cluster To [$rm:$au].\n"); return 1; return -1; } This allows calls only from cfg_get.asterisk.bindip (single gateway) where I would like gateways in the dispatcher list to be selected and marked as a positive match if(ds_is_from_list("1")) $sht(forw=>$ft)=$si; xlog("L_INFO","INVITE: INVITE Passed From [$fu] Asterisk Cluster To [$rm:$au].\n"); return 1; return -1; } Can you help? Thanks in advance Ian

2 2

Guidelines to upgrade to 4.0.0
by Daniel-Constantin Mierla 23 Jan '13

23 Jan '13

Hello, the wiki page collecting the guidelines for upgrading from 3.3.x to 4.0.0 is available at: - http://www.kamailio.org/wiki/install/upgrade/3.3.x-to-4.0.0 I just added the SQL statements needed to upgrade the MySQL database, but there is lot of other stuff to be added. Hopefully, anyone noticing changes will add few notes and instructions about how to upgrade. The wiki can be edited if you make an account, it is not required to be a developer to add notes to this page. Would be great if users of postgres will add the sql statements specific for this database type. Thanks, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

Re: [SR-Users] sr-users Digest, Vol 92, Issue 50:Problem with 4.0 nightlies and IMS - sem_post undefined (Barry Flanagan)
by madhumanjusha＠integramicro.com 23 Jan '13

23 Jan '13

On Tue, January 22, 2013 1:58 pm, sr-users-request(a)lists.sip-router.org wrote: > 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. Modify IP address in SDP (Austin Einter) > 2. Re: Modify IP address in SDP (Olle E. Johansson) > 3. Problem with 4.0 nightlies and IMS - sem_post undefined > (Barry Flanagan) > 4. Patch to resolve segfault in json module (Richard Brady) > 5. Re: Modify IP address in SDP (Austin Einter) > 6. Re: Patch to resolve segfault in json module > (Daniel-Constantin Mierla) > 7. Re: [sr-dev] FOSDEM (Daniel Pocock) > > > > ---------------------------------------------------------------------- > > > > > > > > > > > Message: 3 > Date: Tue, 22 Jan 2013 13:19:44 +0000 > From: Barry Flanagan <barry(a)flanagan.ie> > Subject: [SR-Users] Problem with 4.0 nightlies and IMS - sem_post > undefined To: sr-users(a)lists.sip-router.org > Message-ID: > <CALJb54-rPPbjgGR5iEKYztNGDvT11znthJzgMYovpp_t6N0S8w(a)mail.gmail.com> > Content-Type: text/plain; charset="utf-8" > > > Hi > > > I am trying to run the Ubuntu nightlies from > http://deb.kamailio.org/kamailiodev-nightly and loading the p-cscf config > files. > > Things were working fine while I was on the 20130110 nightlies, but when > I > upgraded to 20130122 I get the following errors. Any have an idea why > sem_post is undefined and how to fix this? > > > > 0(1980) ERROR: <core> [sr_module.c:572]: ERROR: load_module: could not > open module </usr/lib64/kamailio/modules/ims_registrar_pcscf.so>: > /usr/lib64/kamailio/modules/ims_registrar_pcscf.so: undefined symbol: > sem_post 0(1980) ERROR: <core> [sr_module.c:572]: ERROR: load_module: could > not open module </usr/lib64/kamailio/modules/cdp.so>: > /usr/lib64/kamailio/modules/cdp.so: undefined symbol: sem_post > 0(1980) ERROR: <core> [sr_module.c:572]: ERROR: load_module: could not > open module </usr/lib64/kamailio/modules/ims_qos.so>: > /usr/lib64/kamailio/modules/ims_qos.so: undefined symbol: sem_post > > > Thanks > > > -Barry Flanagan >

2 1

Modify IP address in SDP
by Austin Einter 23 Jan '13

23 Jan '13

Dear All When 200 OK to INVITE passes through Kamailio proxy , I want to change the IP address in SDP to a specific IP address. Can this be done by just modifying configuration files or do I need to modify in source code. In case I need to modify source code, which files I need to look at. Kindly provide pointers for said problem. Thanks Austin

3 7

FOSDEM
by Peter Dunkley 22 Jan '13

22 Jan '13

Hello, Are there any plans for a Kamailio meal/get together on the Saturday evening this year? Peter

9 19

Patch to resolve segfault in json module
by Richard Brady 22 Jan '13

22 Jan '13

Hi folks If an empty or invalid JSON string is passed to the json_get_field function from the the json module, it causes a segfault. I have attached a patch. Regards, Richard -- Richard Brady M: +44 (0)7771 623 348 T: +44 (0)20 8144 8160 E: rnbrady(a)gmail.com

2 1

Re: [SR-Users] Kamailio as a front-end for TLS-UDP
by Daniel-Constantin Mierla 22 Jan '13

22 Jan '13

Do no send replies to private address only, keep mailing list otherwise future such emails will be discarded. The contact has to stay the same, because it needs to be the address of the device. Record-Route and Path are designed to take care of routing back to lb, which will use tls to send to device. Cheers, Daniel On 1/22/13 12:26 PM, Anton Kvashenkin wrote: > Thanks for reply. Yes, kamailio acts as a balancer sending REGISTER > and INVITES, etc. The problem is that at L3-L4 it works well (sends > all datagram messages to 10.0.0.1:5060 <http://10.0.0.1:5060> UDP) but > at L7 the Contact header: <>;transport=tls. > > > 2013/1/22 Daniel-Constantin Mierla <miconda(a)gmail.com > <mailto:miconda@gmail.com>> > > Hello, > > > On 1/22/13 10:56 AM, Anton Kvashenkin wrote: >> Guys, I've already asked @ G+ Kamailio coummunity but had no >> answer so I'm asking here: >> >> I'm trying to deploy +Kamailio >> <https://plus.google.com/114984922071338017631> as a front-end >> for TLS-UDP translation, as a back-end would be +FreeSWITCH >> <https://plus.google.com/104108080382960591162> . So, is kamailio >> so smart to fix Contact or Path header for ;transport=tls or it >> translates only on L3-L4 layer and I should fix those headers >> manually? > is it only for calls or you need to forward registrations to > kamailio as well? > > Everything is done automatically, for calls you just need to do > record routing. For registrations you have to use path module and > be sure that freeswitch supports sip path extension. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla -http://www.asipto.com > http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> -http://www.linkedin.com/in/miconda > Kamailio World Conference, April 16-17, 2013, Berlin > -http://conference.kamailio.com - > > -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

Kamailio as a front-end for TLS-UDP
by Anton Kvashenkin 22 Jan '13

22 Jan '13

Guys, I've already asked @ G+ Kamailio coummunity but had no answer so I'm asking here: I'm trying to deploy +Kamailio<https://plus.google.com/114984922071338017631> as a front-end for TLS-UDP translation, as a back-end would be +FreeSWITCH<https://plus.google.com/104108080382960591162> . So, is kamailio so smart to fix Contact or Path header for ;transport=tls or it translates only on L3-L4 layer and I should fix those headers manually?

2 1

VoIP diner at FOSDEM 2013
by Benjamin Henrion 22 Jan '13

22 Jan '13

Hi, I am organizing a diner for all Voice over IP hackers (Asterisk, Freeswitch, Kamailio, OpenSER, Sipvicious :-), etc…) during the Saturday evening of FOSDEM 2013. There is a complete track at the Telephony devroom as well: https://fosdem.org/2013/schedule/track/telephony/ More infos are registration: http://www.zoobab.com/voip-diner-fosdem Best, -- Benjamin Henrion <bhenrion at ffii.org> FFII Brussels - +32-484-566109 - +32-2-3500762 "In July 2005, after several failed attempts to legalise software patents in Europe, the patent establishment changed its strategy. Instead of explicitly seeking to sanction the patentability of software, they are now seeking to create a central European patent court, which would establish and enforce patentability rules in their favor, without any possibility of correction by competing courts or democratically elected legislators."

1 0

Wiki page for what is new in v4.0.0
by Daniel-Constantin Mierla 22 Jan '13

22 Jan '13

Hello, there is a skeleton content for what is new in upcoming v4.0.0, but there are many new features missing in the lists. Please try to add short notes about the features you know they were added for 4.0. The wiki page is available at: * http://www.kamailio.org/wiki/features/new-in-devel Anyone that makes an account can edit: * https://www.kamailio.org/wiki/start?do=register Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

PATH advertised
by Olle E. Johansson 22 Jan '13

22 Jan '13

Hi! If I'm running Kamailio behind NAT and need to add a Path header with the outside IP address and port, that is configured as the "advertise" address - how do I do that? Can I add two Path headers at the same time? I can't find a way to add it like I can with Record-Route headers. /O

2 2

Re: [SR-Users] sr-users Digest, Vol 92, Issue 43
by madhumanjusha＠integramicro.com 21 Jan '13

21 Jan '13

Thanks a lot for quick response. On Sat, January 19, 2013 3:59 am, sr-users-request(a)lists.sip-router.org wrote: > 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. Mid call announcement(kamailio server) > (madhumanjusha(a)integramicro.com) > 2. Re: Mid call announcement(kamailio server) > (rabs(a)dimension-virtual.com) > 3. CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus > dlg - Kamailio 3.3.x (Rinor Hoxha) 4. Huge undertaking and in need of > advices from the community (Rumen Mihailov) > 5. Re: Huge undertaking and in need of advices from the > community (Stoyan Mihaylov) 6. Re: Invite - negotiate to asterisk as peer > not kamailio (Scott, Matt) > > > > ---------------------------------------------------------------------- > > > Message: 1 > Date: Fri, 18 Jan 2013 06:52:15 -0500 (EST) > From: madhumanjusha(a)integramicro.com > Subject: [SR-Users] Mid call announcement(kamailio server) > To: sr-users(a)lists.sip-router.org > Message-ID: > <39960.61.8.152.138.1358509935.squirrel(a)mail.integramicro.com> > Content-Type: text/plain;charset=iso-8859-1 > > > Hello, > As of now am working on Kamailio server. > My task is to announce in between call,my sip clients are EKIGA. > I should able to make my kamailio server act as controller and media > server,for that purpose my first step is : invite hold to called party. > second step is:: invite(sdp for .wav file) to caller and then am sending > rtp streaming.In wireshark am able to see rtp packets,but caller is not > playing the .wav file which am streaming through ortp library from > kamailio. > > Can anyone help me out with this problem?On top of it,is this possible in > reality? > > > Thanks & Regards, > Manjusha A. > Integra Micro Software Services (P) Ltd. > > > > > > > > ------------------------------ > > > Message: 2 > Date: Fri, 18 Jan 2013 15:14:49 +0100 (CET) > From: rabs(a)dimension-virtual.com > Subject: Re: [SR-Users] Mid call announcement(kamailio server) > To: "SIP Router - Kamailio $OpenSER$ and SIP Express Router $SER$ > - Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: > <38001.212.40.242.42.1358518489.squirrel(a)www.rodriguezfeo.es> > Content-Type: text/plain;charset=iso-8859-1 > > >> Hello, >> As of now am working on Kamailio server. >> My task is to announce in between call,my sip clients are EKIGA. >> I should able to make my kamailio server act as controller and media >> server,for that purpose my first step is : invite hold to called party. >> second step is:: invite(sdp for .wav file) to caller and then am >> sending rtp streaming.In wireshark am able to see rtp packets,but caller >> is not playing the .wav file which am streaming through ortp library >> from kamailio. >> >> Can anyone help me out with this problem?On top of it,is this possible >> in reality? > > No, it's not possible using only kamailio, and thats because kamailio > it's a SIP proxy, it have nothing to do with media, for what you are > triying to get you need to router your calls throught a B2BUA, like sems > or asterisk. > > Best regards > > > > > > ------------------------------ > > > Message: 3 > Date: Fri, 18 Jan 2013 14:11:05 +0100 > From: Rinor Hoxha <rinorhoxha(a)gmail.com> > Subject: [SR-Users] CRITICAL: dialog [dlg_timer.c:205]: Trying to > update a bogus dlg - Kamailio 3.3.x To: sr-users(a)lists.sip-router.org > Message-ID: > <CAFvba4ipKqicMJ0qn9UNLXmhjGhf_83fzG=L5w2qVBpc7JC1GQ(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > > Hi list, > > > I'm having some issues with dialog module. From time to time I get the > following error: > > ERROR: dialog [dlg_req_within.c:231]: failed to update dialog lifetime > CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg > tl=0x2aeeea1c6840 tl->next=(nil) tl->prev=(nil) > > [root@proxy ~]# cat /var/log/kamailio.log | grep "Jan 17" | grep "tl=" | > sort | uniq -c > > 1 Jan 17 13:27:00 proxy /usr/local/kamailio/sbin/kamailio[4100]: > CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg > tl=0x2aeeea09bf38 tl->next=(nil) tl->prev=(nil) 1 Jan 17 14:39:30 proxy > /usr/local/kamailio/sbin/kamailio[4100]: > CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg > tl=0x2aeeea07c518 tl->next=(nil) tl->prev=(nil) 1 Jan 17 15:06:21 proxy > /usr/local/kamailio/sbin/kamailio[4100]: > CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg > tl=0x2aeeea2da900 tl->next=(nil) tl->prev=(nil) 1 Jan 17 19:31:02 proxy > /usr/local/kamailio/sbin/kamailio[4100]: > CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg > tl=0x2aeeea278cd0 tl->next=(nil) tl->prev=(nil) 1 Jan 17 19:31:42 proxy > /usr/local/kamailio/sbin/kamailio[4100]: > CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg > tl=0x2aeee9f527a8 tl->next=(nil) tl->prev=(nil) 1 Jan 17 20:05:32 proxy > /usr/local/kamailio/sbin/kamailio[4100]: > CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg > tl=0x2aeeea21fa30 tl->next=(nil) tl->prev=(nil) > > This one was flooding my log file: tl=0x2aeeea1c6840 > > > 24322 Jan 17 21:53:02 proxy /usr/local/kamailio/sbin/kamailio[4100]: > CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg > tl=0x2aeeea1c6840 tl->next=(nil) tl->prev=(nil) 5709 Jan 17 21:53:23 proxy > /usr/local/kamailio/sbin/kamailio[4100]: > CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg > tl=0x2aeeea1c6840 tl->next=(nil) tl->prev=(nil) > > 61314 Jan 17 21:53:23 proxy /usr/local/kamailio/sbin/kamailio[4102]: > CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg > tl=0x2aeeea1c6840 tl->next=(nil) tl->prev=(nil) 22812 Jan 17 21:54:01 proxy > /usr/local/kamailio/sbin/kamailio[4102]: > CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg > tl=0x2aeeea1c6840 tl->next=(nil) tl->prev=(nil) 21137 Jan 17 21:54:02 proxy > /usr/local/kamailio/sbin/kamailio[4102]: > CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg > tl=0x2aeeea1c6840 tl->next=(nil) tl->prev=(nil) > > where pid 4100 is slow timer and pid 4102 is timer. Any idea why is this > happening and how to fix it. (I suspect that may be some issues with our > server memory...checking now...and will inform). However wanted to raise > this issue in case anyone else is having the same. > > Thanks, Rinor >

1 0

Kamailio Development Workshop, Feb 14-15, 2013, Alicante, Spain
by Daniel-Constantin Mierla 21 Jan '13

21 Jan '13

Hello, I want to announce the next Kamailio Development Workshop, in Alicante, Spain, during Feb 14-15, 2013, organized by Asipto and University of Alicante, Department of Master Degrees in Telecommunication Engineering. You can find more details at: - http://www.kamailio.org/w/2013/01/kamailio-development-workshop-feb-14-15-2… Like the previous one of its kind (done in 2011 in Barcelona), this type of workshop targets to explain the source code structure and internal APIs of Kamailio, to speed up the learning of how to write own Kamailio extensions in C (e.g., write a module, which will be part also of the content). Its low registration fee is set to cover organizational logistics, the number of seats is limited, participants are accepted in the order of registration. Besides the workshop during the two days, on the evening of Feb 14 is planned a VoIP dinner. You can participate to it even if not coming to workshop. It will be nice to meet face to face and talk about latest trends in RTC/UC. If you need further details, contact me via email. Regards, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

all modules in one folder
by Daniel-Constantin Mierla 20 Jan '13

20 Jan '13

Hello, please note that all modules are now back in one folder, respectively 'modules/'. There were few modules I couldn't compile due to missing exotic libraries, but based on grep, they should not depend on older modules paths. Anyhow, testing that the set of modules you are using compile and load file would be recommended. The folders modules_k and modules_s are now empty, with a placeholder .gitkeep, to be removed later if there is no need to revert any module relocation. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

updates to perl modules
by Daniel-Constantin Mierla 20 Jan '13

20 Jan '13

Hello, there were several changes pushed to master branch related to perl modules: - perl was renamed to app_perl - to follow the naming pattern for embedded interpreters - embedded perl library is named now Kamailio (it was OpenSER so far, thanks to patches from Stoyan Mihaylov) - perlvdb was renamed to db_perlvd because it is a db driver module If you are using these modules, you will have to update your configuration file and perl scripts. Testing and feedback is appreciated. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

renaming to kamailio in perl and snmpstats modules
by Daniel-Constantin Mierla 20 Jan '13

20 Jan '13

Hello, I'm writing to see if anyone has some spare time to take care of renaming from openser to kamailio in perl and snmpstats modules. I assume is more like search and replace, but has to be carefully done to be coherent and not break stuff there, so it might be a bit more than a sed command. The IANA enterprise number in MIBs is already the one for kamailio.org Thanks, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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Testdrive the coming release 4.0 now!
by Olle E. Johansson 19 Jan '13

19 Jan '13

http://www.kamailio.org/w/2013/01/test-4-0-now/ Friends, We do need your help testing 4.0 now. - Compile on your systems, from small embedded systems to large servers, from Linux to *BSD and Solaris - Use with your configuration script - Test the new modules and functions - Read the updated documentation files On the link above, you have instructions on how to build from GIT sources and install a test system. We do look forward to your feedback! Let's all work to make this a very, very good and succesful release! Regards, /O ---- Kamailio on Twitter: @kamailioproject (also on Facebook and Google plus)

1 0

Huge undertaking and in need of advices from the community
by Rumen Mihailov 18 Jan '13

18 Jan '13

Hi all, Thank you for all the help till now. Until now I was trying to setup completely redundant setup with Kamailio + asterisk boxes and for the moment i have the following configuration working. 2x Kamailios (Active/Passive) 2x Asterisks Kamailio is load balancing I can add more boxes if needed MySQL Cluster for all database needs Now the time has come and I will need to made this setup available for around 10K users. I will need to become an ITSP for an Internet Provider that is missing the telephony part from his package, so I have a couple of questions. 1. Is this setup OK to cover the needs of the customer base ? hardware will be decent 2. The end users will most probably need ATA adapters. What is the best cost effective solution I can go for ? 3. Do I need any other piece of software apart from the SIP Proxy and asterisk as I need the cost of implementation to be as low as possible. 4. Pricing...I can see that there are a lot of plans of the competitors that include 1000 minutes for free...Do you have any statistics what is the avarage % usage of prepaid minutes ? 5. Are there any other "stones" that I might hit and should be aware of ? Thank you for the information Best regards, Rumen

2 1

Merry Christmas!
by Daniel-Constantin Mierla 18 Jan '13

18 Jan '13

Another year getting to its end! Looking back, looks like one with the greatest achievements in the development of the project so far. Year is not done, so I save that summary for one week later, there is still stuff on its way to our source code repository. Now I want to thank to everyone promoting and contributing to the project, from developers to community members, and wish Merry Christmas and great winter holidays to all supporters of our project! Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

8 8

Mid call announcement(kamailio server)
by madhumanjusha＠integramicro.com 18 Jan '13

18 Jan '13

Hello, As of now am working on Kamailio server. My task is to announce in between call,my sip clients are EKIGA. I should able to make my kamailio server act as controller and media server,for that purpose my first step is : invite hold to called party. second step is:: invite(sdp for .wav file) to caller and then am sending rtp streaming.In wireshark am able to see rtp packets,but caller is not playing the .wav file which am streaming through ortp library from kamailio. Can anyone help me out with this problem?On top of it,is this possible in reality? Thanks & Regards, Manjusha A. Integra Micro Software Services (P) Ltd.

2 1

HEAD: Problems with $$ in xlog
by Klaus Darilion 17 Jan '13

17 Jan '13

Hi! In xlog I used to use $$ to escape the dollar sign to not be interpreted as PV, eg: xlog("L_INFO"," $$ru\n"); Current HEAD gives an error: ERROR: <core> [pvapi.c:757]: error searching pvar "" ERROR: <core> [pvapi.c:961]: wrong char [$/36] in [$] at [0 (0)] ERROR: xlog [xlog.c:363]: wrong format[ $$ru#012] ERROR: <core> [route.c:1214]: fixing failed (code=-1) at cfg:/etc/kamailio/kamailio.cfg:627 Where there some changes? Thanks Klaus

2 2

Simple command to insert attributes to domain_attrs
by Philippe Sultan 17 Jan '13

17 Jan '13

Hi, I'm migrating from sip-router/Kamailio 3.1, SER flavour to the latest dev version, with MySQL support. I need to add attributes to domains, and therefore use the domain_attrs table. A handy command I used for that was ser_attr : ser_attr add domain=anydomain.voip attr=value Is there something similar I can use under the Kamailio flavour? kamctl does not seem to help here, unless I missed an option. Regards, Philippe

3 5

Users can't authenticate via external IP
by Rumen Mihailov 17 Jan '13

17 Jan '13

Hello Kamailio guys, I have the following setup Internet -----> REAL STATIC IP -----> Kamailio Virtual IP(10.2.45.101) I have setup everything and when I try to connect to kamailio from the inside I can successfully register. When I try to register to the REAL STATIC IP which is forwarded to 10.2.45.101 I get unauthorized. I have narrowed down the problem to this: auth_check("$fd", "sip", "1") as I said I can authenticate from the internal network but not from the Internet. What do I need to alter ? Side problem... why when using xlog("L_WARN","$Cxx TEST") kamailio fails to start ? seems that my version does not support the colored output... Thank you for your help guys...in advance. Best regards, Rumen http://itradebg.com

2 1

Server sip.iptel.org did not respond to register
by Faisal Farooq 17 Jan '13

17 Jan '13

Dear Support, We are using IPTEL SIP services. Now we are getting Error message on our VOIP software is " Server sip.iptel.org did not respond to register". Moreover, we can successfully ping and complete trace route the sip.iptel.org from our server but failed to register through sip.iptel.org server. Kindly suggest what is the cause and look into this matter. I am sending you our IPTEL SIP ID/Username and please find below. (1) 196211 (2) 195404 Regards, Faisal

2 1

i cannot log on to iptel.org
by kelly randolph 17 Jan '13

17 Jan '13

i cannot log on to iptel.org is everything okay???? thanx, kellyr

2 1

cant login
by Pasindu Hettiachchi 17 Jan '13

17 Jan '13

hi , my linksys router failed to register with iptel.org this happened from last night, any think that I have to change ? Pasindu

2 1

Error while loading modules, if given a custom library path
by Ajay Sabat 16 Jan '13

16 Jan '13

Hi I have trouble loading modules from a custom lib directory. It works fine if the libraries are located under PREFIX/lib/Kamailio/ (PREFIX is the path that I gave while building Kamailio). But on the target I will have to keep the libraries in a custom directory. I tried by changing the mpath in kamailio.cfg. Also I tried by giving the path in -L option while invoking kamailio. It is not working. It fails with the following errors, even if the libraries are present at the location where it compains. I searched but couldn't find any answer why it is failing, though it is supposed to work. Please help me. $ ./sbin/kamailio -E -d -f /home/ajay/projects/video_conf/etc/kamailio/kamailio.cfg -c loading modules under /home/ajay/projects/video_conf/lib/kamailio/:/home/ajay/projects/video_conf/lib/kamailio/modules/:/home/ajay/projects/video_conf/lib/kamailio/modules_k/ 0(27896) ERROR: <core> [sr_module.c:572]: ERROR: load_module: could not open module </home/ajay/projects/video_conf/lib/kamailio/modules_k/mi_fifo.so>: libkmi.so.1: cannot open shared object file: No such file or directory 0(27896) : <core> [cfg.y:3591]: parse error in config file /home/ajay/projects/video_conf/etc/kamailio/kamailio.cfg, line 222, column 12-23: failed to load module 0(27896) ERROR: <core> [sr_module.c:572]: ERROR: load_module: could not open module </home/ajay/projects/video_conf/lib/kamailio/modules_k/kex.so>: libkmi.so.1: cannot open shared object file: No such file or directory 0(27896) : <core> [cfg.y:3591]: parse error in config file /home/ajay/projects/video_conf/etc/kamailio/kamailio.cfg, line 223, column 12-19: failed to load module 0(27896) ERROR: <core> [sr_module.c:572]: ERROR: load_module: could not open module </home/ajay/projects/video_conf/lib/kamailio/modules_k/tmx.so>: libkmi.so.1: cannot open shared object file: No such file or directory 0(27896) : <core> [cfg.y:3591]: parse error in config file /home/ajay/projects/video_conf/etc/kamailio/kamailio.cfg, line 225, column 12-19: failed to load module 0(27896) ERROR: <core> [sr_module.c:572]: ERROR: load_module: could not open module </home/ajay/projects/video_conf/lib/kamailio/modules/sl.so>: libkcore.so.1: cannot open shared object file: No such file or directory $ ls -l /home/ajay/projects/video_conf/lib/kamailio/modules_k/mi_fifo.so -rwxr-xr-x 1 ajay ajay 155651 Jan 16 09:58 /home/ajay/projects/video_conf/lib/kamailio/modules_k/mi_fifo.so $ ls -l /home/ajay/projects/video_conf/lib/kamailio/libkmi.so* lrwxrwxrwx 1 ajay ajay 13 Jan 16 09:56 /home/ajay/projects/video_conf/lib/kamailio/libkmi.so -> libkmi.so.1.0 lrwxrwxrwx 1 ajay ajay 13 Jan 16 09:56 /home/ajay/projects/video_conf/lib/kamailio/libkmi.so.1 -> libkmi.so.1.0 -rwxr-xr-x 1 ajay ajay 78111 Jan 16 09:56 /home/ajay/projects/video_conf/lib/kamailio/libkmi.so.1.0 Regards, Ajay

1 1

make option to build a lean kamailio image
by Ajay Sabat 16 Jan '13

16 Jan '13

Hi, I need to build a simple Kamailio server that could do just registration and basic call routing. I don't need database support, it is fine if it loses data during reboot. The only other option I may need is NAT support. So far I was building the server using make all as given below: $ make FLAVOUR=kamailio PREFIX="$HOME/projects/video_conf/sipserver" cfg $make all $make install I noticed that the total size of library and binary files is huge, little less than 30MB. I have to port and test the server on a target where I don't have much space. Can somebody please suggest me what make option would be appropriate for me? Is it fine to not build the ser modules under modules_s/ and what is make option to do so? I tried "make" instead of "make all", but it somehow builds everything during the step "make install" and the final result is same. Thanks, Ajay

2 2

TLS questions
by Fabian Borot 15 Jan '13

15 Jan '13

Hi, we need to implement TLS in our current setup and have some questions about it: 1- We have a farm of Asterisk servers behind our kamailio proxy. The call flow is Customers/Providers <--> Proxy <--> Asterisks. Is it possible to have TLS only between the Customers/Providers and the Proxy and then just regular SIP over UDP between the Proxy and the Asterisks? 2- Can we have TLS over both TCP and UDP at the same time? Thank you a lot fborot

3 3

Dokuwiki upgraded
by Daniel-Constantin Mierla 15 Jan '13

15 Jan '13

Hello, fyi, I upgraded dokuwiki to latest version, switching to the new default template (which is not bad at all, btw) as the old one is not yet fully compatible. If you find any issue, report to devel mailing list. It is about the wiki for v3.2+: - http://www.kamailio.org/wiki/ Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

2 2

NAPTR for TLS only
by Daniel Pocock 14 Jan '13

14 Jan '13

I'm just wondering if anyone can comment on expected and actual behavior if there is only a NAPTR record for TLS, e.g. I have: sip5060.net. IN NAPTR 10 0 "s" "SIPS+D2T" "" _sips._tcp.sip5060.net. and I don't have any entry for "SIP+D2U" or "SIP+D2T" If some third party Kamailio instance (e.g. sip-server.example.org) receives a request from a user trying to call sip:user@sip5060.net, with a sip: rather than sips: URI, should it (and will it) use the "SIPS+D2T" result, if no other result is available? Or would it ignore the NAPTR record and try to find the default SRV record such as _sip._udp.sip5060.net ? Should there be another NAPTR record to translate sip: to sips: using a regex perhaps, or would such a NAPTR be a bad thing?

3 4

Kamailio, Asterisk with TLS+SRTP
by Roberto Fichera 14 Jan '13

14 Jan '13

Hi All, I would setup a configuration where Kamailio authenticate asterisk SIP trunk using TLS and SRTP. At moment I was able to configure everything, including RTTProxy since most of the asterisks v1.8.19.1 are behind NAT. So far so good it works pretty good using standard authentication and the call goes straight between asterisks. But as soon as I move my configuration for both kamailio & asterisk to TLS+SRTP I'm not able to authenticate asterisk SIP trunks. Especially asterisk seems insisting to use the port 5060 even if I requested the TLS on 5061. kamailio v3.3.3 tls.cfg is configured as: [server:default] method = TLSv1 verify_certificate = no require_certificate = no private_key = /etc/pki/tls/private/server.key certificate = /etc/pki/tls/certs/server.pem ca_list = /etc/pki/tls/certs/ca-bundle.crt #crl = //etc/kamailio/crl.pem # This is the default client domain, settings # in this domain will be used for all outgoing # TLS connections that do not match any other # client domain in this configuration file. # We require that servers present valid certificate. # [client:default] verify_certificate = no require_certificate = no So my asterisk conf is the following: [general] tlsenable=yes tlsbindaddr=0.0.0.0 tlscertfile=/etc/asterisk/5002.pem tlscafile=/etc/asterisk/ca-bundle.crt tlscipher=ALL tlsclientmethod=tlsv1 tlsdontverifyserver=yes transport=tls,udp .... ..... and the SIP trunk is configured as [kamailio] type=peer insecure=invite,port nat=yes disallow=all allow=ulaw host=kamailio_ip outboundproxy=tls://kamailio_ip port=5061 defaultuser=5002 fromuser = 5002 fromdomain =mydomain secret=5002 qualify=yes dtmfmode=rfc2833 context=default callbackextension=5002 directmedia=nonat sendrpid=yes transport=tls encryption=yes register => tls://5002:5002@kamailio_ip:5061/5002 I still get error like: Jan 14 10:45:12] ERROR[5244]: chan_sip.c:5600 create_addr_from_peer: 'UDP' is not a valid transport for 'dicenet'. we only use 'TLS'! ending call. [Jan 14 10:45:12] WARNING[5244]: chan_sip.c:13722 transmit_register: Probably a DNS error for registration to 5002@kamailio_ip, trying REGISTER again (after 20 seconds) [Jan 14 10:45:32] ERROR[5244]: chan_sip.c:5600 create_addr_from_peer: 'UDP' is not a valid transport for 'dicenet'. we only use 'TLS'! ending call. [Jan 14 10:45:32] WARNING[5244]: chan_sip.c:13722 transmit_register: Probably a DNS error for registration to 5002@kamailio_ip, trying REGISTER again (after 20 seconds) [Jan 14 10:45:52] ERROR[5244]: chan_sip.c:5600 create_addr_from_peer: 'UDP' is not a valid transport for 'dicenet'. we only use 'TLS'! ending call. [Jan 14 10:45:52] WARNING[5244]: chan_sip.c:13722 transmit_register: Probably a DNS error for registration to 5002@kamailio_ip, trying REGISTER again (after 20 seconds) [Jan 14 10:46:07] ERROR[7041]: tcptls.c:444 ast_tcptls_client_start: Unable to connect SIP socket to kamailio_ip:5060: Connection timed out Does anyone can suggest me something to read, try, check? Best regards. Roberto Fichera.

2 2

How to correctly check if the variable is set?
by Mino Haluz 14 Jan '13

14 Jan '13

Hi, how should I check if the value is set? if ($avp(s:test) == "") { or is there any null keyword ? If so, does it work for $avp, $sht, $var and $shv ? Thanks, Mino

3 4

Development is frozen for v4.0.0
by Daniel-Constantin Mierla 11 Jan '13

11 Jan '13

Hello, last midnight was the term to freeze the development for next major release v4.0.0. For a while, the GIT must branch should get only commits related to bug fixes, documentation improvements and updates of the additional tools. If any developer needs to store code of new features in the repository, use personal branches (or temporary branches in tmp/ when many developers work on the same code). After unfreeze, these branches can be merged in the master branch. For 1-2 months from now on, the main focus with the master branch is testing. The v4.0.0 will be out after mid of February, exact date to be set later based on the results during testing phase. There were many additions, so any kind of help testing is very appreciated. Fortunately the core and most of the very used classic modules were not changed much, ensuring good stability. But there are old very used pieces such as usrloc/registrar that got touched as well, so try to test as much as possible the components you are using in your configs. Provide any kind of feedback on sr-dev mailing list or the bug tracker. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, April 16-17, 2013, Berlin - http://conference.kamailio.com -

1 0

PV: PAI vs. PPI
by Klaus Darilion 11 Jan '13

11 Jan '13

Hi! I just found out that there is only $ai for the whole P-Asserted-Identity URI, but lots of PVs for P-Preferred-URI: uri, domainpart, userpart ... Is there a reason for that, or is it just that nobody implemented it for PAI? I think it should be easy to extend it for PAI too. regards Klaus

2 1

Kamailio World Conference - Registration is open
by Daniel-Constantin Mierla 11 Jan '13

11 Jan '13

Hello, the registration for Kamailio World Conference is now open! You can see more details and register at: - http://conference.kamailio.com/k01/registration/ There is already a great group of speakers and interesting proposed talks. More regarding the content will be published in the near future, keep an eye on event's web site: - http://conference.kamailio.com If you are considering to speak at the conference, submit your proposal as soon as possible, the slots are filling up quickly. Looking forward to meeting many of you at the conference! Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

3 3

Kamailio on Twitter
by Olle E. Johansson 11 Jan '13

11 Jan '13

Hello! For those of you that have missed it: You can now follow the project on Twitter: @kamailioproject https://twitter.com/kamailioproject Amongst the latest news are Kamailio at Fosdem 2013, Code freeze for kamailio 4.0 and the Kamailio World conference now open for registrations. Tonight we went into code freeze for the coming release of Kamailio 4.0. This means that developers can not add any more features to this release, only fix bugs. In order to fix bugs we will need your help testing. After the weekend we'll come back with instructions on how to test and report bugs. The information is already available on the web site. Have a great weekend! /O

1 0

LCR
by Douglas Ugalde 11 Jan '13

11 Jan '13

Hi, Im trying to configure LCR in Kamailio 3.3.3 but I dont Know how can I do to fix this error: ERROR: lcr [lcr_mod.c:1840]: error while parsing R-URI This is my LCR configuration block: route[LCR] { if(!load_gws(1)){ sl_send_reply("500", "Internal server error, unable to load gateways"); xlog("L_NOTICE","Internal server error, unable to load gateways"); break; } if(!next_gw()){ sl_send_reply("503", "Service not available, no gateways found"); break; } } I not sure if this configuration thats ok, please somebody help me. Note: Params and modules are already loaded, sorry for my english. Best regards.

2 1

Problem with MSILO module forwarding messages
by Owen Lynch 10 Jan '13

10 Jan '13

Hi, I have the kamailio 3.3.2 msilo module basically working but I cannot configure it to forward stored messages to the registered UA. Instead it forwards to itself (via DNS lookup on the domain name). This is causing me grief because I am getting recursive message handling. The module overview describes this behaviour: "Every time when a user registers with Kamailio, the module is looking in database for offline messages intended for that user. All of them will be sent to contact address provided in REGISTER request." The REGISTER request contact header looks OK: Contact: <sip:hxqecvtn@192.168.2.236:57532 ;transport=tls>;expires=3600;received="sip:192.168.2.236:53628 ;transport=TLS";+sip.instance="<urn:uuid:1ee63413-3569-46dc-a9ee-34c6e4e1403c>" However the 'outbound_proxy' parameter seems to contradict this: "The SIP address used as next hop when sending the message. Very useful when using Kamailio with a domain name not in DNS, or when using a separate Kamailio instance for msilo processing. If not set, the message will be sent to the address in destination URI." So, when 'outbound_proxy' is not set, the message is sent to the address in the destination URI, rather than the UA contact address? Can the module be configured to forward stored messages directly to the UA contact address? Thanks, Owen Lynch

2 1

Socket error
by Douglas Ugalde 10 Jan '13

10 Jan '13

Hi, Please I need help with this error: ERROR: ctl [ctl.c:379]: ERROR: ctl: could not delete unix socket /tmp/kamailio_ctl: Operation not permitted (1) I'm using Kamailio 3.3.3 I call 2 times before got this error, how can I do to resolved it, I really really appreciated any help. I attached my Kamailio.cfg. Really sorry for my English. Best Regards.

2 1

NAT Problem! Not relay ACK,CANCEL and BYE to other UA
by Nguyen Anh Tuan 10 Jan '13

10 Jan '13

Hi All, I have 2 interfaces - External A (behind NAT) and Internal B (LAN) in Kamailio Load Balance, and enable Multi-Homing function "mhomed=1". Load Balance dispatches to Kamailio SIP Servers via the Internal B (LAN) (each Kamailio SIP server has both proxy and registrar in one box). For RTP relay, I use RTPproxy in other server, has 2 interfaces A&B like Load Balance. If using 2 interfaces in Load Balance, Registration and calling NORMAL with 2 users behind NAT, but the call will be interrupted after 30 seconds due to the ACK message (after Ringing, 200 OK) STOP in Load Balance in return and not relayed to other UA . CANCEL and BYE have same error like ACK. I tried using only one interface for load balancing (External = Internal IP), and SIP Server binds to Load Balance in the same network, EVERYTHING IS OK. ACK, CANCEL and BYE are not missing behind NAT. For NAT Traversal I use PATH Module in Load Balance, and add "Received" like this: ... modparam("path", "use_received", 1) .. .. record_route(); add_path_received(); force_rport(); t_relay; .. .. And in SIP Server enable Path Parameter for Registrar: modparam("registrar", "use_path", 1) modparam("registrar", "path_mode", 1) modparam("registrar", "path_use_received", 1) Enable "#!define WITH_NAT" in default kamailio.cfg for RTP Controlling, but I removed all fix_nated_register and fix_nated_contact in SIP Server, because using PATH module.I try enable both fix_nat_register and fix_nated_contact, or only fix_nat_register but NOT SOLVE Problem "missing ACK,CANCEL, and BYE in Load Balance" behind NAT. Best Regard, Tuan.

2 1

Testing SIP Outbound / RFC5626 (Was: Freezing for next major release)
by Richard Brady 10 Jan '13

10 Jan '13

Hi Peter Great work on this! We'd like to help you test. The test I have in mind, which we could create using SIPp, would be to register multiple contacts with the same instance-id (i.e. "sip.instance" param) but different reg-id params. Then send an INVITE to that AoR and make sure the forking is only per instance-id and not per reg-id. This could be repeated in multiple permutations of instance-ids and reg-ids. This would be a test of save() and lookup() more than anything else. Is that what you had in mind? Richard On 3 January 2013 14:13, Peter Dunkley <peter.dunkley(a)crocodile-rcs.com>wrote: > I hope to get the outbound edge proxy and flow timer stuff into master by > Monday, but it could really do with some additional testing especially in > conjunction with the reg-id stuff in registrar/usrloc (which I have no idea > how to use). > > Peter > >

5 14

LCR with Siremis
by Douglas Ugalde 09 Jan '13

09 Jan '13

Hi, I was configuring the modparams for LCR and Dialplan, I taken from kamailio web site, but I think the params are outdated, because we need to erase some lines to match with the tables in the DB, Im using Kamailio 3.3.3 and Siremis 3.3 Please can you give 1 example about how to make outbounds calls with siremis 3.3 (send calls to gateway)? I already have the lcr and dialplan module loaded with params. Where can I get update documentation about Siremis 3.3 and Kamailio 3.3.3. Thanks for your help.

1 0

Freezing development for v4.0.0
by Daniel-Constantin Mierla 09 Jan '13

09 Jan '13

Hello, a (26-hours before) reminder that development for next major release v4.0.0 will be frozen tomorrow, Thursday, January 10, 2013, at 23.59 GMT. If any developer has new features to push in the git master branch, please do it before. After that, for a while, the master should get only bug fixes and improvements to documentation and addition tools (e.g., bringing kamctl up to date). New features have to be kept in personal branches until the master branch is open again. The focus should be on testing from the day after tomorrow on, I hope the users community will help us here as well. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Re: [SR-Users] Having "modparam" options in the database
by Glenn O. Larsen 09 Jan '13

09 Jan '13

On Tue, Jan 8, 2013 at 9:21 AM, Olle E. Johansson <oej at edvina.net> wrote: >>>> Is there a way to read modparam config from a database? > In most cases, there should be no need to change modparams at > runtime. > > Can you tell us more about which parameter you want to change during > runtime that doesn't have a mI/rpc reload command? Hi Olle! The module we have been using, and wanted database configuration for was the pike module. But and after spending some more time looking at alternative modules, I see that we can use other modules to help us with traffic control/blocking, like the pipelimit and the ratelimit (both these modules have some sort of database integration, need to look more into this). I was so focused on the pike module, that I didn't see these other modules.. :) -- mvh Glenn O Larsen

3 3

Core fm_realloc function
by Bruno Bresciani 08 Jan '13

08 Jan '13

Hi all, I observed two at 0x08173bf0 in fm_realloc (qm=0xb5f7d000, p=0xb61f0bc0, size=0) at mem/f_malloc.c:536. I don't know exactly what may be caused this core because the debug of kamailio is disabled, but looking at backtrace (gdb) and source code I could notice a relationship with TLS connection establishment (handshake). Other important fact is that core was generated by the kamalio itself with signal 6 (abort) because it can't find the pointer to memory realloc. I'd like to know if someone already observed this issue and if my analyse above is correct? Bellow is backtrace full with gdb: Core was generated by `/home2/local/kamailio/sbin/kamailio -P /var/run/kamailio.pid'. Program terminated with signal 6, Aborted. #0 0x00f30402 in __kernel_vsyscall () (gdb) (gdb) (gdb) (gdb) (gdb) bt full #0 0x00f30402 in __kernel_vsyscall () No symbol table info available. #1 0x009fec10 in raise () from /lib/libc.so.6 No symbol table info available. #2 0x00a00521 in abort () from /lib/libc.so.6 No symbol table info available. #3 0x08173bf0 in fm_realloc (qm=0xb5f7d000, p=0xb61f0bc0, size=0) at mem/f_malloc.c:536 f = (struct fm_frag *) 0xb61f0bb8 pf = <value optimized out> orig_size = 18 L=ÃbÂ¬Ã2n = (struct fm_frag *) 0xb61f0bd2 ptr = <value optimized out> hash = 23548 #4 0x0051a87f in ser_realloc (ptr=0xb61f0bc0, size=32) at ../../mem/shm_mem.h:266 No locals. #5 0x00460295 in OPENSSL_cleanse () from /lib/libcrypto.so.6 No symbol table info available. #6 0x004609f7 in CRYPTO_realloc () from /lib/libcrypto.so.6 No symbol table info available. #7 0x003efb22 in sk_insert () from /lib/libcrypto.so.6 No symbol table info available. #8 0x003efbaa in sk_push () from /lib/libcrypto.so.6 No symbol table info available. #9 0x007c2ac3 in ssl_bytes_to_cipher_list () from /lib/libssl.so.6 No symbol table info available. #10 0x007aa9f3 in ssl3_get_client_hello () from /lib/libssl.so.6 No symbol table info available. #11 0x007ab815 in ssl3_accept () from /lib/libssl.so.6 No symbol table info available. #12 0x007c1c3a in SSL_accept () from /lib/libssl.so.6 No symbol table info available. #13 0x005278e3 in tls_accept (c=0xb621af88, error=0xbf91dfc8) at tls_server.c:346 ret = <value optimized out> ssl = (SSL *) 0xb61c1878 cert = <value optimized out> tls_c = (struct tls_extra_data *) 0xb6208e58 tls_log = <value optimized out> #14 0x0052da8e in tls_read_f (c=0xb621af88, flags=0xbf91e200) at tls_server.c:1028 r = (struct tcp_req *) 0xb621affc bytes_free = 4095 bytes_read = 70 read_size = 4095 ssl_error = 0 ssl_read = 0 ssl = (SSL *) 0xb61c1878 rd_buf = "\026\003\001\000A\001\000\000=\003\001OÃ¿\vÂ¡ywP\fkÃDÃ 0/\000\a\000\005\001\000ÃÃÃÃ¯000\000\026\0009\0008\0005\000\026\000\023\000\n\0003\0002\00 202i<Ã¤<Ã¥Ã¾\031Ã¬234Â \237Ãª030Y){lÑ¶"...06Ã'Ãr9Ã¬Ã«Â¸{[Â©\t\205\037\036Ã¿HPH5Âµ\t(Ã©32Ã027Ã¦Ã²036Â¾\"\rÔ¤YÃ215uÂº\027)Ã¶000Ã±215\vÃÃ Ã©Ea\023xÃ©J\206T\204CÃ wr_buf = "\027\003\001\003Â wÂ²'\016Ñ¼Ã033ÃLÃ021Ã)Ã\220hnLÂ¦\020XÃª\236\200\tIhÃÂ¸Sf\234Ã\rÃ ?Â¯Â¼\024FÃ\200Âª/\2264RÃaÃÂ¤IpXÂ»Â¨Ã035Ã¤Qh\211Ã026Ã½4ÃU+\asrÂ£/`Â¤OÃ»1AÊ¹qUÜ¿Â·vÂ±Ã¼\205"... rd = {buf = 0xbf90dfa4 "\026\003\001", pos = 70, used = 70, size = 65536} wr = { buf = 0xbf8fdfa4 "\027\003\001\003Â wÂ²'\016Ñ¼Ã033ÃLÃ021Ã)Ã\220hnLÂ¦\020XÃª\236\200\tIhÃÂ¸Sf\234Ã\rÃ ?Â¯Â¼\024FÃ\200Âª/\2264RÃaÃÂ¤IpXÂ»Â¨Ã035Ã¤Qh\211Ã026Ã½4ÃU+\asrÂ£/`Â¤OÃ»1AÊ¹qUÜ¿Â·vÂ±Ã¼\205"..., pos = 0, used = 0, size = 65536} tls_c = (struct tls_extra_data *) 0xb6208e58 enc_rd_buf = (struct tls_rd_buf *) 0x0 n = 0 flush_flags = <value optimized out> err_src = 0x541a83 "TLS read:" x = <value optimized out> #15 0x08158ade in tcp_read_headers (c=0xb621af88, read_flags=0xbf91e200) at tcp_read.c:406 bytes = -1 remaining = <value optimized out> p = <value optimized out> r = (struct tcp_req *) 0xb621affc #16 0x08158fd4 in tcp_read_req (con=0xb621af88, bytes_read=0xbf91e204, read_flags=0xbf91e200) at tcp_read.c:871 bytes = <value optimized out> total_bytes = 0 resp = <value optimized out> size = <value optimized out> req = (struct tcp_req *) 0xb621affc dst = {send_sock = 0x1, to = {s = {sa_family = 57540, sa_data = "\221Â¿\020\000\000\000\000\000\000\000\020\000\000"}, sin = {sin_family = 57540, sin_port = 49041, sin_addr = {s_addr = 16}, sin_zero = "\000\000\000\000\020\000\000"}, sin6 = {sin6_family = 57540, sin6_port = 49041, sin6_flowinfo = 16, sin6_addr = {in6_u = {u6_addr8 = "\000\000\000\000\020\000\000\000\001\000\000\000\001\000\000", u6_addr16 = {0, 0, 16, 0, 1, 0, 1, 0}, u6_addr32 = {0, 16, 1, 1}}}, sin6_scope_id = 12}}, id = -1080958468, proto = 4 '\004', send_flags = {f = 0 '\0', blst_imask = 0 '\0'}} c = 10 '\n' #17 0x08159a0f in handle_io (fm=0x83296b0, events=1, idx=-1) at tcp_read.c:1150 ret = 4 n = 4 read_flags = 1 con = (struct tcp_connection *) 0xb621af88 s = 12 resp = <value optimized out> t = <value optimized out> #18 0x0815d653 in tcp_receive_loop (unix_sock=35) at io_wait.h:1091 No locals. ---Type <return> to continue, or q <return> to quit--- #19 0x0812885b in tcp_init_children () at tcp_main.c:4819 r = 3 reader_fd_1 = 35 pid = <value optimized out> si = <value optimized out> #20 0x080ad412 in main_loop () at main.c:1632 i = 4 pid = <value optimized out> si = (struct socket_info *) 0x0 si_desc = "udp receiver child=3 sock=192.168.166.31:5060 \000\221Â¿\002\000\000\000/\200\220\000\001\000\000\000\000\000H\000Â¨Ã¤21Â¿\001\000\000\000Â°Â¶Ã¨026Â¶\000\000\000\000\000ÃÂµ\030s2\b\002\000\000\000Ã¢f\b\000ÃÂµ\001\000\000\000\000\000\000\000\001\000\000\000\000\000\000\000Ã¿Ã¿Ã¿Ã¿Â¸Ã¤21Â¿" #21 0x080afbe7 in main (argc=3, argv=dwarf2_read_address: Corrupted DWARF expression. ) at main.c:2398 cfg_stream = (FILE *) 0x8a2d008 c = <value optimized out> r = 0 tmp = 0xbf91e674 "\212Ã¾\221Â¿\233Ã¾\221Â¿Â°Ã¾\221Â¿Â»Ã¾\221Â¿Ã\221Â¿Ã¹1Â¿\020Ã¿\221Â¿DÃ¿\221Â¿LÃ¿\221Â¿WÃ¿\221Â¿]Ã¿\221Â¿oÃ¿\221Â¿{Ã¿\221Â¿\202Ã¿\221 tmp_len = 10492229 port = 136192361 proto = -1080957480 ret = <value optimized out> seed = 21782655 rfd = 4 debug_save = <value optimized out> debug_flag = 0 dont_fork_cnt = 0 n_lst = <value optimized out> p = <value optimized out> Best Regards

4 24

ERROR 1045 (28000): Access denied for user 'openser'@'localhost' (using password: YES)
by Shreyansh Purwar 08 Jan '13

08 Jan '13

Hello Everyone, While creating an account on Kamailio Sip server (I am running the command "kamctl add test test1"), it is asking for password of MySQL Server and after entering the password, I am getting following errors :- - ERROR 1045 (28000): Access denied for user 'openser'@'localhost' (using password: YES) - ERROR: introducing the new user 'test' to the database failed Please help me in resolving this issue. Thanks in anticipation. -- Best Regards Shreyansh Purwar

2 1

raw_query is crashing....
by Krishna Kurapati 08 Jan '13

08 Jan '13

Hi, Since there is no way to limit the result set to 1, I tried to use raw_query inside msilo module. Everytime it executes, it is crashing. The same query if I run in mysql shell, it is returning successfully. Here is the code That I added: query_len = snprintf(query, 2048, "SELECT * FROM %.*s WHERE %.*s='%.*s' AND %.*s='%.*s' AND %.*s!=%i AND %.*s<%i LIMIT 1;", ms_db_table.len, ms_db_table.s, sc_uri_touser.len, sc_uri_touser.s, puri.user.len, puri.user.s, sc_uri_tohost.len, sc_uri_tohost.s, puri.host.len, puri.host.s, sc_status.len, sc_status.s, 200, sc_status.len, sc_status.s, MSG_NOTIFY_OFFSET); if (query_len < 0) { LM_ERR("error in sql snprintf"); goto error; } LM_DBG("m_dump_msg: Query=%s\n", query); query_str.s = query; query_str.len = query_len; if (msilo_dbf.raw_query(db_con, &query_str, &db_res) < 0) { LM_ERR("Failed to query database. Query = %s\n", query); goto error; } The Debug Log at Level 4 shows following: 11(7520) DEBUG: msilo [msilo.c:1120]: m_dump_msg: Query=SELECT * FROM silo WHERE tousername='968257197' AND todomain='50.62.1.7' AND status!=200 AND status<1000 LIMIT 1; 11(7520) DEBUG: <core> [db_res.c:118]: allocate 48 bytes for result set at 0x7fdd4b605540 11(7520) DEBUG: db_mysql [km_res.c:68]: 16 columns returned from the query 11(7520) DEBUG: <core> [db_res.c:155]: allocate 128 bytes for result names at 0x7fdd4b5d8dc0 11(7520) DEBUG: <core> [db_res.c:165]: allocate 64 bytes for result types at 0x7fdd4b5d8e60 11(7520) DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[0] at 0x7fdd4b5d8d90 11(7520) DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x7fdd4b5d8d90)[0]=[id] 11(7520) DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type 11(7520) DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[1] at 0x7fdd4b5d8d60 11(7520) DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x7fdd4b5d8d60)[1]=[src_addr] 11(7520) DEBUG: db_mysql [km_res.c:136]: use DB1_STRING result type 11(7520) DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[2] at 0x7fdd4b5d8d30 11(7520) DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x7fdd4b5d8d30)[2]=[dst_addr] 11(7520) DEBUG: db_mysql [km_res.c:136]: use DB1_STRING result type <skip> 11(7520) DEBUG: <core> [db_val.c:117]: converting STRING [ac3aeddd9b6f45e6822c824aa2c133b2] 11(7520) DEBUG: <core> [db_val.c:73]: converting INT [0] 11(7520) DEBUG: <core> [db_val.c:73]: converting INT [1] 26(7552) : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 22 26(7552) DEBUG: <core> [tcp_main.c:3592]: DBG: handle_ser_child: dead child 11, pid 7520 (shutting down?) 26(7552) DEBUG: <core> [io_wait.h:617]: DBG: io_watch_del (0x842d20, 22, -1, 0x0) fd_no=44 called 26(7552) DEBUG: <core> [tcp_main.c:3353]: DBG: handle_tcp_child: dead tcp child 1 (pid 7520, no 11) (shutting down?) 26(7552) DEBUG: <core> [io_wait.h:617]: DBG: io_watch_del (0x842d20, 25, -1, 0x0) fd_no=43 called 0(7509) ALERT: <core> [main.c:787]: child process 7520 exited by a signal 11 0(7509) ALERT: <core> [main.c:790]: core was not generated 0(7509) INFO: <core> [main.c:802]: INFO: terminating due to SIGCHLD As you can see something happening to Database connection. Not sure why I am unable to use raw_query. Normal API query from msilo works just fine. What am I doing wrong to make it crash? Thanks Krish Kura

2 1

Having "modparam" options in the database
by Glenn O Larsen 08 Jan '13

08 Jan '13

Hi guys, I have tried to Google this for some time now, but can't seem to find an answer. Is there a way to read modparam config from a database? -- Best regards, Glenn Larsen

3 3

LCR weight
by Ricardo Martinez 07 Jan '13

07 Jan '13

Hello. I have a doubt about the WEIGHT value in the LCR module. In the past we used the Kamailio version 1.5.2 and in that version LCR weight was calculated by a little script called lcr_weight_script.php. There was not “percent” values like 50% and 50% for two gateways despite that the probabilities were numbers calculated by the script. My question is : For Kamailio version 3.2.4 do I have to use the same script to calculate the weight?? I was almost sure that the weight column was a probability value. For example if I have 3 gateways, with 50% 25% and 25% The weight column should be : 50, 25 and 25 ? or 45, 32 and 32 as the script indicates??? [root@vlcr utils]# ./lcr_weight_test.sh 45 32 32 weight 45 probability 0.5235 weight 32 probability 0.2352 weight 32 probability 0.2413 Please enlighten me here. Thanks Ricardo.-

2 1

TCP Path received quotes
by Spencer Thomason 07 Jan '13

07 Jan '13

Hello, I'm forwarding registrations to Freeswitch and adding a Path header using add_path_received() for endpoints behind NAT. When TCP is used, this results in a header like this: Path: <sip:a.a.a.a:5070;lr;received="sip:b.b.b.b:35430;transport=tcp"> Freeswitch doesn't like the transport protocol inside the quotations and refuses to route a call to the registered user. Constructing the header manually without the quotes such as: append_hf("Path: <sip:$Ri:$Rp;lr;received=sip:$si:$sp;transport=$pr>\r\n"); solves the problem. Is there a reason for the quotes? Best Regards, Spencer

4 7

Multi tenant Kamailio
by Rumen Mihailov 07 Jan '13

07 Jan '13

Hello Kamailio, VoIP and asterisk gurus, I have the following really complicated scenario to solve. I have CompanyA which has extensions 100-120 I have CompanyB which has extensions 100-115 I have CompanyC which has extensions 100-130 Now I want to have all those extensions on a single kamailio box, but I want to achieve the following. CompanyA calls 110 and this rings CompanyA 110 not CompanyB or CompanyC. What I want to achive is multitenancy. I know how to make this in asterisk but I want to take advantage of Kamailio's load balancing and HA features. Sammy from the mailing lists advised that I use multidomain and actually I saw that this module is indeed intended to do that, however I have no clue how to use it. I mean do I have to setup a domain for all the companies ? For instance companya.itradebg.com, companyb.itradebg.com and then advise the customers to register to their respective domain or what ? This concept is really a bit tough for me despite all the help from Sammy for which I am really greatful. I would greatly appreciate any tutorials or links to some useful documentation apart from the module documentation. Thank you in advance guys. Best regards, Rumen http://itradebg.com

2 2

NAT'ed RLS setup
by Dmytro Bogovych 07 Jan '13

07 Jan '13

Greetings. I have kamailio 3.3.3 setup on server behind the nat. Nat forwards ports for tls/tcp/udp. This setup worked without rls ok. I added necessary modules - rls, pua, xcap etc. Updated a config file and db. Client application uploads xcap documents ok; after app sends subscribe request to list address. Rls/pua modules generate subscriptions to entries in list. But dns resolver finds the public ip as target for subscription. The result is: sip packets cannot reach the server - they are sent to public ip of nat machine. I see it in ngrep output. Nat does not route them back. I add the domain name and _sip._udp.domain with internal ip address to /etc/hosts. But kamailio does not use /etc/hosts when dns caching is active. I do not want to loose dns caching. I made an attempt to run without caching but server was not stable. It crashed couple of times - so i stopped these attempts. Is there way to put eternal domain -> ip bindings into dns cache? I want to put bingings to internal ip - hope it will help. Maybe there is another way to handle this problem?

1 0

Timeout for t_uac_dlg per request
by David 07 Jan '13

07 Jan '13

Hello, I am running Kamailio 3.2.x, I am using udp mi to send a t_uac_dlg command, :t_uac_dlg: NOTIFY sip:phone1@proxy.lan . . "From:sip:daemon@proxy.lan To:<sip:phone1@proxy.lan> Contact:sip:daemon@proxy.lan" # I would like to know if I can change the timeout for this request. Running a packet sniffer on the proxy, I see the packet is retransmitted after 500ms, then 1000ms, then 2000ms, etc... This is the proper behaviour for SIP, so we can see my tm module is working well. My problem is that I want a response within 3 seconds ( I know this doesn't respect the normal SIP timer values ). I do not want to change the timers in the tm module for the entire proxy, so my question is if there is a way to set the timeout for the dialog created by the t_uac_dlg command, and can I set it on a per dialog basis ? David

1 0

TLS with self-singned certificates
by Iurii Andamasov 07 Jan '13

07 Jan '13

Hello, trying setup kamailio with TLS, have 2 peers, tls.cfg: http://pastebin.com/DvdDzx0v i'm getting Jan 4 10:38:43 fs-tls /usr/sbin/kamailio[3798]: ERROR: tls [tls_server.c:1190]: TLS read:error:14094410:SSL routines:SSL3_READ_BYTES:sslv3 alert handshake failure Can someone point me to solution? Thanks

3 4

tls_domain.h:37: fatal error: openssl/ssl.h: No such file or directory
by Shreyansh Purwar 07 Jan '13

07 Jan '13

Hello Everyone, While building kamailio server 3.0.0 with TLS(I have removed tls from exclude module variable) I am getting following error :- - tls_domain.h:37: fatal error: openssl/ssl.h: No such file or directory Please help me how to resolve this issue. Thanks in anticipation. -- Best Regards Shreyansh Purwar

2 1

Malicious traffic comming from 37.75.210.90
by Nick Khamis 06 Jan '13

06 Jan '13

Hello Osama, and Hisham, At 1330GMT there was some malicious activity coming from your network IP 37.75.210.90. Please act accordingly. Things that may be of use "972599779558" N.

1 0

dialog ka_timer and ka_interval problem
by Rinor Hoxha 06 Jan '13

06 Jan '13

Hi, I had an issue with dialog module of kamailio v3.3.3. Is the same described on http://lists.sip-router.org/pipermail/sr-users/2012-November/075735.html. I think the issue is related to dgl_hash.c on line:192 *dlg_ka_list_head = NULL; should be *dlg_ka_list_tail = NULL; and line:216 *dlg_ka_list_tail = dka; should be *dlg_ka_list_head = dka;. I will test it as soon as I can, but meanwhile I'm attaching the patch that should fix it so you can test it and report back the results if you want to. Br, Rinor

2 1

Kamailio DB version error
by Douglas Ugalde 06 Jan '13

06 Jan '13

Hi, I want to load the "dialplan.so" with parameters but when I reload kamailio, I got this error: /usr/local/sbin/kamailio[1348]: ERROR: <core> [db.c:439]: invalid version 0 for table my_table found, expected 1 (check table structure and table "version") Im using kamailio 3.3.3 Please what can I do to resolve it? Regards.

2 1

Next major release proposed as v4.0.0
by Daniel-Constantin Mierla 04 Jan '13

04 Jan '13

Hello, using v4.0.0 for the next major release was proposed and debated at previous development IRC meeting. First was about the addition of a new transport layer (websockets) along with other important new features. During the debate was touched the completion of kamailio-ser integration (no more duplicated modules) as another milestone for increasing the first digit (which was finished just before the end of last year). Now we also have the IMS modules, pushed in the GIT master branch few days ago. It seems that there are lot of good reasons for numbering the next major version 4.0.0. Other comments, opinions? Have a great and happy year ahead, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

4 4

Multiple rtimer processes + mqueue
by Alex Balashov 04 Jan '13

04 Jan '13

Will multiple rtimer processes consuming the same mqueue, i.e. via a blocking mechanism while(mq_fetch(...)) ... hurt anything? -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

2 1

Freezing for next major release
by Daniel-Constantin Mierla 04 Jan '13

04 Jan '13

Hello, based on last irc devel meeting and feedback afterwards, most of the people seem to be fine releasing next major release a bit earlier, in order to: - make websockets support and the other new features available in a stable version - allow proper development time frame for other planned new features that wouldn't have been ready if the release would be 1-2 months later So, considering the winter holidays, I propose to freeze development on January 7, 2013, with full release about one month later. It seems that many people are already using devel version to play with websockets, meaning that we are staying very well with testing. Other opinions? Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

3 5

"maximum number of branches exceeded" after kamalilio restart
by Andrew Pogrebennyk 04 Jan '13

04 Jan '13

Hello, I'm getting some errors - if I restart kamailio on a live system, the first call after restart fails with "max nr of branches exceeded". However, we are not doing append_branch() or anything fancy for that call at all: Jan 3 15:26:54 sp1 /usr/sbin/kamailio[8078]: ERROR: <core> [dset.c:306]: max nr of branches exceeded Jan 3 15:26:54 sp1 /usr/sbin/kamailio[8078]: ERROR: registrar [lookup.c:298]: failed to append a branch Also it appears on the other system with a different config: Jan 3 17:06:59 sip /usr/sbin/kamailio[7022]: ERROR: tm [t_fwd.c:656]: ERROR: add_uac: maximum number of branches exceeded Jan 3 17:06:59 sip /usr/sbin/kamailio[7022]: ERROR: tm [t_fwd.c:1534]: ERROR: t_forward_nonack: failure to add branches Jan 3 17:06:59 sip /usr/sbin/kamailio[7022]: ERROR: tm [tm.c:1369]: ERROR: w_t_relay_to: t_relay_to failed The second call however goes through and this error does not appear until the next kamailio restart. What could be the problem here? P.S. The kamailio version is 3.3.2. Andrew

1 0

km_dbase.c:38: fatal error: mysql/mysql.h: No such file or directory
by Shreyansh Purwar 04 Jan '13

04 Jan '13

Hello Everyone, I need a help, I was installing kamailio SIP server according to the instruction provided at wiki page - http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.0.x-from-git but while compiling it(make all) i am getting following error - km_dbase.c:38: fatal error: mysql/mysql.h: No such file or directory Can anybody tell me how to resolve this issue. Thanks in anticipation. -- Best Regards Shreyansh Purwar

2 1

Re: [SR-Users] [sr-dev] Freezing for next major release
by Peter Dunkley 03 Jan '13

03 Jan '13

I hope to get the outbound edge proxy and flow timer stuff into master by Monday, but it could really do with some additional testing especially in conjunction with the reg-id stuff in registrar/usrloc (which I have no idea how to use). Peter Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: >Hello, > >a reminder that the freezing date of the development for the next major >release is next Monday, January 7. > >Considering it was vacation time so far, we can prolong development for >few days if it is needed - just reply here. > >Besides few small additions, I want to add internal hash table to track >MSRP connections, hopefully I should be able to do it in time. If not, I >will ask for few more days as well. > >Cheers, >Daniel > >On 12/5/12 5:05 PM, Daniel-Constantin Mierla wrote: >> Hello, >> >> based on last irc devel meeting and feedback afterwards, most of the >> people seem to be fine releasing next major release a bit earlier, in >> order to: >> - make websockets support and the other new features available in a >> stable version >> - allow proper development time frame for other planned new features >> that wouldn't have been ready if the release would be 1-2 months later >> >> So, considering the winter holidays, I propose to freeze development >> on January 7, 2013, with full release about one month later. >> >> It seems that many people are already using devel version to play with >> websockets, meaning that we are staying very well with testing. >> >> Other opinions? >> >> Cheers, >> Daniel >> > >-- >Daniel-Constantin Mierla - http://www.asipto.com >http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > > >_______________________________________________ >sr-dev mailing list >sr-dev(a)lists.sip-router.org >http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev

1 0

recreating kamilio_fifo without restart
by Uri Shacked 03 Jan '13

03 Jan '13

Hi, I entered the command "kamailio -v" instead of "kamailio -V". It looks like it starts kamailio again even that it was running.... the problem is that the kamctl_fifo and kamctl_ctl file are gone now. how do i create the files again without restarting kamailio? Thanks, Uri

3 4

All The Best And A Happy 2013!
by Daniel-Constantin Mierla 31 Dec '12

31 Dec '12

This was it for 2012, one of the greatest years in the history of the project! One major release was done in the summer (version 3.3.x), another one is few days before testing phase, most probably to be out by mid of February. The next major release includes several great features and edevelopments: - support for websockets - IMS extensions in the main GIT branch - no more duplicated modules, integration of Kamailio and SER ended There are big plans for 2013. First is the next major release, together with participation at FOSDEM conference in Berlin. A bunch of other events will open the path to our first dedicated conference for Kamailio project - Kamailio World. We hope to meet many of you at Kamailio World conference, in Berlin, Germany, during April 16-17. Last organizing bits were sorted out and registration will open in the first days of 2013 -- you can see more details at: - http://conference.kamailio.com Looking forward to a great 2013 for Kamailio and everyone involved in the project, from developers to community members! All the best! Happy New Year! Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Kamailio TLS setup and certificate
by Burak ÖZTÜRK 31 Dec '12

31 Dec '12

Hi, I m a new kamailio user. Kamailio version is 3.3. I want to enable kamailio's tls support. I fallowed instructions from this link : http://www.kamailio.org/dokuwiki/doku.php/tls:create-certificates And I also enabled tls module in the kamailio.cfg file. (definig WITH_TLS and adding listen=tls:myIP:5061 command) But after trying to start kamailio i got "failed to start" error. When I checked the logs there is nothing about tls or ssl. Actually I got an error while fallowing the link above. "openssl s_client -connect localhost:5061 -tls1" command returns connection failed 111 error. I thought that this is the reason failed to start. But where or what am I doing wrong? I searched the web and hear about SRV records. But there is no explanations about this, on your guide/wiki. I also read this link too but cant get my answer : http://sip-router.org/docbook/sip-router/branch/master/modules/tls/tls.html Is there any idea or a better well-explained user guide about tls support? Regards, Burak

2 1

Zero left padding IP in Kamailio
by Techie Sup 30 Dec '12

30 Dec '12

Hello, My IP is 192.168.1.25 and I have added left zero padding i,e: 192.168.001.025 in the configuration file and try to start kamailio, it works.. but along with that rtpproxy does not start. Once I revert back it works. My script requires to add zero left padding for a specific purpose and for that I cannot remove the left zero padding. Any solution to get this work? Regards Selvam

2 1

kamailio-ser merging finished
by Daniel-Constantin Mierla 29 Dec '12

29 Dec '12

Hello, in November 2008 was decided to start integration of kamailio and ser project, to get to same source code. About one year later, there was a single source code tree, but there were several duplicated modules that kept some confusion around the two projects. As the time passed, it was shown better what modules are in the focus, making the decisions to sort them out easier. The past weeks were dedicated to removing duplicated, process that just ended -- at this moment there is no duplicated module, all of them being integrated. In summary, the modules from ser that had no conflict were moved as is to modules folder, eventually after rename. The modules with conflicts were merged, adding missing features to the most developed ones, moving the others to obsolete folder. It happened that kamailio version of the modules stayed in the majority of the cases, because they had larger set of features (the core in v3.0.0 was kept from ser). Please note that the extra features from ser modules might not be in the same form, but alternatives that should offer same functionality at the end. People using ser flavour so far should start testing the master branch. Do ask questions on the mailing lists if something that you used to have in config is not longer available, so we can assist in migration and, hopefully, create a migration guide on wiki. There might few bits and pieces to tune here and there, soon we will get into code freezing period for testing the next major release, a good time for such adjustments. It is important that everyone using former ser modules to report the issues that are found. Many thanks to all those involved in integration efforts as well as in new developments that kept the project flying! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

GROUP - change database table
by Daniel W. Graham 28 Dec '12

28 Dec '12

I am attempting to move the location of the GROUP 'grp' table. I created column grp in subscriber table and configured column with same settings as default table and updated column name as follows: modparam("group", "db_url", DBURL) modparam("group", "table", "subscriber") modparam("group", "use_domain", 1) As the other required column names are present in the subscriber table and also default params for this module, I didn't specify them, but I have also tried doing that with same result. I have debugging enabled and when attempting to restart / start kamailio it simply doesn't start and no debug errors are detected, also nothing in log file. If I change to modparam("group", "table", "grp"), kamailio starts fine. Any ideas out there as to what would cause this? Thanks for your time- -dan [cid:image001.gif@01CDDEC4.BE0AD670]

2 4

Is somebody at 29C3 in Hamburg and wants to meet for a drink and some VoIP gossip?
by Klaus Darilion 28 Dec '12

28 Dec '12

regards Klaus

1 0

No Conect
by Matheus Perillo - Rede Itatiaia 27 Dec '12

27 Dec '12

Good afternoon , My name is Matthew and Perillo on IPtel have an account with the username of djperillo . I am not able to connect with anyone. We have to work in the company of various equipment and all with Comrex IPtel account and you can not connect with any of them . Everything indicates to the page Comrex was made and does not connect . The Access are with the SIP Status On LINE . On my iphone the phone symbol on Media5 is green but when I try to connect with the Comrex Access the following message appears " user / password denied , please check your settings and try again . " What can I do to solve this problem ? Thank you . Matheus Perillo Técnico de Manutenção Tel +55 31 2105.3545 Fax +55 31 2105.3677 matheusperillo(a)itatiaia.com.br www.itatiaia.com.br Rua Itatiaia 117 • Bonfim • Belo Horizonte • MG • Brasil • 31210-170

1 0

retransmitted INVITE's
by schwatzi＠web.de 27 Dec '12

27 Dec '12

2 1

extension of SNMPStats module
by 马天 27 Dec '12

27 Dec '12

Hi all, Merry X'mas. I'm working on a project which uses kamailio as sip server. Since I'm really new to kamailio and sip protocool and c, I need some help.My question is that is it possible to extend snmpstats module so the snmpstats module can respond to customized oid.I already wrote the mib file and generated the corresponding c files which handle the request. But I couldn't compile it into snmpstats module.Always get a "No Such Object available on this agent at this OID" when i send GET request using snmpget.Thanks. Regards, Sebastian

2 1

OPTIONS packet from dispatcher module
by Morten Isaksen 27 Dec '12

27 Dec '12

Hi, I have an issue with the OPTIONS packets from the dispatcher module does not contain a max-forward header. This causes the gateway in the other end to not reply to the OPTIONS packet. Can anyone please give a hint where en the code I need to change that. I noticed that dispatcher module uses the tm module to send the OPTIONS but after that I got lost in the code. -- Morten Isaksen

2 1

Problem with RLS
by Dmytro Bogovych 26 Dec '12

26 Dec '12

Hi. I still try to setup RLS capable kamailio instance. Now the server accepts my services and resource list documents. It generated backend subscriptions - i can see it in pua table. When another peer publish its presence - presence tries to send notify to sip address specified in modparam("rls", "server_address", " sip:rls@sip.voipobjects.com"). But the notify does not come back to rls - it is rejected by somewhere in core or other module with error 404. I tried to create the rls(a)sip.voipobjects.com user - it did not help. May anyone give me advice?

1 1

Presence rules?
by Dmytro Bogovych 25 Dec '12

25 Dec '12

Greetings. I try to add rls/xcap presence into resiprocate-based user agent. May anyone tell me what module is responsible for parsing presence rules documents? I'm trying to setup "allow all" presence rules document but did not get success... Thank you!

2 4

Merry Christmas!!!
by Raj Roy Ghandhi 24 Dec '12

24 Dec '12

I WISH Merry Christmas!!! to my Kamailio FAMILY !!!! Best Regards, Roy.

1 0

obsoleting modules_s/*radius modules
by Daniel-Constantin Mierla 24 Dec '12

24 Dec '12

Hello, according to the archive, modules_s/*radius modules were kept because of duplicate and incompatible auth modules at the time there was an attempt to remove them: - http://lists.sip-router.org/pipermail/sr-users/2010-June/064318.html Now there is a single auth modules, thus I guess the modules_s/*radius can be moved to obsolete. If anyone has different opinion, reply with the arguments here. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

6 15

Queries
by Ankit sharma 24 Dec '12

24 Dec '12

Hi, I have newly installed Siremis sip router. But I have no idea How I could start or How do I start calls? If you can provide me any Manual so it would be great. Sincerely Ankit

4 3

How to set variable externally
by Mino Haluz 22 Dec '12

22 Dec '12

Hi, I would like to set my custom different debug levels (with flag?) externally with kamctl command. So I neednt restart kamailio if I want to enable/disable debug. Which module should I use in that case? Thanks, Mino

6 12

AUTOREPLY Re: Problemdebian squeze uac module load failed
by strobel＠hfo-telecom.de 21 Dec '12

21 Dec '12

******************* AUTOREPLY VON miconda(a)gmail.com ***************************** Vielen Dank für Ihre Nachricht. Leider bin ich in der Zeit vom 27.12.2012 bis einschließlich 01.01.2013 nicht im Haus. Ihre E-Mail wird in meiner Abwesenheit nicht weitergeleitet. Bitte wenden Sie sich in dringenden Fällen an meine Kollegen Herrn Brunner oder Herrn Schrenk. eMail: technik(a)hfo-telecom.de Phone: +49 9281 / 1448 - 100 Fax: +49 9281 / 1448 - 123 Mit freundlichen Grüßen, Rainer Strobel Network Management HFO Telecom AG Bahnhofstr. 18 D-95028 Hof / Saale Hello, do you get another error message before? If not, set debug=3 in kamailio.cfg and run again. Then look at the logs to see if there is another message giving hints about what is wrong. Probably some mis-configured parameter. Cheers, Daniel On 12/21/12 4:38 PM, Thomas Hoellriegel wrote: > Hi all, i using debian squeeze. > I installled kamailio from the package system. > I set in the kamailio.cfg file: > loadmodule "uac.so" > I become the error: > Dec 21 16:37:35 1a-7567 > /usr/sbin/kamailio[3015]: ERROR: <core> [sr_module.c:939]: init_mod(): > Error while initializing module uac > (/usr/lib64/kamailio/modules_k/uac.so) > > Can you help plese? > thanks. > > --------------- > Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: > http://www.blindi.net/callback > homepage: http://www.blindi.net > blinde-misc mailingliste für blinde. anmeldung unter: > http://www.blindi.net/mailman/listinfo/blinde-misc > > > _______________________________________________ > 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

1 0

RPMs for v3.3.3
by Daniel-Constantin Mierla 20 Dec '12

20 Dec '12

Hello, for the records, the RPMs for v3.3.3 are now available (32b and 64b: CentOS 5 and 6, RedHat 5.0 and 6.0, OpenSUSE 11.4 and 12.1, Fedora 15 and 16). They are hosted by opensuse build service, links specific per distribution are listed at: * http://www.kamailio.org/wiki/packages/rpms Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

Re: [SR-Users] [sr-dev] kamailio-ser integration - db modules
by Ovidiu Sas 20 Dec '12

20 Dec '12

On Thu, Dec 20, 2012 at 11:20 AM, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: >> What's the difference between the db1 and db2 interface? > > It is not the case here, modules_k/ldap does not have any relation to > database interface. Any database module db_[DatabaseType] implements an API. SER was using one API and opensips/kamailio a different API (and therefore today we have version 1 and 2). All db modules in the stable release are supporting both APIs. The new db ldap module has support only for the SER API version. The API is documented in each lib/srdb[1|]/*.h header files. The ldap module from ser is completely different than ldap module from kamailio (despite the fact that both are using the same name). It's the same with the dialog module. Check the README file for each. Hope this brings a little bit of light :) -ovidiu

1 0

Re: [SR-Users] [sr-dev] kamailio-ser integration - status update
by Daniel-Constantin Mierla 20 Dec '12

20 Dec '12

On 12/20/12 4:13 PM, Ovidiu Sas wrote: > On Thu, Dec 20, 2012 at 9:39 AM, Daniel-Constantin Mierla > <miconda(a)gmail.com> wrote: >>>> - nathelper - some extra functionality, not sure if can be kept >>>> completely >>> Maybe Andreas can look into this, as there is a lot of work going on with >>> nathelper and the new rtpproxy anyways. >> I think Ovidiu Sas looked at it when he split the rtpproxy out in a >> dedicated module. > IIRC, ping_contact whas the extra functionality in nathelper: > http://sip-router.org/docbook/sip-router/branch/master/modules_s/nathelper/… > I don't know how widely used is this functionality. > > Maybe we should have a separate thread per module (in user mailing > list to gather more imput) and see if it's worth merging the code or > use only the k version. I re-cc-ed the thread to users in case someone has comments to it. > > Also, on a separate note, I saw the we have a few db2_[module]. > I think it would make sense for these modules to rename them into > [module]_db[1|2]. > For example: ldap - we should have both versions under modules: > - ldap_db1 > - ldap_db2 > Just a suggestion ... The type of the two ldap modules are different, modules_k/ldap is a connector to ldap server from configuration file, offering possibility to do ldap search queries from config. The former modules_s/ldap (now db2_ldap) is a DB API v2 implementation driver module, so it can be used as a replacement for db_mysql (for example) when using some modules (such as db2_ops). I prefixed with db2_ to indicate that is not implementing DB API v1. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

2 1

kamailio-ser integration - status update
by Daniel-Constantin Mierla 20 Dec '12

20 Dec '12

Hello, most of the duplicated or ser-specific modules were sorted out at this time, next is the situation with the remaining ones. A) modules that will be renamed using a prefix 'uid_', because they have a database schema using unique id instead of username and domain: - auth_db => uid_auth_db - avp_db => uid_avp_db - domain => uid_domain - gflags => uid_gflags (not documented, but this module loads so called ser global avps) - uri_db => uid_uri_db B) modules to become obsolete with some work to be done - cpl-c - seems that giving a parameter with what user to be used for loading the cpl scripts will make k and s versions compatible - maxfwd - different names for config functions - nathelper - some extra functionality, not sure if can be kept completely - textops - seems to have some extra cfg functions - pike - has a rpc command - registrar/usrloc - avps stored per contact C) modules to become obsolete, most probably without further work - permissions - different structures/database table internally, but all should be achieved using k version (data migration might be required) D) modules that needs further investigations - rr - there seems to be some feature related to avps and record-route parameters/cookies - this module could be just renamed at the end (like rrs) E) Sample modules, not for real use - they can be kept in this directory (modules_s[ample]) - print - print_lib If anyone has comments regarding these remarks, reply here. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

2 2

Bug in uac_replace_from if used more than once?
by Daniel Tryba 20 Dec '12

20 Dec '12

Apparantly it should be simple to mangle the From in request: http://kamailio.org/docs/modules/stable/modules_k/uac.html#id2525538 The task at hand is use the prefix 131 to suppress callerid for this call: if($rU=~"^131[0-9]+$") { strip(3); append_hf("Privacy: id\r\n"); uac_replace_from("anonymous","sip:anonymous@anonymous.invalid:$rp"); uac_replace_to("","sip:$rU@$rd"); } Just before this replace there is an other uac_replace_from to rewrite an alphanumeric username to the default callerid for that device. If there are 2 uac_replace_from in the path, things go wrong like below. A uac_restore_from() before the second uac_replace_from doesn't change this behavior. device calls 1310123456789: U 10.0.34.226:1300 -> 10.0.35.226:5060 INVITE sip:1310123456789@kamailio.local:5060 SIP/2.0. From: <sip:anonymous@10.0.3.44:5060>;tag=26e5ce5850. To: <sip:1310123456789@kamailio.local:5060>. Kamailio forwards request to dispatcher but fails to properly replace the from uri: U 10.0.35.226:5060 -> 10.0.32.36:5060 INVITE sip:+31123456789@kamailio.local:5060 SIP/2.0. From: "anonymous" <sip: +31987654321@10.0.3.44:5060sip:anonymous@anonymous.invalid:5060>;tag=26e5ce5850. To: <sip:0123456789@kamailio.local>. Even though the debug log suggests it did the proper replacement: uac [replace.c:265]: value to store is is '<sip:anonymous@10.0.3.44:5060>' and len is '30' uac [replace.c:268]: Storing in FROM-AVP (for use in reply): '<sip:anonymous@10.0.3.44:5060>' with len '30' uac [replace.c:324]: uri to replace [sip:anonymous@10.0.3.44:5060] uac [replace.c:325]: replacement uri is [sip:anonymous@anonymous.invalid:5060] uac [replace.c:383]: encode is=<AAAAAAAAAAAAAAAAAABQXkFeV15BQUcUXF5AUWxpZDo1MDYw> len=48 uac [replace.c:265]: value to store is is '<sip:1310619024127@kamailio.local:5060>' and len is '54' uac [replace.c:271]: Storing in TO-AVP (for use in reply): '<sip:1310123456789@kamailio.local:5060>' with len '54' uac [replace.c:324]: uri to replace [sip:1310123456789@kamailio.local:5060] uac [replace.c:325]: replacement uri is [sip:0123456789@kamailio.local] uac [replace.c:383]: encode is=<AAAAAAEFAAkGAw0BAANxQlYuERJDCRxeHAoXWgYRXRdbAxtNHxwNCgdACxgubmw6NTA2MA--> len=72 Kamailio version: kamailio 3.3.2 (x86_64/linux) (the latest stable debian/squeeze package). -- POCOS B.V. - Croy 9c - 5653 LC Eindhoven Telefoon: 040 293 8661 - Fax: 040 293 8658 http://www.pocos.nl/ - http://www.sipo.nl/ K.v.K. Eindhoven 17097024

3 5

Changing METHOD
by Olle E. Johansson 20 Dec '12

20 Dec '12

Hi! I want to change METHOD on a SIP message from UPDATE to INVITE or REGISTER to INFO. Seems like $rm is read-only. Any other known way to change a requests method? /O

4 8

Send reply with Reason header
by Mino Haluz 19 Dec '12

19 Dec '12

Hi, I know how to use sl_send_reply, but I would like to add "Reason:" header to the reply generated. Is there any command or variable for that? Thank you, Mino

3 3

Video conference with many users
by Raj Roy Ghandhi 19 Dec '12

19 Dec '12

Hi, I want to do web (HTML5 + WebRTC) Sip client which can do the video conference with multiple users. Current release of SIPML <http://www.sipml5.org/> does 1 to 1 call. I have no idea of conference with many users. Is it the client that we need to modify to accept call and join the conference ? Do I need to send INVITE with extra parameters ? Please advice me. Best Regards, Roy.

3 2

Question registering sip providers in kamailio and forward all?
by Thomas Hoellriegel 19 Dec '12

19 Dec '12

Hi all, I.m. a newbi in Kamailio. My problem: I.m. a blind user and i programmed a voicechat in Freeswitch. 3 have 2 sip providers. vodafone.de sipgate.at and freenet.de. The problem: i must register these proviers in kamailio. And forward all incoming calls to Freeswitch, and outgoing calls from freeswitch to kamailio. Dtmf modes are: auto rfc2833 and inband. Then i call: dorf(a)dorf.blindi.net i like to forward these calls to my freeswitch box. the kamailio ip ist: 217.172.180.108 and the freeswitchbox: 217.172.170.120 Can your help me please? Thanks. --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste für blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc

2 1

NAT Traversal issue
by Raj Roy Ghandhi 19 Dec '12

19 Dec '12

Hi, My Kamalio development version works very well with websocket and webrtc clients. But when I try to call the guy in remote area (he had connected to the same server with 3G dongle) no voice and video. Here is how I have set it up. 1. Kamailio 3.4 development version running on public IP 2. NAT Traversal is done with RTPProxy 1.2. 3. IP Phones work very well. (phones are behind NAT) 4. Web page with WebRTC works well in LAN behind the NAT But I try to call a account which in logged into same Kamailio server we do not hear voice nor media. I have attached the sip capture into 2 files 1. LAN webrtc client->LAN client web page call 2. LAN webrtc client -> 3G Dongle webrtc client Please help me out to figure this out. Best Regards, Roy.

2 2

New developer: Richard Good
by Daniel-Constantin Mierla 19 Dec '12

19 Dec '12

Hello, I want to announce that a new developer got GIT write access to repository: Richard Good - he is among those developing the IMS extensions for Kamailio (http://code.google.com/p/smile-ims-extensions/), to be merged in the next days to the GIT master branch. His git commit id is: richard.good My warm welcome and looking forward to future work within the project! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

3 2

modules_s/timer review
by Daniel-Constantin Mierla 19 Dec '12

19 Dec '12

Hello, I was comparing modules_s/timer with modules_k/rtimer to spot the differences. modules_s/timer is working only with core timers (fast and slow timers), while the modules_k/rtimer can start own timer processes, or register a task to main core timer. Other differences ... modules_s/timer: - run on milisecond basis - option to enable/disable timer modules_k/rtimer - run either on second or micro-second basis - option to execute the many route blocks on same timer So, apart of enable/disable timer option, rtimer can do more than what timer module offer (mili-seconds can be run as 1000x micro-second). Enabling/disabling a timer can be workarounded from script, using a shared variable tested at begininning of rtimer route, like: route[RTIMER] { if($shv(rtimer)==0) return; ... } Enabling will mean setting $shv(rtimer)=1, either from other parts of the config or via MI/RPC commands. Now, both modules can be kept, no naming conflict, it is just a matter of code maintenance if we should keep timer module. Enabling/disabling timers can be added to rtimer in the future, but it is unlikely to happen before next major release (unless someone else does it). Is anyone here using modules_s/timer module aware of other differences comparing with rtimer? Looking for comments to keep or obsolete modules_s/timer module ... Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Kamailio v3.3.3 Released
by Daniel-Constantin Mierla 18 Dec '12

18 Dec '12

Hello, Kamailio SIP Server v3.3.3 stable release is out. This is a maintenance release of the latest stable branch, 3.3, that includes fixes since release of v3.3.3. There is no change to database schema or configuration language structure that you have to do on installations of v3.3.0, v3.3.1 or v3.3.2. Deployments running previous v3.x.x versions are strongly recommended to be upgraded to v3.3.3. For more details about version 3.3.3 (including links and hints to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2012/12/kamailio-v3-3-3-released/ RPM, Debian/Ubuntu packages will be available soon as well. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Kamailio v3.3.2 Released
by Daniel-Constantin Mierla 18 Dec '12

18 Dec '12

Hello, Kamailio SIP Server v3.3.3 stable release is out. This is a maintenance release of the latest stable branch, 3.3, that includes fixes since release of v3.3.3. There is no change to database schema or configuration language structure that you have to do on installations of v3.3.0, v3.3.1 or v3.3.2. Deployments running previous v3.x.x versions are strongly recommended to be upgraded to v3.3.3. For more details about version 3.3.3 (including links and hints to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2012/12/kamailio-v3-3-3-released/ RPM, Debian/Ubuntu packages will be available soon as well. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Planning next minor release - v3.3.3
by Daniel-Constantin Mierla 18 Dec '12

18 Dec '12

Hello, I am considering releasing next minor version from branch 3.3 at the beginning of next week, Monday or Tuesday (Dec 17 or 18). For outstanding issues, please use the tracker or send to sr-dev mailing list. Developers that pushed fixed in the master branch should backport them in 3.3 before the next week starts. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 2

dlg_manage() and add_contact_alias() problem
by Pavel Miskov 18 Dec '12

18 Dec '12

Hi all, I want to end dialog (Kamailio sends BYE to both parties) that is lasting longer then some defined time. everything is working fine when using old fix_nated_contact function, but when using new add_contact_alias function kamailio sends BYE to private addresses. 1. "kamctl fifo dlg_list" output when using fix_nated_contact(): caller_contact:: sip:bob@PUBLIC_IP:50784 callee_contact:: sip:alice@PUBLIC_IP:64472 2. "kamctl fifo dlg_list" output when using add_contact_alias(): caller_contact:: sip:bob@10.2.5.206:10000 callee_contact:: sip:alice@10.2.5.205:20000 dlg_manag() doesn't store contact alias for caller and callee so Kamailio is sending BYE to private addresses. Am I doing something wrong or I can't use possibility of ending dialog with contact aliases? Thank you Pavel

2 3

Support
by Murali Krish 18 Dec '12

18 Dec '12

Hi this is Murali krishnan i am final student of master degree in Dalhousie University, I am Working on my project, i m need of a sip server which can support H.264 codec video calls...I created an account in iptel sip account when i used the account it doesn't support video calls. If i use SER will this support video calls? -- *Regards Murali Krish*

2 1

From sip phone to provider trunk
by andre second 18 Dec '12

18 Dec '12

Hi,I have some sip phones and using them to register at Kamailio which is located behind 2 asterisk servers. There 2 SIP trunks to my SIP provider on asterisk machines. Also I have rtpproxy running. What I want to do is to put some of the calls directly from the phones to SIP Provider without involving asterisk. I think I need to use Dispatcher module - what is the best way of doing that?Thank you!

2 4

p_usrloc registration update problem
by Hans Goossens 18 Dec '12

18 Dec '12

Hi, I'm testing the p_usrloc module (Kamailio version 3.3.2, installed via ubuntu package) and it looks like it's working but for some reason a re-registration is causing a new entry being added to the distributed location table instead of updating the existing one. The kamailio configuration looks like this: #!define DBURL "mysql://openser:openserrw@localhost/openser" loadmodule "p_usrloc.so" modparam("p_usrloc", "db_mode", 3) modparam("p_usrloc", "write_db_url", DBURL) modparam("p_usrloc", "read_db_url", DBURL) modparam("p_usrloc", "write_on_db", 1) modparam("p_usrloc", "alg_location", 1) modparam("p_usrloc", "reg_db_table", "locdb") mysql> select * from locdb; +----+----+-------------------------------------------+--------+--------+---------------------+-------+----+ | id | no | url | status | errors | failover | spare | rg | +----+----+-------------------------------------------+--------+--------+---------------------+-------+----+ | 1 | 1 | mysql://root:secret@172.16.10.131/openser | 1 | 0 | 1900-01-01 00:00:01 | 0 | 0 | | 1 | 2 | mysql://root:secret@172.16.10.132/openser | 1 | 0 | 1900-01-01 00:00:01 | 0 | 0 | +----+----+-------------------------------------------+--------+--------+---------------------+-------+----+ When using the default usrloc functionality everything is working fine. The only difference I noticed is that the default usrloc is using MYSQL UPDATE to update the location entry and that p_usrloc is using MYSQL REPLACE. Regards Hans

1 0

variable in rewritehost ?`
by Ali Jawad 17 Dec '12

17 Dec '12

Hi I am using slqops to get some user related info from location table, I want to use that info in rewritehost, however when I do use a variable in rewritehost I get an error "Unable to resolve variable". Is it possible to use a variable in rewritehost ? If not what is the workaround ? Thanks

3 9

Dialog module not decrementing profile size
by Grant Bagdasarian 17 Dec '12

17 Dec '12

Hello Community, I've recently started to use the Dialog module to manage the amount of incoming calls from our SIP provider allowed to pass through by using profiles_with_values and setting the size of these profiles to a certain amount. Most of the time this works as it should, but sometimes the 'slots' available for a profile aren't cleared even though a dialog has finished. This seems to happen when an high amount of regular traffic is hitting kamailio. Has anyone else experienced this? Are there perhaps any other ways to control the slots available for certain incoming SIP traffic? Below parts of my configuration file. Perhaps I'm doing something wrong which causes this to happen. In my route[WITHINDLG] -> loose_route -> is_method("BYE") I'm not calling dlg_manage(), could this be the problem? Other comments are welcome too, as I'm fairly new to Kamailio. modparam("dialog", "enable_stats", 1) modparam("dialog", "dlg_flag", 4) modparam("dialog", "hash_size", 4096) modparam("dialog", "profiles_with_value", "318005004"); modparam("dialog", "default_timeout", 43200) modparam("dialog", "dlg_match_mode",2) modparam("dialog", "detect_spirals", 1) request_route { # per request initial checks route(REQINIT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE")) { xlog("L_INFO", "Incoming request for $rU from $fU"); if ($rU == "318005004") { route(DIALOG); } } # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations route(DISPATCH); route(RELAY); } route[DIALOG] { dlg_manage(); record_route(); $var(SIZE) = 0; sql_query("vf", "exec dbo.SelectAvailableSlots '00$rU'", "ra"); $avp(s:limit) = $dbr(ra=>[0,7]); xlog("L_INFO", "Maximum simultaneous calls is configured at $avp(s:limit) ...."); sql_result_free("ra"); if ($rU == "318005004") { get_profile_size("318005004", "$rU", "$var(SIZE)"); } xlog("L_INFO", "There are currently $var(SIZE) calls for $rU, excluding the current call"); if($var(SIZE) >= $avp(s:limit)) { xlog("L_INFO", "Simultaneous calls limit of $avp(s:limit) reached for $rU: $var(SIZE)\n"); sl_send_reply("603", "Simultaneous calls limit reached"); exit; } if ($rU == "318005004") { set_dlg_profile("318005004", "$rU"); } if ($rU == "318005004") { if(get_profile_size("318005004", "$var(SIZE)")) { xlog("L_INFO", "There are currently $var(SIZE) calls for $rU, including the current call"); } } } route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { #Used for call transfer if($fU == "318005004") { if (is_method("UPDATE")) { sl_send_reply("405","Method not allowed"); xlog("L_INFO", "Dropping UPDATE message...\n"); exit; } if (is_method("INVITE")) { record_route(); } if (is_method("BYE")) { dlg_manage(); } } if (is_method("BYE")) { setflag(1); # do accounting ... setflag(3); # ... even if the transaction fails } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; # must be ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard. exit; } } sl_send_reply("404","Not here"); } exit; } } Regards, Grant

2 8

status of kamailio stun server?
by Juha Heinanen 16 Dec '12

16 Dec '12

i tried to use kamailio built-in stun server, but failed, because it supports only binding requests. my ua tried to send it "allocate request lifetime" request, which failed with 600 global failure response. what is the status of kamailio's stun server? is anyone interested in developing it further or should we obsolete the server and start using some external server? -- juha

2 5

Writing User-preferences Username into Radacct Table on Call Forward
by Jonathan Hunter 15 Dec '12

15 Dec '12

Hi All, I was wondering if anyone would be kind enough to let me know if they have implemented anything like as follows in Kamailio? We are looking to implement call fowarding, activating it as usual in usr_preferences table. I was wondering if on the outbound leg its possible to write the username from usr_preferences with CFW active into the radacct table easily? This is with Radius and opensips DBs in Mysql. Many thanks Jon

2 2

NAT Traversal :: RTPProxy and Jitsi client
by Raj Roy Ghandhi 14 Dec '12

14 Dec '12

Hi, I am trying to communicate 2 Jitsi clients in 2 separate private networks. Both clients are behind NAT. Kamailio Server is on Public IP. Both Jitsi clients does register and presence works fine. And also text messages works well. But unable to call each other. Had some one had the same issue before. I tried with many soft clients and all failed with same result. Same scenario works with IP Phones. Can anybody point out what I miss there. Regards, Roy.

2 2

OT: MSRP over WebSocket Javascript stack
by Peter Dunkley 14 Dec '12

14 Dec '12

Hi, Crocodile has just open-sourced our MSRP over WebSocket (see http://tools.ietf.org/html/draft-pd-msrp-websocket) Javascript stack. The project is hosted on Google Code: http://code.google.com/p/crocodile-msrp/ The stack is distributed using the MIT License and was developed and tested along side the Kamailio MSRP over WebSocket implementation. Regards, Peter -- Peter Dunkley Technical Director Crocodile RCS Ltd

2 2

RTP Proxy Interaction
by Austin Einter 14 Dec '12

14 Dec '12

Dear All I am looking at Kamailio's module, file, function that receives rtp and rtcp port from rtp proxy. Can somebody please let me know which file/function I need to check. Thanks Austin

2 1

Re: [SR-Users] Help
by Peter Dunkley 14 Dec '12

14 Dec '12

Hi Roy, Please keep emails on list. Regards, Peter On Fri, 2012-12-14 at 09:31 -0800, Raj Roy Ghandhi wrote: > Dear Peter, > > Thanks for all the support that you gave me during the Kamalio > configuration with RTPProxy. > Currently it works fine with IP Phones but not soft clients. (Jitsi) > > > If I give you my server information would you be able to check that > out for me. > > > I was lost for that issue. > > > Best Regards, > Roy. -- Peter Dunkley Technical Director Crocodile RCS Ltd

1 0

Local call routing accross many kamailio servers
by Ali Jawad 14 Dec '12

14 Dec '12

Hi I do have 3 kamailio servers sharing the same db, calls go to Asterisk and that works great, I did purchase a DID and point it to one of the kamailio servers, it works fine so far for that kamailio server as the call is forwarded to the softphone ..but ONLY if the user is registered on the kamailio server the DID is forwarded to..if the kamailio user is on another kamailio server it does not work..although the database and location tables are shared. Simple diagram DID Provider <=== DID Kamailio ===> Kamailio Servers A, B, C If a softphone registers on DID Kamailio and I call the DID it works, if a softphone registers on Kamailio A or B or C, I never get the call, looking at ngrep the call just stays on DID Kamailio and it does not try to send it to any other server, how can I make the Kamailio DID lookup the destination from the location table and send the call there ? Regards

2 9

Hardware requirements voip servers
by Grant Bagdasarian 14 Dec '12

14 Dec '12

Hello, We are considering upgrading our VoIP platform and I'm wondering what kind of hardware we need for our requirements. The architecture will consist of a SIP Proxy Server (Kamailio) in front and a cluster of Media/Application Servers behind it. The requirements are based on a single server. SIP Proxy Server: - Up to 100 - 300 calls per second. - Up to 5000 - 10000 concurrent calls. - RTP Proxy - Record-Route Media/Application Server: - Up to 100 - 300 calls per second. - Up to 5000 - 10000 concurrent calls. What do we need in terms of CPU, Memory, HDD when looking at the following specific requirements: - 100 CPS - 200 CPS - 300 CPS Which hardware components are critical for handling this many calls, is it CPU, Memory, Disk IO bound? Also which open source media/application server software is suitable for these requirements? We are currently using Asterisk, which works great, but perhaps there are others available which are more suitable for these requirements. Thanks in advance! Regards, Grant Bagdasarian

1 0

Kamailio <-> Asterisk MWI
by Daniel Tryba 13 Dec '12

13 Dec '12

I'm trying to get the MWI indicator on endpoints to work by redirecting the SUBSCRIBE to the voicemail server (asterisk) using usc_replace_(to|from). This works fine for the initial request but the next subscribe will fail since the to/from don't get rewritten, which is indicated by the error: ERROR: uac [replace.c:250]: decline FROM replacing in sequential request in auto mode (has TO tag) This occurs when the modparam "uac","restore_mode" is manual or auto. When set to none the redirect will be correct but all responses have to be rewritten manually (I'm a beginner and I don't want to do this just yet). And since uac has knowledge about this having being rewritten, sending the original from/to looks like like a bug to me (more probable a misconfiguration/omission of something crucial withing dialog responses :) Changes to the (default debian package) config, in route route[PRESENCE] in if(is_method("SUBSCRIBE")) if( is_method("SUBSCRIBE")) { if(search_hf("Event", "message-summary", "a")) { if (!www_authenticate("$fd", "subscriber")) { www_challenge("$fd", "1"); exit; } if(avp_db_query("SELECT value FROM usr_preferences WHERE username='$au' and attribute='voicemail' ORDER BY value limit 1")) { $avp(voicemail)=$avp(i:1); $avp(subscrfrom)=$fu; avp_subst("$avp(subscrfrom)", "/sip:.*@/sip:$avp(voicemail)@/"); uac_replace_from("$avp(subscrfrom)"); uac_replace_to("","sip:$avp(voicemail)@$sel(cfg_get.voicemail.srv_ip): $sel(cfg_get.voicemail.srv_port)"); remove_hf("Authorization"); $ru = "sip:" + $avp(voicemail) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); route(RELAY); } } else { handle_subscribe(); t_release(); } } exit; 192.168.34.226 is the sip device, 192.168.32.40 kamailio, 192.168.32.41 the voicemailserver. Initial request: U 192.168.34.226:5852 -> 192.168.32.40:5060 SUBSCRIBE sip:grandstream@sip.local SIP/2.0. From: "Daniel" <sip:grandstream@sip.local>;tag=67d8f67065827fe2. To: <sip:grandstream@sip.local>. Gets rewritten to: U 192.168.32.40:5060 -> 192.168.32.41:5060 SUBSCRIBE sip:0880100782@192.168.32.41:5060 SIP/2.0. From: "Daniel" <sip:0880100782@sip.local>;tag=67d8f67065827fe2. To: <sip:0880100782@192.168.32.41:5060>. The response gets relayed to the endpoint and MWI will light up since there is a message. But next subscribe update from the endpoint triggers the "decline FROM replacing in sequential request" error and thus the subscribe send to voicemail is the original to/from: U 192.168.32.40:5060 -> 192.168.32.41:5060 SUBSCRIBE sip:0880100782@192.168.32.41:5060 SIP/2.0. From: "Daniel" <sip:grandstream@sip.local>;tag=67d8f67065827fe2. To: <sip:grandstream@sip.local>;tag=as324de8f3. Resulting in a 404 since the to/from are not the real peername on the voicemail server. How do I force Kamailio to rewrite the from/to without doing this manually for a requests/responses (restore_mode none)? The easy way out of this problem would be to have a 1 to 1 relation for the Kamailio subscribers and voicemail peers. But uac_replace should work (somehow). Attached is a full capture of the subscribe/notify. Running 3.3.2+squeeze from http://deb.kamailio.org/kamailio/. Machine is also running topoh module (192.168.0.1). -- POCOS B.V. - Croy 9c - 5653 LC Eindhoven Telefoon: 040 293 8661 - Fax: 040 293 8658 http://www.pocos.nl/ - http://www.sipo.nl/ K.v.K. Eindhoven 17097024

2 6

NAT for Load Balancer and Proxies
by Nguyen Anh Tuan 13 Dec '12

13 Dec '12

Hi everyone, I have 1 Load Balancer and 2 SIP Servers, all of them are Kamailio. Does anyone suggest how to add NAT for them? Best Regard, Nguyen Anh Tuan

2 1

SIP proxy changing Contact header of presence server requests
by Owen Lynch 13 Dec '12

13 Dec '12

Hi, we have 2 separate instances of kamailio running as a SIP proxy and presence server. The proxy uses the nathelper module to correctly change the Contact headers of client requests. However it also does the same for notify requests from the presence server, which results in incorrect RURI from subsequent client subscribes. The NATDETECT route is shown below, I plan to leave the existing nat_uac_test but also explicitly skip the fix_nated...() calls for requests from the presence server. Does this seem reasonable? # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } Thanks, Owen Lynch

2 1

Dispatching to Asterisk cluster
by Grant Bagdasarian 12 Dec '12

12 Dec '12

Hello, We have a cluster of three Asterisk machines. Each machine answers an incoming call and transfers it to unique remote destination. So, Asterisk01 transfers to Destination01, Asterisk02 to Destination02, Asterisk03 to Destination03. The Asterisk machines are allowed to receive and answer only 50 calls each. I have a Kamailio in front of these three machine. It is currently used to dispatch calls to the three machines using round-robin. But I need some way to equally dispatch a maximum of 50 calls to each device in the Asterisk dispatcher set. I was thinking of using the dispatcher module and using the 10 algorithm, to load balance based on call load. But I'm not sure if that will guarantee a maximum of 50 calls being dispatched to each Asterisk. Theoretically, if I set the ds_hash_size to 7, which is 128 slots, at some point two of them should have 43 calls and one 42 calls. Also the value of ds_hash_size should be the power of two. Are decimal values allowed, like 7,229 which equals to almost 150? Perhaps I should use the ratelimit module for this? I'd appreciate it if someone could give me some advice on this. Regards, Grant

2 2

Syntax of xavp
by Andreas Granig 12 Dec '12

12 Dec '12

Hi, I'm playing around with xavp, but there are some things I can't wrap my head around. What basically works is this: $xavp(a=>foo) = 'foo'; $xavp(a[0]=>bar) = 'bar'; Getting the value like this: xlog("L_INFO", "a-foo='$xavp(a=>foo)' and a-bar='$xavp(a=>bar)'"); as well as xlog("L_INFO", "a-foo='$xavp(a[0]=>foo)' and a-bar='$xavp(a[0]=>bar)'"); ... works, because when omitting the index, it assumes [0]. I really like to avoid such implicit assumptions though, to make the config really clear, so I prefer the second approach. What doesn't work for me then is explicit assignment of the very first value, like this: $xavp(a[0]=>foo) = 'foo'; $xavp(a[0]=>bar) = 'bar'; If I do this, the "a" xavp is never created. Wouldn't it be good practice to create the xavp on first assignment? That way, I don't have to take care in the config file whether the xavp is used for the first time, where I have to omit the index. What I'm also wondering is whether it's possible to directly access nested xavps, like this: $xavp(a=>foo) = 'afoo'; $xavp(b=>foo) = 'bfoo'; $xavp(c=>a) = $xavp(a); $xavp(c=>b) = $xavp(b); xlog("L_INFO", "a-foo='$xavp(c=>a[0]=>foo)'"); or xlog("L_INFO", "a-foo='$xavp(c=>a=>foo)'"); Both ways give me "a-foo='<<xavp:0x7f0d387fe178>>'". Is this even intended? Would be really cool if that's possible! Thanks, Andreas

3 4

NAPTR, SRV and sips vs. transport=tls
by Andreas Granig 11 Dec '12

11 Dec '12

Hi, Hope to get some guidance here over the usage of "sips" and "sip plus transport=tls" when following RFC3263. The RFC3263 says that a NATPR record could return something like this for a query like "host -t NAPTR example.com": ; order pref flags service regexp replacement IN NAPTR 50 50 "s" "SIPS+D2T" "" _sips._tcp.example.com. IN NAPTR 90 50 "s" "SIP+D2T" "" _sip._tcp.example.com IN NAPTR 100 50 "s" "SIP+D2U" "" _sip._udp.example.com. This means that the client should use sips if possible when contacting example.com. Concluding from that, I suppose it should perform an SRV lookup "host -t SRV _sips._tcp.example.com", and the result would be: ;; Priority Weight Port Target IN SRV 0 1 5061 server1.example.com IN SRV 0 2 5061 server2.example.com What I'm curious about is how requests towards one of these servers should look like, and how they are being handled by kamailio. A lot of clients and servers are sending "sip:user@example.com;transport=tls" in request URIs and Contact headers and Record-Route headers, and you can check with uri_param("transport","tls") which transport socket to use. This is pretty useful as you can determine hop-by-hop whether or not to use TLS. This approach has been obsoleted by RFC3261 though, and there doesn't seem to be a mechanism in RFC3263 to indicate "use schema sip, but use transport=tls". On encounter of a NAPTR record like the one above, how does kamailio act? Does it set a "sips" schema for the next hop? And what's the general take on this "sips" schema? As far as I understand RFC3261, it means that if a client sends a request to a sips-URI, the request is sent to the domain via TLS, and from there "the request is sent securely to the callee, but with security mechanisms that depend on the policy of the domain of the callee." (RFC3261, Chapter 4). What does this really mean in practice? Are you allowed to rewrite the schema to "sip" and pass it on for example via UDP to the callee if the callee didn't indicate transport=tls (deprecated anyways) or "sips:" in the Contact of the registration? Or should you keep "sips" as schema, but still send it via UDP, because you know based on local policy or based on client registration that the next hop is not supporting TLS? How would widespread clients react when getting a call to a "sips" URI, especially if they receive it via UDP? Looking forward hearing your input on that, Andreas

5 14

MySql error
by Hermes F. 11 Dec '12

11 Dec '12

I'm trying to create MySql database using the following /usr/local/sbin/kamdbctl create I followed the instructions on http://kb.asipto.com/kamailio:install:3.3.x-from-git-centos5x but I receive the following error message: "ERROR: database engine not specified, please setup one in the config script" Can someone help me please ?

2 1

Regarding Kamailio Registrar
by AMIT SHAH 11 Dec '12

11 Dec '12

Dear all, We are using Kamailio SIP Server in our application. We want to know 1 thing: We are using Kamailio SIP server as registrar and proxy server. For registration database, we are using dbtext module. We maintain record of SIP Extensions who have registered in Our proprietary file too. Now when we boot the Kamailio SIP server, At that time our file should be empty i.e. previously registered Extension entries should be deleted or it must be kept as it is. But if during power off, any extension had sent REGISTER request to Unregister then it was not executed as kamailio SIP server was not ON. But at boot time, as its entry was present, Kamailio SIP server will Consider it a registered extension. Please guide us for this situation. Thank you, Amit Shah.

2 1

Unable to set custom RPID
by Mino Haluz 11 Dec '12

11 Dec '12

Hi, what things should I set in order to set my custom RPID. What I am doing: remove_hf("Remote-Party-ID"); append_rpid_hf("", ";party=calling;id-type=subscriber;screen=no"); but RPID is only removed, second command does not append any rpid. Thanks Mino

2 1

New developer: Konstantin Mosesov
by Daniel-Constantin Mierla 10 Dec '12

10 Dec '12

Hello, I want to announce that a new developer got GIT write access to repository: Konstantin Mosesov - he has contributed patches to app_python and joins the team to help maintaining and developing this module. His git commit id is: ez My warm welcome and looking forward to future work within the project! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

2 1

How to use LIMIT 1 in db_query api
by Krishna Kurapati 10 Dec '12

10 Dec '12

Hi, I need a way to limit the results to just one row. The only way I know I could do is to call db_query followed by fetch_result of 1 row. However this is not optimal as all the rows are fetched to dynamic memory with db_query. Is there an alternative approach? Thanks Krish Kura

2 1

Help
by ameneh pourmoghadas 07 Dec '12

07 Dec '12

Dear Admin, My name is Ameneh from India. I am new in SIP. I have installed asterisk 1.8 on my ubuntu system and could have registered the internal users in my LAN to talk over x-lite. Now my question is how can I make a call to a mobile in India or a PSTN number in another country that would require a public SIP I think. how can I make my SIP server as public? would you please help me? thanks and regards, Ameneh

7 6

CTL module and alternatives for remote execution mi commands.
by Pavel Zalunin 07 Dec '12

07 Dec '12

Hi. In our project we need to query current users, dialogs and ability to end dialog using some rpc. Requirements are - do this remotely. I already tried to use xmlrpc module. When we had 42 users online it returned 17kb broken xml file. Then I tried to use binrpc protocol from CTL module. Usage ends with returning broken message too (in case, when message is biger than 21kb). Are there any other more stable solutions to query mi information using some tcp protocol? P.S. When I discovered sources of CTL module I found usage of writev - here is the problem about partial packets, when buffer is longer, than 21kb it don't pushes all vector buffers into socket. And in general, this module is using async sockets, AFAIK writev don't gives any guarantees, that it will write all vectors in async sockets. P.P.S. Of couse, last resort solution will be writing proxy for using mi_fifo on kamailio side to forward all records to other servers, but if there are more pretty solutions, please give me know:) Thanks. Pavel.

2 2

OFFTOPIC - Realtime testers needed - Asterisk Path header support (Oolong branch)
by Olle E. Johansson 07 Dec '12

07 Dec '12

While this is not Kamailio - it's related. If you want to use Kamailio in front of Asterisk, you do need SIP Path header support in Asterisk to do it right. Especially if you have one Kamailio handling Internet-facing communication and one handling internal communication. Please test! /O Vidarebefordrat brev: > Från: "Olle E. Johansson" <oej(a)edvina.net> > Ämne: Realtime testers needed - Path header support (Oolong branch) > Datum: 7 december 2012 11:07:09 CET > Till: Asterisk Developers Mailing List <asterisk-dev(a)lists.digium.com> > Kopia: "Olle E. Johansson" <oej(a)edvina.net> > > Friends, > > I have taken Klaus Darillion's patch for Path header support in Asterisk and created branches for 1.4, 1.8 and trunk. I need help testing it with realtime support by someone who are more used to the realtime databases than I am. I also need help completing the LDAP schemas with the new attribute. > > General testing is also encouraged. > > I have been using this patch without realtime for a couple of months. So far no Squirrels where hurt in the progress. And no bugs hit me in the back. > > Read more about the patch here: > http://svnview.digium.com/svn/asterisk/team/oej/oolong-path-support-trunk/R… > > Klaus' original review: > https://reviewboard.asterisk.org/r/991/ > > The SIP Path header support is documented in RFC 3327 (that has a strange title). > > Basically, this is used when you have a proxy between a phone and your Asterisk. Asterisk needs to remember the path to the device when sending an INVITE, so we save both the contact and the path (basically a route) to the phone when Asterisk receives a REGISTER. When sending an INVITE (or anything based on the registry) a route header is added and the message finds the proper way to the device. > > Path header support exists in the path module in Kamailio if you want to test. > http://kamailio.org/docs/modules/3.3.x/modules_k/path.html > > Looking forward to your feedback! > > /O

1 0

Kamailio\ Message 300 Contacts
by Fatima Chahrour~Vanrise Support 06 Dec '12

06 Dec '12

Dears, In the redirect message 300, is there any way to control the maximum number of contacts(IP's) returned in the message? (am only getting 5 IP's) Also, if I have one gateway with 5 IP's and trying the same call 3 times; the order of the IP's returned in the message 300 is different in each time. Why is that? Thanks in advance, F Chahrour

2 2

Kamailio 3.X.X + BLF configuration trouble
by Hemanshu Patel 06 Dec '12

06 Dec '12

Hello everyone, Till now we were working with asterisk to handle BLF. But then we decided to move everything to kamailio and hence i started working on BLF with kamailio. I have already configured kamailio for Call handling and presence information. >From different pages on Internet, I have found some information about configuration of kamailio+blf. And i did so, but somehow things are not working as they should. I have done following changes to kamailio.cfg file loadmodule "presence.so" loadmodule "presence_xml.so" loadmodule "presence_dialoginfo.so" loadmodule "presence_mwi.so" loadmodule "dialog.so" loadmodule "pua.so" loadmodule "pua_dialoginfo.so" modparam("presence", "fallback2db", 1) modparam("dialog", "dlg_flag", 4) modparam("dialog", "db_url", "mysql://xxxxxxxxx:xxxxxxxxxxxxxx@localhost/openser") modparam("dialog", "db_mode", 1) modparam("pua", "db_url", "mysql://xxxxxxxxx:xxxxxxxxxxxxxx@localhost/openser") Now when my phone(yealink VP530), sends subscribe request it looks as below: SUBSCRIBE sip:1001@172.16.27.66 SIP/2.0 Via: SIP/2.0/UDP 172.16.27.61:5063;branch=z9hG4bK10464096 From: "1000" <sip:1000@172.16.27.66>;tag=393923564 To: <sip:1001@172.16.27.66> Call-ID: 1572901975(a)172.16.27.61 CSeq: 1 SUBSCRIBE Contact: <sip:1000@172.16.27.61:5063> Accept: application/dialog-info+xml Max-Forwards: 70 User-Agent: VP530P 23.70.0.40 Expires: 120 Event: dialog Content-Length: 0 In response to this message, Kamailio returns 202 with following information SIP/2.0 202 OK Via: SIP/2.0/UDP 172.16.27.61:5063;branch=z9hG4bK245273058 From: "1000" <sip:1000@172.16.27.66>;tag=393923564 To: <sip:1001@172.16.27.66>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-8ee2 Call-ID: 1572901975(a)172.16.27.61 CSeq: 2 SUBSCRIBE Expires: 120 Contact: <sip:172.16.27.66:5060> Server: kamailio (3.0.0 (i386/linux)) Content-Length: 0 Followed by a NOTIFY NOTIFY sip:1000@172.16.27.61:5063 SIP/2.0 Via: SIP/2.0/UDP 172.16.27.66;branch=z9hG4bK3ccd.adfbc416.0 To: sip:1000@172.16.27.66;tag=2058189864 From: sip:1001@172.16.27.66;tag=a6a1c5f60faecf035a1ae5b6e96e979a-6cd8 CSeq: 1 NOTIFY Call-ID: 2040123403(a)172.16.27.61 Content-Length: 0 User-Agent: kamailio (3.0.0 (i386/linux)) Max-Forwards: 70 Event: dialog Contact: <sip:172.16.27.66:5060> Subscription-State: active;expires=170 During subscribe message asked for XML based reply, yet NOTIFY doesnt send any XML. and hence there is no impact of this message on device. Nothing is changed.Also there's nothing when i call from this device to another, ideally i should get some NOTIFY messages So, i think i am doing something wrong in Configuration. Can anyone please help me in it. If you need any other info, i'll provide you. -- Regards, Hemanshu Patel Senior Software Engg P Help the environment – please don't print this email unless you really need to!

3 7

Internal/External SIP setup
by Zero Aggression 06 Dec '12

06 Dec '12

Hello, I am a bit new to VoIP, and I just need a quick yes/no (and maybe pointers) answer. Is the following possible (and smart) to do with Kamailio? Kamailio server. Dual-homed (internal network, Internet). Internal users connect to Kamailio to make calls to internal and Internet SIP peers. Users have NO internet connection themselves (they can only reach the server). Can Kamaillio (when dual-homed) transport all the data/voice between Internet and intranet? Can it also be used to carry all traffic between intranet users (if there is no direct route between them)? Thanks so much. Zane

3 3

db_cluster configuration
by Juan José Ivars 06 Dec '12

06 Dec '12

Hello, i am configuring db_cluster module in kamailio 3.3 I have configured in roun-robin read between two MySQL servers but the load is not distributed equally My configuration is that: modparam("db_cluster", "connection", "con1=>mysql://USER1:PASS1@1.1.1.1/DB") modparam("db_cluster", "connection", "con2=>mysql://USER2:PASS2@2.2.2.2/DB") modparam("db_cluster", "cluster", "cls1=>con1=9r9p;con2=9r9p") And the auth module with: modparam("auth_db", "db_url", "cluster://cls1") But i have enabled the mysql logs and the major part of the querys are targeted to the second server Is there any cache by peer? What am i doing wrong? Thanks in advance.

2 1

Preparing for the next major release
by Olle E. Johansson 06 Dec '12

06 Dec '12

Hi! Daniel has set a code freeze date for the next major release and the Kamailio community needs to work together to prepare the release. That means you as well! There will be a lot of work in the coming weeks to merge code from SIP router into the Kamailio core to complete the merger process. We also have a large set of new modules and functions to test. The creation of a new release is much more than code - it's testing, documentation proof-reading and feedback. With a large and complex piece of software like Kamailio we can't test all functions with all possible combinations of configurations and connections to third party products - this is where YOU can help out! Read more on http://www.kamailio.org/w/2012/12/preparations-for-new-release/ and join us in creating the new release! /O

1 0

Kamailio Load Balancing, problems ACK and BYE.
by Nguyen Anh Tuan 05 Dec '12

05 Dec '12

Hi everyone, I am a beginner in using Kamailio. I am currently building a load balancer for multiple SIP Servers, all of them are Kamailio. The Load balancer can help end-users REGISTER and INVITE normally, but has problems with ACK and BYE. Messages 180 Ringing and 200 OK sent successfully to the Caller, but the ACK failed, is transmitted to the SIP Server then stop here, doesn't go back to Load balancing. (Caller's BYE -> LB -> Proxy then stop here). Because doesn't receive ACK, the Called continuously sent 200 call OK to Caller (sent successfully) and wait for confirmation ACK (after arrival to the SIP Server, ACK doesn't come back to Load Balancing). Because the Called party doesn't receive ACK, the call will be automatically disconnected after 30 seconds. With BYE have rather, if the Caller hangs up first, BYE is transferred over SIP Server and returned to Load balancer, but Load Balancing can't sent it to the Called (Caller's BYE -> LB -> Proxy -> LB then do not go to the side of called). Caller had picked off the phone, but the Caller is not received BYE and after 30 seconds the call will automatically disconnect (by ACK errors). If the called party hangs up first, HUNG UP button of the called even useless, and I don't see any packet of the Called party's BYE message in Wireshark. I use X-Lite and EyeBeam for two end users, the configuration of the SIP Server is the default configuration of Kamailio. DISPATHCHER.LIST: I have 3 Kamailio SIP Proxies, all of them OK, share one DB located in Proxy 2. But for capturing packets more convenient, I turn off the Proxy 2 and 3. # group sip addresses of your * units # Kamailio SIP Proxies/Registrars 1 sip:192.168.3.11:5060 #1 sip:192.168.3.12:5060 #1 sip:192.168.3.13:5060 Previously I used the Stateless Load Balancing configuration by default, but could not INVITE: # -- dispatcher params -- modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list") modparam("dispatcher", "force_dst", 1) route{ if ( !mf_process_maxfwd_header("10") ) { sl_send_reply("483","To Many Hops"); drop(); }; ds_select_dst("1", "0"); record_route(); forward(); # t_relay(); } Then I changed the following, sent INVITE and setup call successfully, but met ACK and BYE error above. if ( is_method("INVITE")) { # Not Dispatch INVITE from SIP Proxies if ( ds_is_from_list("1") ) { t_relay(); } else if ( !ds_is_from_list("1") ) { route(DISPATCH); } } else { # If not INVITE route(DISPATCH); } } route[DISPATCH] { # dispatch destinations ds_select_dst("1", "0"); record_route(); t_relay(); #forward(); } Please tell me How to modify in the configuration of Proxy and Load Balancing? Thank so much for helping! Best Regard Nguyen Anh Tuan, -------------------------------------------------------------------- The latest 200 OK from Called after Ringing : LB->Caller Via: SIP/2.0/UDP 172.24.104.159:9040;received=192.168.3.88;branch=z9hG4bK-d87543-a7349502ae55 cf46-1--d87543-;rport=9040 Record-Route: <sip:192.168.3.11;lr;nat=yes> Record-Route: <sip:192.168.3.10;lr=on> Contact: <sip:1004@192.168.3.10:5060;rinstance=040c24111dc72132> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 1102q stamp 51814 Content-Length: 435 Caller's ACK (1001(a)192.168.3.88) to Load Balancer (192.168.3.10): Via: SIP/2.0/UDP 172.24.104.159:9040;branch=z9hG4bK-d87543-6f1c8659392a0773-1--d87543-;rport Max-Forwards: 70 Route: <sip:192.168.3.10;lr> Route: <sip:192.168.3.11;lr;nat=yes> Contact: <sip:1001@192.168.3.88:9040> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 2 ACK Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@registrar.sip.vn",response="ec2a4c8832bc7bf7665811c26ec9 b292",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Content-Length: 0 ACK 192.168.3.10:5060 (Load Balance) to 192.168.3.11:5060 (Proxy) then STOP HERE (Not transfer further) Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKcydzigwkX Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKcydzigwkX Via: SIP/2.0/UDP 172.24.104.159:9040;received=192.168.3.88;branch=z9hG4bK-d87543-6f1c8659392a 0773-1--d87543-;rport=9040 Max-Forwards: 68 Contact: <sip:1001@192.168.3.88:9040> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 2 ACK Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@registrar.sip.vn",response="ec2a4c8832bc7bf7665811c26ec9 b292",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Content-Length: 0 Caller's BYE (1001(a)192.168.3.88) to Load Balance (192.168.3.10) Via: SIP/2.0/UDP 172.24.104.159:9040;branch=z9hG4bK-d87543-616e232f86191a37-1--d87543-;rport Max-Forwards: 70 Route: <sip:192.168.3.10;lr> Route: <sip:192.168.3.11;lr;nat=yes> Contact: <sip:1001@192.168.3.88:9040> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 3 BYE Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@192.168.3.10:5060;rinstance=040c24111dc72132",response=" c738898739dc49c02676e04a28761971",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Reason: SIP;description="User Hung Up" Content-Length: 0 BYE was transferred from Load Balance(192.168.3.10) to Proxy (192.168.3.11), repeated THREE TIMES, possibly due to an error loop (Route(DISPATCH)?) Record-Route: <sip:192.168.3.10;lr=on> Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKc5be.03f49894.1 Via: SIP/2.0/UDP 172.24.104.159:9040;received=192.168.3.88;branch=z9hG4bK-d87543-616e232f8619 1a37-1--d87543-;rport=9040 Max-Forwards: 69 Route: <sip:192.168.3.10;lr> Contact: <sip:1001@192.168.3.88:9040> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 3 BYE Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@192.168.3.10:5060;rinstance=040c24111dc72132",response=" c738898739dc49c02676e04a28761971",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Reason: SIP;description="User Hung Up" Content-Length: 0 Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKc5be.13f49894.0 Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKc5be.03f49894.0 Via: SIP/2.0/UDP 172.24.104.159:9040;received=192.168.3.88;branch=z9hG4bK-d87543-616e232f8619 1a37-1--d87543-;rport=9040 Max-Forwards: 68 Contact: <sip:1001@192.168.3.88:9040> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 3 BYE Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@192.168.3.10:5060;rinstance=040c24111dc72132",response=" c738898739dc49c02676e04a28761971",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Reason: SIP;description="User Hung Up" Content-Length: 0 Record-Route: <sip:192.168.3.10;lr=on> Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKc5be.13f49894.1 Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKc5be.03f49894.0 Via: SIP/2.0/UDP 172.24.104.159:9040;received=192.168.3.88;branch=z9hG4bK-d87543-616e232f8619 1a37-1--d87543-;rport=9040 Max-Forwards: 68 Contact: <sip:1001@192.168.3.88:9040> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 3 BYE Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@192.168.3.10:5060;rinstance=040c24111dc72132",response=" c738898739dc49c02676e04a28761971",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Reason: SIP;description="User Hung Up" Content-Length: 0 Send back BYE from SIP Proxy (192.168.3.11) to Load Balance (192.168.3.10) Record-Route: <sip:192.168.3.10;lr=on> Via: SIP/2.0/UDP 192.168.3.11;branch=z9hG4bKc5be.0070e643.0 Via: SIP/2.0/UDP 192.168.3.10;rport=5060;branch=z9hG4bKc5be.03f49894.1 Via: SIP/2.0/UDP 172.24.104.159:9040;received=192.168.3.88;branch=z9hG4bK-d87543-616e232f8619 1a37-1--d87543-;rport=9040 Max-Forwards: 68 Contact: <sip:1001@192.168.3.10:5060> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 3 BYE Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@192.168.3.10:5060;rinstance=040c24111dc72132",response=" c738898739dc49c02676e04a28761971",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Reason: SIP;description="User Hung Up" Content-Length: 0 BYE LOOPs Back from Load Balance(192.168.3.10) to Proxy (192.168.3.11), possibly due to an error loop (Route(DISPATCH)?) Record-Route: <sip:192.168.3.10;lr=on> Record-Route: <sip:192.168.3.10;lr=on> Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKc5be.33f49894.0 Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKc5be.23f49894.0 Via: SIP/2.0/UDP 192.168.3.11;branch=z9hG4bKc5be.0070e643.0 Via: SIP/2.0/UDP 192.168.3.10;rport=5060;branch=z9hG4bKc5be.03f49894.1 Via: SIP/2.0/UDP 172.24.104.159:9040;received=192.168.3.88;branch=z9hG4bK-d87543-616e232f8619 1a37-1--d87543-;rport=9040 Max-Forwards: 66 Contact: <sip:1001@192.168.3.10:5060> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 3 BYE Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@192.168.3.10:5060;rinstance=040c24111dc72132",response=" c738898739dc49c02676e04a28761971",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Reason: SIP;description="User Hung Up" Content-Length: 0

1 0

Kamailio Load Balancing, problems ACK and BYE.
by Nguyen Anh Tuan 05 Dec '12

05 Dec '12

Hi everyone, I am a beginner in using Kamailio. I am currently building a load balancer for multiple SIP Servers, all of them are Kamailio. The Load balancer can help end-users REGISTER and INVITE normally, but has problems with ACK and BYE. Messages 180 Ringing and 200 OK sent successfully to the Caller, but the ACK failed, is transmitted to the SIP Server then stop here, doesn't go back to Load balancing. (Caller's BYE -> LB -> Proxy then stop here). Because doesn't receive ACK, the Called continuously sent 200 call OK to Caller (sent successfully) and wait for confirmation ACK (after arrival to the SIP Server, ACK doesn't come back to Load Balancing). Because the Called party doesn't receive ACK, the call will be automatically disconnected after 30 seconds. With BYE have rather, if the Caller hangs up first, BYE is transferred over SIP Server and returned to Load balancer, but Load Balancing can't sent it to the Called (Caller's BYE -> LB -> Proxy -> LB then do not go to the side of called). Caller had picked off the phone, but the Caller is not received BYE and after 30 seconds the call will automatically disconnect (by ACK errors). If the called party hangs up first, HUNG UP button of the called even useless, and I don't see any packet of the Called party's BYE message in Wireshark. I use X-Lite and EyeBeam for two end users, the configuration of the SIP Server is the default configuration of Kamailio. DISPATHCHER.LIST: I have 3 Kamailio SIP Proxies, all of them OK, share one DB located in Proxy 2. But for capturing packets more convenient, I turn off the Proxy 2 and 3. # group sip addresses of your * units # Kamailio SIP Proxies/Registrars 1 sip:192.168.3.11:5060 #1 sip:192.168.3.12:5060 #1 sip:192.168.3.13:5060 Previously I used the Stateless Load Balancing configuration by default, but could not INVITE: # -- dispatcher params -- modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list") modparam("dispatcher", "force_dst", 1) route{ if ( !mf_process_maxfwd_header("10") ) { sl_send_reply("483","To Many Hops"); drop(); }; ds_select_dst("1", "0"); record_route(); forward(); # t_relay(); } Then I changed the following, sent INVITE and setup call successfully, but met ACK and BYE error above. if ( is_method("INVITE")) { # Not Dispatch INVITE from SIP Proxies if ( ds_is_from_list("1") ) { t_relay(); } else if ( !ds_is_from_list("1") ) { route(DISPATCH); } } else { # If not INVITE route(DISPATCH); } } route[DISPATCH] { # dispatch destinations ds_select_dst("1", "0"); record_route(); t_relay(); #forward(); } Please tell me How to modify in the configuration of Proxy and Load Balancing? Thank so much for helping! Best Regard Nguyen Anh Tuan, -------------------------------------------------------------------- The latest 200 OK from Called after Ringing : LB->Caller Via: SIP/2.0/UDP 172.24.104.159:9040;received=192.168.3.88;branch=z9hG4bK-d87543-a7349502ae55 cf46-1--d87543-;rport=9040 Record-Route: <sip:192.168.3.11;lr;nat=yes> Record-Route: <sip:192.168.3.10;lr=on> Contact: <sip:1004@192.168.3.10:5060;rinstance=040c24111dc72132> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 1102q stamp 51814 Content-Length: 435 Caller's ACK (1001(a)192.168.3.88) to Load Balancer (192.168.3.10): Via: SIP/2.0/UDP 172.24.104.159:9040;branch=z9hG4bK-d87543-6f1c8659392a0773-1--d87543-;rport Max-Forwards: 70 Route: <sip:192.168.3.10;lr> Route: <sip:192.168.3.11;lr;nat=yes> Contact: <sip:1001@192.168.3.88:9040> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 2 ACK Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@registrar.sip.vn",response="ec2a4c8832bc7bf7665811c26ec9 b292",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Content-Length: 0 ACK 192.168.3.10:5060 (Load Balance) to 192.168.3.11:5060 (Proxy) then STOP HERE (Not transfer further) Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKcydzigwkX Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKcydzigwkX Via: SIP/2.0/UDP 172.24.104.159:9040;received=192.168.3.88;branch=z9hG4bK-d87543-6f1c8659392a 0773-1--d87543-;rport=9040 Max-Forwards: 68 Contact: <sip:1001@192.168.3.88:9040> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 2 ACK Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@registrar.sip.vn",response="ec2a4c8832bc7bf7665811c26ec9 b292",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Content-Length: 0 Caller's BYE (1001(a)192.168.3.88) to Load Balance (192.168.3.10) Via: SIP/2.0/UDP 172.24.104.159:9040;branch=z9hG4bK-d87543-616e232f86191a37-1--d87543-;rport Max-Forwards: 70 Route: <sip:192.168.3.10;lr> Route: <sip:192.168.3.11;lr;nat=yes> Contact: <sip:1001@192.168.3.88:9040> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 3 BYE Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@192.168.3.10:5060;rinstance=040c24111dc72132",response=" c738898739dc49c02676e04a28761971",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Reason: SIP;description="User Hung Up" Content-Length: 0 BYE was transferred from Load Balance(192.168.3.10) to Proxy (192.168.3.11), repeated THREE TIMES, possibly due to an error loop (Route(DISPATCH)?) Record-Route: <sip:192.168.3.10;lr=on> Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKc5be.03f49894.1 Via: SIP/2.0/UDP 172.24.104.159:9040;received=192.168.3.88;branch=z9hG4bK-d87543-616e232f8619 1a37-1--d87543-;rport=9040 Max-Forwards: 69 Route: <sip:192.168.3.10;lr> Contact: <sip:1001@192.168.3.88:9040> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 3 BYE Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@192.168.3.10:5060;rinstance=040c24111dc72132",response=" c738898739dc49c02676e04a28761971",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Reason: SIP;description="User Hung Up" Content-Length: 0 Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKc5be.13f49894.0 Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKc5be.03f49894.0 Via: SIP/2.0/UDP 172.24.104.159:9040;received=192.168.3.88;branch=z9hG4bK-d87543-616e232f8619 1a37-1--d87543-;rport=9040 Max-Forwards: 68 Contact: <sip:1001@192.168.3.88:9040> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 3 BYE Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@192.168.3.10:5060;rinstance=040c24111dc72132",response=" c738898739dc49c02676e04a28761971",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Reason: SIP;description="User Hung Up" Content-Length: 0 Record-Route: <sip:192.168.3.10;lr=on> Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKc5be.13f49894.1 Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKc5be.03f49894.0 Via: SIP/2.0/UDP 172.24.104.159:9040;received=192.168.3.88;branch=z9hG4bK-d87543-616e232f8619 1a37-1--d87543-;rport=9040 Max-Forwards: 68 Contact: <sip:1001@192.168.3.88:9040> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 3 BYE Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@192.168.3.10:5060;rinstance=040c24111dc72132",response=" c738898739dc49c02676e04a28761971",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Reason: SIP;description="User Hung Up" Content-Length: 0 Send back BYE from SIP Proxy (192.168.3.11) to Load Balance (192.168.3.10) Record-Route: <sip:192.168.3.10;lr=on> Via: SIP/2.0/UDP 192.168.3.11;branch=z9hG4bKc5be.0070e643.0 Via: SIP/2.0/UDP 192.168.3.10;rport=5060;branch=z9hG4bKc5be.03f49894.1 Via: SIP/2.0/UDP 172.24.104.159:9040;received=192.168.3.88;branch=z9hG4bK-d87543-616e232f8619 1a37-1--d87543-;rport=9040 Max-Forwards: 68 Contact: <sip:1001@192.168.3.10:5060> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 3 BYE Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@192.168.3.10:5060;rinstance=040c24111dc72132",response=" c738898739dc49c02676e04a28761971",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Reason: SIP;description="User Hung Up" Content-Length: 0 BYE LOOPs Back from Load Balance(192.168.3.10) to Proxy (192.168.3.11), possibly due to an error loop (Route(DISPATCH)?) Record-Route: <sip:192.168.3.10;lr=on> Record-Route: <sip:192.168.3.10;lr=on> Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKc5be.33f49894.0 Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKc5be.23f49894.0 Via: SIP/2.0/UDP 192.168.3.11;branch=z9hG4bKc5be.0070e643.0 Via: SIP/2.0/UDP 192.168.3.10;rport=5060;branch=z9hG4bKc5be.03f49894.1 Via: SIP/2.0/UDP 172.24.104.159:9040;received=192.168.3.88;branch=z9hG4bK-d87543-616e232f8619 1a37-1--d87543-;rport=9040 Max-Forwards: 66 Contact: <sip:1001@192.168.3.10:5060> To: "1004"<sip:1004@registrar.sip.vn>;tag=8e782d60 From: "1001 registrar.sip.vn"<sip:1001@registrar.sip.vn>;tag=7c72f24e Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc. CSeq: 3 BYE Proxy-Authorization: Digest username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P 32lS",uri="sip:1004@192.168.3.10:5060;rinstance=040c24111dc72132",response=" c738898739dc49c02676e04a28761971",algorithm=MD5 User-Agent: eyeBeam release 1003s stamp 31159 Reason: SIP;description="User Hung Up" Content-Length: 0

1 0

Segmentation fault in sip-capture
by Owen Lynch 05 Dec '12

05 Dec '12

Hi, I am running Kamailio 3.3 as a Homer capture agent, the sip-capture module receives HEP packets from a sip-trace module in a remote kamailio instance. Periodically the capture agent dies. The log and core dump suggest the initial problem is an invalid HEP msg received, but kamailio crashes when trying to send an error reply. Can anyone shed any light on this? Thanks, Owen Lynch Back trace: Program terminated with signal 11, Segmentation fault. #0 0x0813c916 in send2child (tcpconn=0xb55c1c60) at tcp_main.c:3967 3967 tcp_main.c: No such file or directory. in tcp_main.c Missing separate debuginfos, use: debuginfo-install glibc-2.12-1.80.el6.i686 keyutils-libs-1.4-4.el6.i686 krb5-libs-1.9-33.el6.i686 libcom_err-1.41.12-12.el6.i686 libselinux-2.0.94-5.3.el6.i686 mysql-libs-5.1.61-4.el6.i686 nss-softokn-freebl-3.12.9-11.el6.i686 openssl-1.0.0-20.el6_2.5.i686 zlib-1.2.3-27.el6.i686 (gdb) bt #0 0x0813c916 in send2child (tcpconn=0xb55c1c60) at tcp_main.c:3967 #1 0x08136e78 in handle_tcpconn_ev (fm=<value optimized out>, ev=<value optimized out>, idx=<value optimized out>) at tcp_main.c:4310 #2 handle_io (fm=<value optimized out>, ev=<value optimized out>, idx=<value optimized out>) at tcp_main.c:4362 #3 0x0813ea3c in io_wait_loop_epoll () at io_wait.h:1092 #4 tcp_main_loop () at tcp_main.c:4656 #5 0x080a502b in main_loop () at main.c:1727 #6 0x080a6c00 in main (argc=11, argv=0xbfeafd14) at main.c:2546 Log: Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: : <core> [pass_fd.c:288]: ERROR: receive_fd: recvmsg on 10 failed: Connection reset by peer Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: ERROR: <core> [tcp_main.c:2347]: BUG: tcp_send: failed to get fd(receive_fd): Connection reset by peer (104) Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: ERROR: sl [../../forward.h:193]: msg_send: ERROR: tcp_send failed Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=704 a=3 n=exit Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [xavp.c:365]: destroying xavp list (nil) Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [receive.c:293]: receive_msg: cleaning up Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: ERROR: sipcapture [sipcapture.c:682]: ERROR: sipcapture:hep_msg_received: not supported version or bad length: v:[10] l:[10] vs [40] Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [parser/msg_parser.c:634]: SIP Reply (status): Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [parser/msg_parser.c:636]: version: <SIP/2.0> Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [parser/msg_parser.c:638]: status: <500> Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [parser/msg_parser.c:640]: reason: <Invalid CSeq> Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [parser/parse_via.c:1286]: Found param type 234, <received> = <yadayadayada>; state=6 Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [parser/parse_via.c:1286]: Found param type 232, <branch> = <z9hG4bK2348.d1620f62.0>; state=16 Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [parser/parse_via.c:2561]: end of header reached, state=5 Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [parser/msg_parser.c:511]: parse_headers: Via found, flags=2 Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: <core> [parser/msg_parser.c:513]: parse_headers: this is the first via Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19567]: ALERT: <core> [main.c:785]: child process 19576 exited by a signal 11 Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19567]: ALERT: <core> [main.c:788]: core was generated Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19567]: INFO: <core> [main.c:800]: INFO: terminating due to SIGCHLD Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19575]: INFO: <core> [main.c:851]: INFO: signal 15 received Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19574]: INFO: <core> [main.c:851]: INFO: signal 15 received Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19573]: INFO: <core> [main.c:851]: INFO: signal 15 received Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19572]: INFO: <core> [main.c:851]: INFO: signal 15 received Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19567]: DEBUG: tm [t_funcs.c:122]: DEBUG: tm_shutdown : start Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19567]: DEBUG: tm [t_funcs.c:125]: DEBUG: tm_shutdown : emptying hash table Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19567]: DEBUG: tm [t_funcs.c:127]: DEBUG: tm_shutdown : removing semaphores Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19567]: DEBUG: tm [t_funcs.c:129]: DEBUG: tm_shutdown : destroying tmcb lists Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19567]: DEBUG: tm [t_funcs.c:132]: DEBUG: tm_shutdown : done Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19567]: DEBUG: tls [tls_init.c:674]: tls module final tls destroy Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19567]: DEBUG: <core> [mem/shm_mem.c:242]: shm_mem_destroy Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19567]: DEBUG: <core> [mem/shm_mem.c:245]: destroying the shared memory lock Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19567]: DEBUG: <core> [main.c:804]: terminating due to SIGCHLD

2 5

modules_s/eval - survey of usage
by Daniel-Constantin Mierla 05 Dec '12

05 Dec '12

Hello, is anyone using the modules_s/eval module? In the context of merging modules_s/* to modules/*, this one lacks the docbook xml files for readme. It has a text readme file, from where I understand it does script operations using a stack and polish notation expressions. Probably it was developed before the expressions support in config file language, at this moment all its operations can be done directly by the interpreter. I am considering moving the module to obsolete/ folder. If anyone is using it and wants to keep it in modules/*, then he/she has to convert the readme text into docbook format (which is not complex at all, just takes a bit of time, probably like 60 min or even less). If there is no answer in one week about usage, will be sent to obsolete/, from where it can be rescued in case of need (just to be clear for everybody that code is not lost). Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

dns resolver issue (RFC3263)
by MÉSZÁROS Mihály 05 Dec '12

05 Dec '12

Hi All, I am experiencing an issue when i try to contact xy(a)cisco.com. I found that kamailio/sip-router can't resolve by default resolving way a TCP + SRV records from domain cisco.com. e.g. cisco.com misi@alma:~$ host -t NAPTR cisco.com cisco.com has no NAPTR record misi@alma:~$ host -t SRV _sip._udp.cisco.com Host _sip._udp.cisco.com not found: 3(NXDOMAIN) misi@alma:~$ host -t SRV _sip._tcp.cisco.com _sip._tcp.cisco.com has SRV record 1 0 5060 vcsgw.cisco.com. misi@alma:~$ host -t SRV _sips._tcp.cisco.com _sips._tcp.cisco.com has SRV record 1 0 5061 vcsgw.cisco.com. I can't call xy(a)cisco.com because it does not exists an NAPTR record in domain cisco.com, and furthermore no SRV with udp. But it exists sip+tcp record and also exists an secure sip SRV, so sips+tcp record! I read rfc3263 I find kamailio dns resolver is not working as it should according RFC3263. http://tools.ietf.org/html/rfc3263 _If no NAPTR records are found, the client constructs SRV queries for those transport protocols it supports, and does a query for each._ So kamailio should query all (udp, tcp, tls, sctp whatever) protocols. Can anyone help/guide me to create a fix to this issue? My plan is to create a patch to kamailio resolver, to correct and behave according RFC3263. Any help or guidance appreciated! Thanks, Misi

4 14

syslog message about invalid domain name
by Juha Heinanen 05 Dec '12

05 Dec '12

when i t_relay in-dialog bye: Request-Line: BYE sip:0x8eb94d0@[2002:c062:660a::1]:5050;transport=tcp SIP/2.0 Message Header Via: SIP/2.0/TCP [2002:C062:660A:0:0:0:0:1];branch=z9hG4bKde53.6aaed77ea908953bbb133e5021d74036.0 Via: SIP/2.0/UDP [::1]:5090;received=0:0:0:0:0:0:0:1;branch=z9hG4bKq~ouJaGs;rport=5090 From: <sip:jh@as.test.fi>;tag=5A29C828-50BEFE55000C7118-B4FF9B70 To: "" <sip:test@test.fi>;tag=92e26ffbfbc16e53 CSeq: 10 BYE Call-ID: 28d1e4872cc5f83e User-Agent: Sip Express Media Server (1.6.0 (x86/linux)) Max-Forwards: 69 Content-Length: 0 with $du set to sip:[2002:C062:660A:0:0:0:0:1]:59933;transport=tcp, i get to syslog: Dec 5 09:53:48 sip /usr/sbin/sip-proxy[1947]: INFO: <core> [resolve.c:729]: invalid domain name '[2002:C062:660A:0:0:0:0:1]' since everything is and works as expected, why is the info message produced and how can i get rid of it? -- juha

2 2

Xcap server chooice among openSER, mobicents and openXCAP
by zhi sun 05 Dec '12

05 Dec '12

hi guys, it is my first time to use XCAP, i want to implement XCAP server to support maybe huge amount of user accesses (as cloud solution), so far, i can find out three XCAP server solution: 1. openSER with XCAP Server Module http://kamailio.org/docs/modules/devel/modules_k/xcap_server.html 2. mobicents with XCAP component/server build-in http://code.google.com/p/mobicents/wiki/Welcome?tm=6 http://www.mobicents.org/diameter/index.html http://hudson.jboss.org/hudson/view/Mobicents/job/Mobicents-Books/lastSucce… 3. openXCAP http://www.ag-projects.com/projects-products-96/101-openxcap but i am not clear to choose with solution to best match my situation (high volumn, high scale, and quick response). - does openSER support the cloud solution with high volum, high scale? - is it easy to use/integrate the XCAP server independently from mobicents? - is there performance issue for openXCAP, because it is implemented using python? any suggestions would be appreciated! best regards, -iamsy

1 0

io_wait errors
by Juha Heinanen 05 Dec '12

05 Dec '12

any idea what could be the cause for lots of these kind syslog messages: Dec 3 23:11:36 sip2 /usr/sbin/sip-proxy[5490]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 56 Dec 3 23:11:36 sip2 /usr/sbin/sip-proxy[5490]: ERROR: <core> [io_wait.h:628]: ERROR: io_watch_del: trying to delete already erased entry 56 in the hash(-1, 0, 0xb2f28adc) flags 0) Dec 3 23:11:36 sip2 /usr/sbin/sip-proxy[5490]: ERROR: <core> [io_wait.h:628]: ERROR: io_watch_del: trying to delete already erased entry 59 in the hash(-1, 0, 0xb725a9d8) flags 0) Dec 3 23:11:36 sip2 /usr/sbin/sip-proxy[5490]: : <core> [pass_fd.c:209]: ERROR: send_fd: sendmsg failed sending 58 on 59: Broken pipe (32) Dec 3 23:11:36 sip2 /usr/sbin/sip-proxy[5490]: ERROR: <core> [tcp_main.c:4003]: ERROR: send2child: send_fd failed for 0xb3024a48 (flags 0x4018), fd 58 -- juha

2 2

[Kamailio-Users] BYE message not relayed to caller
by Vikram Ragukumar 04 Dec '12

04 Dec '12

Hello, My setup includes a cellphone that uses Wi-Fi to hookup to the internet with IP 192.168.1.101. Cellphone app, tries to register with VOIPSwitch through an outbound proxy server(Kamailio + rtpproxy), which rewrites SDP to force rtp through an rtpproxy. I can make a call from the cellphone to a PSTN number. When the PSTN endpoint hangs up, VOIPSwitch relays a BYE message to the proxy that looks like : No. Time Source Destination Protocol Info 27 18.383929 IPaddr_VoipSwitch IPaddr_proxy SIP Request: BYE sip:1234@IPaddr_VoipSwitch Frame 27 (420 bytes on wire, 420 bytes captured) Ethernet II, Src: Unispher_40:b5:39 (00:90:1a:40:b5:39), Dst: Supermic_bd:b9:bc (00:30:48:bd:b9:bc) Internet Protocol, Src: IPaddr_VoipSwitch , Dst: IPaddr_proxy User Datagram Protocol, Src Port: sip (5060), Dst Port: 7160 (7160) Session Initiation Protocol Request-Line: BYE sip:1234@IPaddr_VoipSwitch SIP/2.0 Method: BYE [Resent Packet: False] Message Header Route: <sip:IPaddr_proxy:7160;lr=on;nat=yes> CSeq: 1 BYE Via: SIP/2.0/UDP IPaddr_VoipSwitch:5060; branch=z9hG4bk180242100434182912333954 From: sip:5678@IPaddr_VoipSwitch;tag=18024210041929123187506289 Call-ID: LUOyIDm-ecy36NcUfj5G3v1lM6g5snkM To: "1234" <sip:1234@IPaddr_VoipSwitch>;tag=9XHJmiuf58vTrF7.vwflEK-G63JVPQSi Content-Length: 0 The request URI in the above message is the IP address of VOIPSwitch itself, which causes the proxy to forward the BYE message back to VOIPSwitch. The cellphone never receives the BYE message and consequently never hangs up. Any suggestions on how i can get around this problem ? Thanks and Regards, Vikram.

3 9

openser_receiver
by Uri Shacked 04 Dec '12

04 Dec '12

Hi, I notice that under my /tmp folder there are openser_receiver filles as well (kamailio_fifo and kamailio_ctl are know....) Version is 3.3.2 latest branch. Any ideas? Thanks, Uri

1 0

Problem with Stateless Dispatcher Call Load Distribution (Algorithm 10)
by SIP Mailing-list 04 Dec '12

04 Dec '12

Hi guys, I wonder if someone could sanity check something for me. I currently have a simple load balancer set up using algorithm 10 (call load distribution) from the dispatcher module. I'm using Record-Routing so that I can clear down the call load record at the end of the call. Everything looked like it was working okay until I just spotted a problem that I think might be a bug. It's to do with ACKs to non-200 responses. In a normal call, the flow goes: UAC Proxy UAS 1 INVITE --> 2 INVITE--> 3 <-- 200 OK --- ds_load_update() 4 <-- 200 OK 5 ACK --> 6 ACK --> Now between step 3 and 4 the proxy runs ds_load_update which according to the documentation "set internal state to confirmed for the call load" entry in the internal call state store. This means the proxy knows where to send the ACK at step 5 because it is safely in the call state store so you can use ds_select_dst again on the ACK to get it to the right place. Works fine! Now consider the non-200 case: UAC Proxy UAS 1 INVITE --> 2 INVITE--> 3 <-- 603 Decline --- ds_load_unset() 4 <-- 603 Decline 5 ACK --> 6 ACK --> ??? Between step 3 and 4 the proxy is supposed to run ds_load_unset() which unconditionally removes the call from the call state store. Now when the ACK comes in at step 5, the proxy has no record of the call and so doesn't know where to send the ACK. This results in retried Declines and ACKs. Broken :-( If the proxy doesn't run ds_load_unset() on the 603 response, is there some kind of timer that will cause the call to be removed from the call state store on unconfirmed calls? I don't think I can run ds_load_unset on the ACK because there's nothing in the ACK that tells me it is because of a 603, rather than a 200. Edit: actually there is! The Route header is present on the 200 ACK, but * not* on the 3xx to 6xx ACK. Maybe I can use loose_route... Anyway, any thoughts would be appreciated! Cheers, Richard

2 4

default flavour set to kamailio
by Daniel-Constantin Mierla 04 Dec '12

04 Dec '12

Hello, based on community interaction, development, package distributions and documentation maintenance in the recent years, the default flavour has been switched now to 'kamailio'. As written in my previous email, we started the last phase to complete the integration between kamailio-ser, aiming to get to a unique database schema and set of modules in maximum 6 months. Kamailio is the most spread and popular flavour, presented in many linux/unix distributions, being the one released and packaged periodically. So it will stay as main flavour, the one officially released by the project. Those that want can take the sources and build their own flavour, as the support in Makefile system is kept. Once the integration is finished, the flavour system will just set the application name, nothing else. From the open source project perspective, the resources will be allocated only on releasing under the main/default flavour (Kamailio). Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Last phase of integration for kamailio - ser
by Daniel-Constantin Mierla 04 Dec '12

04 Dec '12

Hello, we plan to start the last phase of integration between kamailio and ser. After about 4 years with same source code tree, there is still some overhead caused by several duplicated modules. Following the decision to go on with the most developed components and absorb the missing features from the duplicates, kamailio modules are those that will stay (move to modules/ eventually) and get updates when needed. We used same decision to keep the core and makefile system from ser, adding to it what was extra in kamailio. I present next a summary of what is planned to happen: 1) Modules from modules_s/ that will be simply moved to modules/ because there is no conflict at all: - avp, avp_db, db_ops, eval, mangler, print, print_lib 2) Modules from modules_s/ that will be renamed and moved modules/ because it is mainly only conflict of name, but not in the backend (e.g., database): - auth_db => auth_dbattrs - because it uses (attribute,value) style to store subscriber profiles, which can be a preferred alternative for many out there - ldap => db2_ldap - it is a DB APIv2 connector to ldap server - bdb => db2_bdb - it is a DB APIv2 connector to berkeley db (to be double-checked) - xlog => xprintf - it uses %spec to print log messages with all the code inside the module - the only reason not discarding it and stick only to modules_k/xlog for the moment is because it is used from other modules, otherwise modules_k/xlog has more features (e.g., syslog facility, color printing, a.s.o) 3) Modules from modules_s/ that will be removed, after absorbing some extra features: - usrloc/registrar => get the ability to store AVPs per contact (modules_k is more advanced in features, like path/gruu/partial outbound support, publish presence states, etc.) - several modules must be reviewed: acc*, radius*, permissions, pike, gflags, options, timer 4) Modules from modules_s/ that will be removed, by moving to obsolete/ directory - the rest of them from modules_s/, like dialog, xcap, cpl-c, presence_b2b, dispatcher, dbtext, domain, etc. The categories and their content are not nailed down, it is the purpose of this discussion to see if something was overlooked. We hope to end this process in max 6 months time - after that will be a unique database schema and a unique set of modules. Flavour system in the Makefile will stay just for letting people choose easier the name of binary, otherwise everything will be the same (e.g., what is compiled, installed, etc). Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Kamailio v3.3.2 Released
by Daniel-Constantin Mierla 04 Dec '12

04 Dec '12

Hello, Kamailio SIP Server v3.3.2 stable release is out. This is a maintenance release of the latest stable branch, 3.3, that includes fixes since release of v3.3.1. There is no change to database schema or configuration language structure that you have to do on installations of v3.3.0 or v3.3.1. Deployments running previous v3.x.x versions are strongly recommended to be upgraded to v3.3.2. For more details about version 3.3.2 (including links and hints to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2012/10/kamailio-v3-3-2-released/ RPM, Debian/Ubuntu packages will be available soon as well. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

4 4

sample SCA kamailio.cfg added
by Andrew Mortensen 04 Dec '12

04 Dec '12

I've added a basic working kamailio.cfg with Shared Call Appearances to the SCA docs. It should help clarify where and when it's necessary to have the sca module inspect traffic. Those of you simply wanting to test SCA can just copy and paste the configuration, updating the listen address as necessary. Until the sca module doc on kamailio.org gets updated, you can find the sample configuration in the git master branch, or view it through gitweb at the link below. Best, andrew <http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob_plain;f=mo…>

1 0

kamailio vs.ser
by Ovidiu Sas 03 Dec '12

03 Dec '12

Hello all, By inspecting the source code, the only difference that I could see between kamailio and ser flavours is that kamailio has support for the "tm:local-request". Are there any constrains in having the tm:local-request present for ser flavour? Does it make sense to continue to build two flavours? My suggestion would be to focus one one single flavour and go forward with it for the next release. Comments? Regards, Ovidiu Sas -- VoIP Embedded, Inc. http://www.voipembedded.com

2 1

health check
by Alex Solt 03 Dec '12

03 Dec '12

Hi, Is there a way to perform health check on kamailio application? AS

2 1

LCR Failover
by Daniel W. Graham 03 Dec '12

03 Dec '12

I am having issues with LCR failure route. I have two gateways specified in LCR table, if I change the address of the gateway with highest priority to a bogus IP, the failure route never seems to takes place and call is never routed to second gateway. Also, I have tested successful calls to both gateways by switching the priorities around. Here is what I receive after the first gateway times out: U kamailio:5060 -> asterisk:5060 SIP/2.0 408 Request Timeout. Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK633b44d8;rport=5060. From: "user" <sip:user@asterisk>;tag=as67c0037f. To: <sip:dialednumber@domain>;tag=d7177f97ed67076dddc3163c85ab0656-f2e5. Call-ID: 248d69ed6495d0a65315458e2ba6c528@asterisk:5060. CSeq: 102 INVITE. Server: SIP Proxy. Content-Length: 0. Config: modparam("tm", "fr_timer", 10000) modparam("tm", "fr_inv_timer", 120000) modparam("lcr", "db_url", DBURL) modparam("lcr", "gw_uri_avp", "$avp(i:709)") modparam("lcr", "ruri_user_avp", "$avp(i:500)") modparam("lcr", "flags_avp", "$avp(i:712)") modparam("lcr", "lcr_id_avp", "$avp(s:lcr_id_avp)") route[LCR] { if (!load_gws(1)) { sl_send_reply("503", "Internal Server Error, Unable to load gateways"); exit; } else { if (next_gw()) { t_on_failure("LCR_FAIL"); route(RELAY); } else { sl_send_reply("503", "Service Unavailable, No available gateways"); exit; }; }; exit; } failure_route[LCR_FAIL] { if (next_gw()) { t_on_failure("LCR_FAIL"); route(RELAY); } else { t_reply("503", "Service Unavailable, No gateways"); exit; }; exit; }

2 7

Kamailio World - Conference & Exhibition - April 16-17, 2013, Berlin, Germany
by Daniel-Constantin Mierla 03 Dec '12

03 Dec '12

Kamailio project is announcing its first dedicated event - Kamailio World - to take place in Berlin, Germany, during April 16-17, 2013. The event is organized by Asipto (http://www.asipto.com) in collaboration with Fraunhofer Fokus Institute (http://www.fokus.fraunhofer.de). The conference is targeting to present commercial products and services build around Kamailio and related projects, as well as technical workshops of how to use Kamailio for various purposes. Along with the conference will be space for exhibition, that will allow companies to make demos and show cases for participants. For more details, visit the web site of the event: http://conference.kamailio.com Besides Kamailio, you will have the chance to interact with other open source VoIP projects such Asterisk, FreeSwitch, SIP Express Media Server, Homer SIP Capture, Siremis, a.s.o., as well as with commercial vendors in real time communications market. -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Jitsi to Jitsi call
by Raj Roy Ghandhi 03 Dec '12

03 Dec '12

Hi Peter, Still I am unable to make call to Jitsi to Jitsi. Call established But no media. But text message works. NAT traversal work with RTPProxy and I can do IPPhone to IPPhone call and voice works very well. Tried with many soft phones and voice did not go through. Did I miss something in the configuration ? Please advice me. Best Regards, Roy.

2 2

Call pickup through a NAT
by Mark Boyce 03 Dec '12

03 Dec '12

Hi All, I've hit my next stumbling block on my mission to get BLF/Call pickup working on Linksys/Cisco phones which are behind a NAT working with Kamailio. I'm using RFC 3265 and pua_dialoginfo to handle subscribes from phones and generate notify packets on status change. The phones are on two different sites behind a couple of NATs. Kamailio is on public IP in the data centre. BLFs work fine. However the <target uri="x.x.x.x."> given in the XML of the notify packet is the that of the Public IP of the NAT so that call pickup fails. I'm guessing that this actually needs to be the address of the Kamailio server? I cannot see where to correct this without breaking tables the server needs to know how to contact the phone when it needs to. Has anyone seen this before? Cheers Mark

2 1

Kamailio SCA Module
by Robert Boisvert 30 Nov '12

30 Nov '12

Andrew, Congratulations on your new position as a Kamailio developer! When I saw your module (SCA) I began testing immediately since it may be a stepping stone to help me get the Polycom barge-in feature working with Kamailio. However, I ran into a few difficulties. Can you assist me? To make your existing code work as a K module I had to change a few lines of code and recompile under modules_k. Specifically, since the K definition of the ucontact structure defines ucontact->aor as str* and the S definition as str**, I changed “&c->aor” to “c->aor” in the following lines of sca_usrloc_cb.c: 73: if ( !sca_uri_is_shared_appearance( sca, c->aor )) { 74: LM_DBG( "%.*s is not a shared appearance line", STR_FMT( c->aor )); 79: &SCA_EVENT_NAME_CALL_INFO, c->aor ) < 0 ) { Since I use PostgreSQL as my SQL engine I created the SCA entries using this code: INSERT INTO version (table_name, table_version) values ('sca_subscriptions', '0'); CREATE TABLE sca_subscriptions ( id SERIAL PRIMARY KEY NOT NULL, subscriber VARCHAR(255) NOT NULL, aor VARCHAR(255) NOT NULL, event INTEGER NOT NULL, expires INTEGER NOT NULL, state INTEGER NOT NULL, app_idx INTEGER NOT NULL, call_id VARCHAR(255) NOT NULL, from_tag VARCHAR(128) NOT NULL, to_tag VARCHAR(128) NOT NULL, notify_cseq INTEGER NOT NULL, subscribe_cseq INTEGER NOT NULL, CONSTRAINT sca_subscriptions_idx UNIQUE (subscriber, call_id, from_tag, to_tag)); I applied the recommendations from your documentation on configuring Kamailio (3.3.1) and inserted the call to “sca_handle_subscribe ()” in kamailio.cfg. Next, I connected two Polycom phones (SP335 and SP650) to my network and configured them to enable presence and use a shared line. Four lines were configured on the phones as shown below: SP335 Line 1 (p11) = private line 6013 SP335 Line 1 (p12) = shared line 1519 SP650 Line 1 (p21) = private line 6014 SP650 Line 2 (p22) = shared line 1519 My tests get inconsistent results on the line LEDs and it seems that they do not work as a shared line. When I ran a tcpdump of the SIP communication (see attached pcap files) I found several anomalies: - Some packets (sca-reboot1 #22, 44, 113, 120 and sca-reboot2 #17, 38) contain spurious NUL characters in the tags section. In all tests it seems to occur in the Expire tag when Kamailio accepts the subscription. - When Kamailio sends call-info NOTIFY the Content-Length tag is missing. - Sometimes the phone does not respond to a call-info NOTIFY event ( sca-reboot1 #23, 45 and sca-reboot2 #18, 39). - I don't have an example here but sometimes the phones respond to the call-info NOTIFY with "481 Call Leg/Transaction Does Not Exist". I think some of my problems are related to these anomalies. Could you help me to determine if this is a problem with the code itself, my failure to properly port it to the K module, or some other issue? Once again, thank you for making this very useful module for Kamailio and for the documentation to make it work. I hope that resolving my problems may help the code to become a part of Kamailio. Sincerely yours, Bob Boisvert

2 1

Linphone presence
by Moacir Ferreira 30 Nov '12

30 Nov '12

I am trying to get Linphone to show presence information. I did my Kamailio setup using just the template from a stardard install and it is working properly with Bria (X-Ten). However, when using Linphone it does not show buddies presence. Any hint? Cheers! Moacir

2 1

Call forking
by Moacir Ferreira 30 Nov '12

30 Nov '12

All the topics/examples I found on the net is not what I am looking for, so here is the question: How do I achieve parallel forking (call all devices) if I have a single user registered using the same account from different devices (i.e.: table phone, iPad and PC)? What I want is to ring all devices a user may be using. Cheers! Moacir

2 1

t_fork_to?
by Juha Heinanen 30 Nov '12

30 Nov '12

core cookbook reads: append_branch Similarly to t_fork_to, it extends destination set by a new entry. The difference is that current URI is taken as new entry. what is t_fork_to? i could not find it in tm README. -- juha

3 3

WITH_PRESENCE charsh Kamailio
by Raj Roy Ghandhi 30 Nov '12

30 Nov '12

Hi, I am running Kamailio 3.4 with web-socket and everything works fine with NAT traversal for IP Phones. But when I enable the PRESENCE it crash with "(3) errors in config file" I have copied the config.cfg to paste-bin at http://pastebin.ca/2257817<http://pastebin.ca/2257815> Please guide me on this issue. Best Regards, Roy.

2 1

Enforced TLS, still some packets are through UDP
by Kamal Palei 30 Nov '12

30 Nov '12

Dear All I ma trying to have all SIP signalling through TLS. Using Kamailio 3.1.5. In a typical setup, endpoints initiates call, messages reach Kamailio proxy, Kamailio proxy forwards these packets to main proxy. In Kamailio.cfg, I have added below code. route { # per request initial checks route(REQINIT); setflag(FLT_NATS); if(is_method("REGISTER")) { t_relay_to("tls:115.114.48.19:443"); exit(); } else { xlog("incoming request\n"); route(WITHINDLG); route(RTPPROXY); t_on_reply("REPLY_ONE"); t_on_failure("FAIL_ONE"); t_relay_to("tls:115.114.48.19:443"); exit(); } .............. I am finding most of the messages are getting transmitted in TLS. But few ACK and REFER / REFER response packets are still being transported in UDP. Can somebody tell me whats the wrong I am doing here. By any chance route(WITHINDLG); route(RTPPROXY); before t_relay_to("tls:115.114.48.19:443"); can cause this ???? And one more doubt.., 115.114.48.19 I get in route header. How can I extract the value of route header, so that I do not have to hard-code Many thanks in advance. -kamal

3 2

Lua env not initialized in event_route
by Vicente Hernando 30 Nov '12

30 Nov '12

Hello I'm trying to execute a Lua script on kamailio startup. I'm calling the script inside event_route[hable:mod-init], but it fails to be executed with this error: [app_lua_mod.c:167]: Lua env not initialized It happens both with lua_run and lua_dofile. I've played with the order in wich the modules are loaded, but I haven't been able to get rid of that message. Kamailio version is 3.3.0. Do you think this is feasible or should I try a different solution? Thanks in advance Regards, Vicente.

2 1

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