- sr-users - kamailio.org

a=nortpproxy:yes
by Austin Einter 13 Oct '11

13 Oct '11

If the SIP UA is not behind the NAT, and Kamailio insersts a=nortpproxy:yes in SDP body, can it cause one sided audio. I am using PJSIP as my UA, kamailio 3.1.5 , rtp proxy 1.2.1 as intermediate proxy for signalling and media. In kamailio config, I have enabled WITH_NAT. On call setup, only one way audio is there. When I looked at SIP messages, the only additional stuff I can see a=nortpproxy and nat=yes in recpord route. Without kamailio as intermediate proxy, audio is good both way. So I feel, when I bring up kamailio as intermediate proxy I am doing some mistake configuration wise either at PJSIP UA or in Kamailio. Has anybody faced this issue previously. If so please share how can we fix this issue. Thanks in advance Austin

2 1

Tls protocol testing with iptel account
by Lello 13 Oct '11

13 Oct '11

Hi, I'm working with a softphone project named mjSip, now I've implemented the TLS protocol in the softphone and I would try it with 2 iptel account setted on two pc, on the first I have: # MjSip UA configuration file # ___________________________________________ # host_port=5061 transport_protocols= tls outbound_proxy=sip.iptel.org from_url=<sip:alexlel@iptel.org> username=alexlel realm=iptel.org passwd=2387lelloh debug_level=8 log_path=log max_logsize=2000 do_register=yes keepalive_time=8000 _______________________________ the second is such as this. However I have no response in SSL/TLS by the iptel server. Does it work in TLS? Thanks, Regards.

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Employment you've been searching!
by apache＠iptel.org 13 Oct '11

13 Oct '11

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1 0

Can't create a new account with kamctl
by Peter Schrock 12 Oct '11

12 Oct '11

I have finally got to the section of the install section "10. Ready to Rock". Unfortunately, I did get the error message "ERROR: domain unknown: use usernames with domain or set default domain in SIP_DOMAIN". I did the two steps that the instructions gave and cannot find ".kamctlrc" to "add SIP_DOMAIN=mysipserver.com" Please help. I am not quite understanding what should be done to fix this Peter

2 1

Kamailio Developer Seminar, Oct 24-24, San Francisco
by Daniel-Constantin Mierla 12 Oct '11

12 Oct '11

Hello, the place and date for Kamailio/SER Developer Seminar have been set: San Mateo (just south of San Francisco, CA, in the nearby of international airport), for one day and a half, Oct 24-25, 2011. I posted more details, including about content and registration, at: * http://www.kamailio.org/w/2011/10/kamailio-developer-seminar-san-francisco-… Participating fee is 250USD. Selection of attendees is done by the rule of first come first served. The number of seats is limited, don't wait too long if you want to attend. Looking forward to meeting many of you there. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 1

Can anyone help for a small amount of money.
by Henrik Aagaard Sørensen 12 Oct '11

12 Oct '11

Hi everyone. I'm still facing problems with my Kamailio Dispatcher <-> FreeSWITCH setup. Everything works when connecting directly to FreeSWITCH. When connecting to Kamailio, which should work as a dispatcher (load-balancer) in front of FreeSWITCH I get "Too many hops", "Timeouts" etc. My test setup with SIP clients is an iPhone, an Android, an iPad and a laptops with Ubuntu. Can someone help me out, for a small amount of money, until I can get it to work? Hopefully it's just the kamailio.cfg which needs to do things correctly. Currently I'm using this http://pastebin.com/mahKECAw with help from Morten in this mailing list.

1 0

Creating a database error
by Peter Schrock 12 Oct '11

12 Oct '11

This is what I am getting: INFO: test server charset /usr/local/lib/kamailio//kamctl/kamdbctl.mysql: line 105: mysql: command not found /usr/local/lib/kamailio//kamctl/kamdbctl.mysql: line 106: mysql: command not found Usage: grep [OPTION]... PATTERN [FILE]... Try `grep --help' for more information. /usr/local/lib/kamailio//kamctl/kamdbctl.mysql: line 112: [: =: unary operator expected INFO: creating database openser ... /usr/local/lib/kamailio//kamctl/kamdbctl.mysql: line 71: mysql: command not found ERROR: Creating core database and grant privileges failed! I looked at the files and cannot find the problem. Please help. Peter

3 4

Kamailio Dispatcher and FreeSwitch, Too many hops.
by Henrik Aagaard Sørensen 12 Oct '11

12 Oct '11

I have a setup with Kamailio as dispatcher in front of a FreeSwitch server. This is my kamailio.cfg: http://pastebin.com/8PR2GFBD I'm currently getting "Too many hops" when calling between SIP clients. I am able to call to FreeSwitch and listen to voicemail, hold music etc. After a long conversation with a FreeSwitch expert, and some tests, I was told that Kamailio delivers the wrong IP (NAT problems) to FreeSwitch. I've also run tshark on both FreeSwitch and Kamailio and when calling between clients they just send the packets between each other. Can anyone help me out? I've tried to Google a lot for this problem and asked in several IRC channels, mailing lists and forums. Without any luck.

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Re: [SR-Users] Problem with Route containing two elements
by davy van de moere 12 Oct '11

12 Oct '11

Owki, reading RFC-3261 again ;) Imho Kamailio does what it is supposed to do, it forwards the packet to the first element in the Route header. But as I received the BYE packet from <sip:their_sbc> , that one should have stripped itself out of the Route header, making everything just work, right? Now I can hackishly fix it with a rewritehostport in the middle. But that will eventually not scale up that nicely. Ergo, does anyone have a good idea on how to e.g. have Kamailio in this situation ignore the first uri in the route header? Or am I still not getting it :) ? thx ! BYE sip:05xxxxxx at my_kamailio <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>:5060;transport=udp SIP/2.0 Record-Route: <sip:their_sbc;lr=on;ftag=as5716fec8> Via: SIP/2.0/UDP their_sbc;branch=z9hG4bK1ae8.269e4f67.0 Via: SIP/2.0/UDP their_internals:5060;received=their_internals;branch=z9hG4bK4f9044a7;rport=5060 Route: <sip:their_sbc;lr=on;ftag=as5716fec8>,<sip:36d0a5cfbaeb6b055d5bb02c32127b49_32123456789 at my_kamailio <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> ;lr=on;ftag=as5716fec8> Max-Forwards: 69 From: "Some_Telecom" <sip:02xxxxxxxx at their_sbc <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>>;tag=as5716fec8 To: <sip:05xxxxxx at their_sbc <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>:5060>;tag=U3FrFDm5Zr0KF Call-ID: 68fe6a470986bd087ce4ed860f603af6 at their_sbc <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> CSeq: 103 BYE User-Agent: Integrics Enswitch X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 X-Enswitch-RURI: sip:05xxxxxx at my_kamailio <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>:5060;transport=udp X-Enswitch-Source: their_internals:5060

2 2

proxy BYE to itself
by caio 12 Oct '11

12 Oct '11

Hello, I have the following issue. A calls B, talk for a while, then B sends a BYE, but A never receives it. The BYE is transmited from the proxy listening interface:port to itself, but not to the asterisk port. U 2011/10/08 00:29:29.207335 Kamailio_IP:5060 -> Kamailio_IP:5060 BYE sip:Kamailio_IP;lr=on SIP/2.0. The scenario is: A (pstn subs) --> Asterisk:5080 --> Kamailio:5060 --> B (registered sip ua) Asterisk and Kamailio are in the same box, listening both at the same IP (eth iface) address, but one at port 5080, and the proxy at 5060 (udp). The configuration is based on kamailio.cfg file provided by the v3.1.5 (with a few modules added, like dialplan and cr..etc). If A is who send the BYE, everything goes well. I attach the full sip capture. Thanks for you time. Claudio

1 1

Problem with Route containing two elements
by davy van de moere 11 Oct '11

11 Oct '11

I'm trying to integrate to an integrics enswitch, almost everything works like a charm, except on BYE packets as Kamailio in my setup forwards these incorrectly. Digging somewhat I believe the culprit sits in the Route header which comes from the enswitch: Route: <sip:their_sbc;lr=on;ftag=as5716fec8>,<sip:36d0a5cfbaeb6b055d5bb02c32127b49_32123456789@my_kamailio ;lr=on;ftag=as5716fec8> Kamailio takes into account the first argument (the thing sip:their_sbc...) on which it cannot find the correct transaction. And forwards it back to the sender, instead of forwarding it correctly to my freeswitch. which gives a storm of traffic ;) If I compare that with an ACK packet in the same transaction , having this as a Route header : Route: <sip:36d0a5cfbaeb6b055d5bb02c32127b49_32123456789@my_kamailio ;lr=on;ftag=as5716fec8> Kamailio sends it along correctly. IMHO if I could convince Kamailio to always take the last part of the Route header into account and ignore the first one it would correctly work. What would be a good approach? Rewriting the route header looks abit harsch. disclaimer : I'm not sure which component is disrespecting the RFC, if any, or if I just fully not get something :) Short overview + packet detail (with some obfuscation): My setup: their_sbc --- my_kamailio --- freeswitch --- my_kamailio --- enduser BYE sip:05xxxxxx@my_kamailio:5060;transport=udp SIP/2.0 Record-Route: <sip:their_sbc;lr=on;ftag=as5716fec8> Via: SIP/2.0/UDP their_sbc;branch=z9hG4bK1ae8.269e4f67.0 Via: SIP/2.0/UDP their_internals:5060;received=their_internals;branch=z9hG4bK4f9044a7;rport=5060 Route: <sip:their_sbc;lr=on;ftag=as5716fec8>,<sip:36d0a5cfbaeb6b055d5bb02c32127b49_32123456789@my_kamailio ;lr=on;ftag=as5716fec8> Max-Forwards: 69 From: "Some_Telecom" <sip:02xxxxxxxx@their_sbc>;tag=as5716fec8 To: <sip:05xxxxxx@their_sbc:5060>;tag=U3FrFDm5Zr0KF Call-ID: 68fe6a470986bd087ce4ed860f603af6@their_sbc CSeq: 103 BYE User-Agent: Integrics Enswitch X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 X-Enswitch-RURI: sip:05xxxxxx@my_kamailio:5060;transport=udp X-Enswitch-Source: their_internals:5060 BYE sip:36d0a5cfbaeb6b055d5bb02c32127b49_32123456789@my_kamailio;lr=on;ftag=as5716fec8 SIP/2.0 Record-Route: <sip:05xxxxxx@my_kamailio;lr=on;ftag=as5716fec8> Record-Route: <sip:their_sbc;lr=on;ftag=as5716fec8> Via: SIP/2.0/UDP my_kamailio;branch=z9hG4bK1ae8.b4f95af2.0 Via: SIP/2.0/UDP my_kamailio;rport=5060;branch=z9hG4bK1ae8.a4f95af2.0 Via: SIP/2.0/UDP their_sbc;rport=5060;branch=z9hG4bK1ae8.269e4f67.0 Via: SIP/2.0/UDP their_internals:5060;received=their_internals;branch=z9hG4bK4f9044a7;rport=5060 Route: <sip:their_sbc;lr=on;ftag=as5716fec8> Max-Forwards: 67 From: "Some_Telecom" <sip:02xxxxxxxx@their_sbc>;tag=as5716fec8 To: <sip:05xxxxxx@their_sbc:5060>;tag=U3FrFDm5Zr0KF Call-ID: 68fe6a470986bd087ce4ed860f603af6@their_sbc CSeq: 103 BYE User-Agent: Integrics Enswitch X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 X-Enswitch-RURI: sip:05xxxxxx@my_kamailio:5060;transport=udp X-Enswitch-Source: their_internals:5060

2 1

[OT] SIP on the Web
by Iñaki Baz Castillo 11 Oct '11

11 Oct '11

Hi folks, I would like to show a project my colleage José Luis Millán and me are working on. It's about SIP protocol running in a web browser. When RTCweb (media capable web browsers) becomes a reality both technologies together will allow real SIP endpoints coded in JavaScript running in a web browser and interoperating with real SIP networks. Bye bye Flash and custom protocol gateways. http://sip-on-the-web.aliax.net/ Cheers. -- Iñaki Baz Castillo <ibc(a)aliax.net>

4 10

limiting concurrent calls with dialog module
by Uri Shacked 11 Oct '11

11 Oct '11

Hi, I am trying to create concurrent calls limitations. two questions: 1. should i use dialog module? or is there a better module? 2. i was trying to work with the dialog module, but didnt understand how and where to configure the commands and which to use.... any examples or ideas? BR, Uri

10 17

releasing v3.2.0
by Daniel-Constantin Mierla 11 Oct '11

11 Oct '11

Hello, it is time to set the time to release v3.2.0. From my point of view, testing went very well, considering also that core and main modules were not touched much during this devel cycle, the most of the efforts were on new modules and application service components such as presence server, Lua ... A convenient date for me will be Tuesday, October 18, 2011, allowing to work a bit on packaging and docs. If there are other preferences, jump in this discussion. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

4 8

(no subject)
by alan chiu 10 Oct '11

10 Oct '11

http://toehugger.com/CRE/info.php?html143

1 0

Audio quality issue
by Austin Einter 10 Oct '11

10 Oct '11

Hi I am using Kamailio 3.1.5 . I am using RTP proxy also. I have used default kamailio.cfg.sample fiile , and just added line #!define WITH_NAT. I have another Main proxy. I wanted all my signalling and media packets should just pass through machine where Kamailio and RTP proxy are running. With this I found, call is established, all signalling and media packets are passing through kamailio / rtp-proxy. So far so good. One way audio stream (from called party to calling party) quality is good. The other audio stream (from calling party to called party is very bad. Did anybody face this issue? Please help me to sort out this issue audio quality issue. Regards Austin

4 5

Carrierroute: Could not allocate shared memory
by Mino Haluz 10 Oct '11

10 Oct '11

Hi, is there any setting which could allow me to set maximum memory per module ? As I am testing the carrier route module, I've added for testing purposes 100 000 rules sofar. When I start the kamailio, it gives me : Oct 10 14:46:46 kamrouter /usr/sbin/kamailio[2182]: ERROR: carrierroute [../../ut.h:702]: could not allocate shared memory from available pool Oct 10 14:46:46 kamrouter /usr/sbin/kamailio[2182]: ERROR: carrierroute [cr_rule.c:155]: could not allocate shared memory from available pool Thanks.

3 4

Memory Allocation in Kamailio
by Austin Einter 08 Oct '11

08 Oct '11

Hi Just a quick question. How memory is allocated in Kamailio code. Few places I saw pkg_malloc function. Is it the only function used to allocate memory or there are similar functions present in kamailio. Thanks Austin

3 2

tm and msilo module
by Mark Anthony C. Delfin 08 Oct '11

08 Oct '11

Dear List, I am trying to enable msilo on Kamailio but I'm getting an error ct 8 11:30:30 localhost kamailio[13856]: ERROR: tm [tm_load.c:48]: tm:load_tm: Module not initialized yet, make sure that all modules that need tm module are loaded after tm in the configuration file Oct 8 11:30:30 localhost kamailio[13856]: INFO: sl [sl.c:163]: could not bind tm module - only stateless mode available Oct 8 11:30:30 localhost kamailio[13856]: ERROR: tm [tm_load.c:48]: tm:load_tm: Module not initialized yet, make sure that all modules that need tm module are loaded after tm in the configuration file Oct 8 11:30:30 localhost kamailio[13856]: ERROR: msilo [msilo.c:333]: can't load TM API Oct 8 11:30:30 localhost kamailio[13856]: ERROR: <core> [sr_module.c:875]: init_mod(): Error while initializing module msilo (/usr/local/lib64/kamailio/modules_k/msilo.so) Any ideas how can I fixed it. Thank you in Advance. Regards, Mark Anthony C. Delfin

2 1

Kamailio RTP Proxy Questions
by Austin Einter 07 Oct '11

07 Oct '11

Hi I downloaded 3.1.5 Kamalio source, did make and install. Tried to run, and tested few basic scenarios, everything working fine. I have a main proxy, want to use kamailio as intermediate proxy and my requirement is all RTP packets should pass through machine in which Kamalio proxy is running. I beleive RTP Proxy I can run in same machine where Kamailio proxy is running with proper configuration . For this I am referring, http://nil.uniza.sk/sip/nat-fw/configuring-nat-traversal-using-kamailio-31-… page. In one of the steps it says "So, if we are using the Rtpproxy server with default configuration, we have to open /etc/default/rtpproxy file and uncomment following line regarding of udp socket, that will be sued for interconnection:". I tried to search rtpproxy configuration file, could not locate. While making and installing Kamalio proxy, the commands I used "make prefix=/usr/local/ all and make prefix=/usr/local/ install". Can somebody point me why rtpproxy config file is missing. Also the link I am refering to is correct or wrong one. Best Regards, Austin

2 6

Multiple INVITEs and discarded call_control
by Mino Haluz 07 Oct '11

07 Oct '11

Hi, my kamailio server is receiving from some customers 3 identical INVITEs when call is initiated (separated by 200ms). Those 3 INVITEs are making a big problem with call_control: WARNING: call_control [call_control.c:1156]: dialog to trace controlled call was not created. discarding callcontrol. That is why, the prepaid limit is not working at all in this case. This way the user can hack the prepaid protection of the account. Otherwise the call_control is fuilly functional. Anybody experienced the similar problem? If so, how to resolve it? Thanks, Mino Haluz

2 4

no more pkg mem on ul_dump
by Klaus Darilion 06 Oct '11

06 Oct '11

Hi! Kamailio 1.3.4: "kamctl ul monitor" causes: ERROR: <core> [attr.c:80]: no more pkg mem (21) ERROR: mi_fifo [fifo_fnc.c:513]: command (ul_dump) processing failed There are ~ 2000 entries in location table. Default PKG size is used. I remember there were some similar reports recently, but couldn't find the relevant threads. Any hints for the cause of this problem and workarounds? Thanks Klaus

4 8

Compiling on a 2.4 kernel
by Pedro Antonio Vico Solano 06 Oct '11

06 Oct '11

2 3

Problems with domain on registrations.
by Henrik Aagaard Sørensen 06 Oct '11

06 Oct '11

I have the current setup: SIP client <-> SER Proxy <-> SIP Trunk. My reason for the SER Proxy is that the SIP trunk only allows connections from approved IP addresses (such as my server with SER Proxy). My user on the SIP Trunk has the following authentication username@domain. I am able to register to the SIP Trunk as I'm setting up my SIP klient with username to username, domain to domain and then add my SER Proxy IP address as proxy. This works and means that I'm able to place calls to the SIP Trunk. BUT people cannot place calls from the SIP Trunk to my SIP client, due to the fact that on the SIP Trunk my locations is registered as username@SER-Procy-IP. So when people are calling me, the SER Proxy is looking for username@SER-Procy-IP and NOT username@domain, as I'm registered with. And then SER Proxy reports: 404 Not Found. Can anyone help me how to set this up correctly? I do not have access to the SIP Trunk, but I have all access to the SIP client and SIP Proxy.

2 2

Removing Contact from location table when TCP connection from UA is dropped...
by Krishna Kurapati 06 Oct '11

06 Oct '11

Hi, When UA connects with the Proxy over TCP, the contact is mostly bound to the firewall's binding IP and port for that connection. When the connection is dropped, the contact in Kamailio is no longer valid even though there is time left to expire the contact. Is there some way to link TCP connection and the contact in such a way that when TCP connection is dropped, contact from the location can be removed? Thanks Krish Kura

2 1

New in Kamailio
by Leonardo Sena 05 Oct '11

05 Oct '11

Hello I´m a new contribuitor of the list and my english dont very good then i hope of help more in the future I am installing Kamailio and Siremis but i dont understand like siremis works and where i can create the routes i would like step by step to routes rules. Best Regards, Leonardo Sena

1 0

Balancing incoming traffic (DID)
by Anto 05 Oct '11

05 Oct '11

Hello It presents a problem because I do not know how to assimilate the implementation. Until now we had a design of a proxy to balance out between several gateway, but it will add another proxy for another service (other users) and use the same gateway output or perhaps different (although not that be better if divide or unify the gateway). The problem as I have with the routing entry, which is not very well how to implement it. Now we have the design like this: -GW ----\ / - CARRIER1 USER ------ KAMAILIO ----- GW ---- + \ - CARRIER2 -GW ----/ And becomes like this: -GW ----\ / - CARRIER1 USER ------ KAMAILIOSERVICE1 ----- GW ---- + \ - CARRIER2 -GW ----/ -GW ----\ / - CARRIER1 USER ------ KAMAILIOSERVICE2 ----- GW ---- + \ - CARRIER2 -GW ----/ Or maybe using the same gateway. The problem is that each carrier provides DID number (I use the same for both services). Kamailio had thought to put a gateway between the carriers and that they did check the service and send it to the corresponding gateway (in the case of the gateway separately), or put it in the proxy gateway and making checking for routing to proxy (in the case of using the same gateway). What would be better? Any recommendations on how to put this into practice?. The biggest question that I believe is the routing table, as if this new proxy routes the traffic into and leaves her in the field via ip, also pass through the output data (I have doubts whether SIP level, following standard, you can not pass through the exit) and want it to be direct (USER -> Proxy -> GW -> CARRIER), just past the entrance. Anyone could advise me where I could find more information to set up this scenario?. thanks and sorry for bothering you with this question ;-). best regards

2 2

limiting concurrent calls with dialog module
by Uri Shacked 05 Oct '11

05 Oct '11

Hi, Thanks for the help. The idea of using SQLOPS for many things crossed my mind.... But....I am using ACC with DB, Dialog with DB and so... isnt it too much? i mean for load and capacity reasons? BR, Uri

2 1

Re: [SR-Users] Dialog ref Error
by Uri Shacked 05 Oct '11

05 Oct '11

On Wed, Oct 5, 2011 at 2:34 PM, Uri Shacked <ushacked(a)gmail.com> wrote: > OK... > now i see the problem is some where in the cfg file... > i get te following error : > > 0(3774) : <core> [cfg.y:3406]: parse error in config file > //etc/kamailio/kamailio.cfg, from line 944, column 2 to line 945, column 0: > syntax error > 0(3774) : <core> [cfg.y:3406]: parse error in config file > //etc/kamailio/kamailio.cfg, from line 944, column 2 to line 945, column 0: > bad command > ERROR: bad config file (2 errors) > 0(3774) DEBUG: tm [t_funcs.c:122]: DEBUG: tm_shutdown : start > 0(3774) DEBUG: tm [t_funcs.c:125]: DEBUG: tm_shutdown : emptying hash > table > 0(3774) DEBUG: tm [t_funcs.c:127]: DEBUG: tm_shutdown : removing > semaphores > 0(3774) DEBUG: tm [t_funcs.c:129]: DEBUG: tm_shutdown : destroying tmcb > lists > 0(3774) DEBUG: tm [t_funcs.c:132]: DEBUG: tm_shutdown : done > 0(3774) DEBUG: <core> [mem/shm_mem.c:236]: shm_mem_destroy > 0(3774) DEBUG: <core> [mem/shm_mem.c:239]: destroying the shared memory > lock > > > here is my complete cfg file. > any ideas? > > > ... > > ######## Routing Logic ######## > > # Main SIP request routing logic > > # - processing of any incoming SIP request starts with this route > > route { > > setflag(FLT_TRC); # Trace message > > # per request initial checks > > route(REQINIT); > > # NAT detection > > route(NAT); > > # 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(); > > # authentication > > route(AUTH); > > # 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")) > > { > > #!ifdef WITH_DIALPLAN > > dp_translate("1", "$rU/$rU"); > > xlog("translated to $rU \n"); > > #!endif > > #!ifdef WITH_CARRIERROUTE > > ####cr_user_carrier("$rU", "$td", "$avp(carrierid)"); > > ####xlog("translated to var $avp(carrierid) \n"); > > cr_route("OSP1", "kamailio2", "$rU", "$rU", "call_id"); > > #!endif > > setflag(FLT_ACC); # do accounting > > } > > # dispatch requests to foreign domains > > route(SIPOUT); > > ### requests for my local domains > > # handle presence related requests > > route(PRESENCE); > > # handle registrations > > route(REGISTRAR); > > if ($rU==$null) > > { > > # request with no Username in RURI > > sl_send_reply("484","Address Incomplete"); > > exit; > > } > > # dispatch destinations to PSTN > > route(PSTN); > > # user location service > > route(LOCATION); > > route(RELAY); > > } > > route[RELAY] { > > #!ifdef WITH_NAT > > if (check_route_param("nat=yes")) { > > setbflag(FLB_NATB); > > } > > if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) { > > route(RTPPROXY); > > } > > #!endif > > /* example how to enable some additional event routes */ > > if (is_method("INVITE")) { > > #t_on_branch("BRANCH_ONE"); > > t_on_reply("REPLY_ONE"); > > t_on_failure("FAIL_ONE"); > > } > > if (!t_relay()) { > > sl_reply_error(); > > } > > exit; > > } > > # Per SIP request initial checks > > route[REQINIT] { > > #!ifdef WITH_ANTIFLOOD > > # flood dection from same IP and traffic ban for a while > > # be sure you exclude checking trusted peers, such as pstn gateways > > # - local host excluded (e.g., loop to self) > > if(src_ip!=myself) > > { > > if($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; > > } > > #!ifdef WITH_PIPELIMIT > > # perform pipe match for INVITE method > > if (is_method("INVITE")) { > > if (!pl_check("avg")) { > > pl_drop(); > > exit; > > } > > if (!pl_check("peek")) { > > pl_drop(); > > exit; > > } > > #endif > > #!ifdef WITH_DIALOG > > xlog("currently active dialogs are $DLG_count\n"); > > dlg_manage(); > > dlg_setflag(FLT_DLG); > > if (is_method("INVITE")) { > > $var(size) = 0; > > get_profile_size("test","$rU","$var(size)"); > > xlog("currently, the user $rU has $var(size) active calls\n"); > > if( $var(size) >= 5 ) > > { > > sl_send_reply("507", "Simultaneous calls limit reached"); > > xlog("Limit of $var(size) active calls has been reached \n"); > > exit; > > } > > set_dlg_profile("test","$rU"); > > } > > #!endif > > } > > # 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()) { > > if (is_method("BYE")) { > > setflag(FLT_ACC); # do accounting ... > > setflag(FLT_ACCFAILED); # ... 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() ) { > > # 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")) > > { > > if(isflagset(FLT_NATS)) > > { > > setbflag(FLB_NATB); > > # uncomment next line to do SIP NAT pinging > > ## setbflag(FLB_NATSIPPING); > > } > > if (!save("location")) > > sl_reply_error(); > > exit; > > } > > } > > # USER location service > > route[LOCATION] { > > #!ifdef WITH_ALIASDB > > # search in DB-based aliases > > alias_db_lookup("dbaliases"); > > #!endif > > if (!lookup("location")) { > > switch ($rc) { > > case -1: > > case -3: > > t_newtran(); > > t_reply("404", "Not Found"); > > exit; > > case -2: > > sl_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; > > #!ifdef WITH_PRESENCE > > if (!t_newtran()) > > { > > sl_reply_error(); > > exit; > > }; > > if(is_method("PUBLISH")) > > { > > handle_publish(); > > t_release(); > > } > > else > > if( is_method("SUBSCRIBE")) > > { > > handle_subscribe(); > > t_release(); > > } > > exit; > > #!endif > > # if presence enabled, this part will not be executed > > if (is_method("PUBLISH") || $rU==$null) > > { > > sl_send_reply("404", "Not here"); > > exit; > > } > > return; > > } > > # Authentication route > > route[AUTH] { > > #!ifdef WITH_AUTH > > if (is_method("REGISTER")) > > { > > # authenticate the REGISTER requests (uncomment to enable auth) > > if (!www_authorize("$td", "subscriber")) > > { > > www_challenge("$td", "0"); > > exit; > > } > > if ($au!=$tU) > > { > > sl_send_reply("403","Forbidden auth ID"); > > exit; > > } > > } else { > > #!ifdef WITH_IPAUTH > > if(allow_source_address()) > > { > > # source IP allowed > > return; > > } > > #!endif > > # authenticate if from local subscriber > > if (from_uri==myself) > > { > > if (!proxy_authorize("$fd", "subscriber")) { > > proxy_challenge("$fd", "0"); > > exit; > > } > > if (is_method("PUBLISH")) > > { > > if ($au!=$tU) { > > sl_send_reply("403","Forbidden auth ID"); > > exit; > > } > > } else { > > if ($au!=$fU) { > > sl_send_reply("403","Forbidden auth ID"); > > exit; > > } > > } > > consume_credentials(); > > # caller authenticated > > } else { > > # caller is not local subscriber, then check if it calls > > # a local destination, otherwise deny, not an open relay here > > if (!uri==myself) > > { > > sl_send_reply("403","Not relaying"); > > exit; > > } > > } > > } > > #!endif > > return; > > } > > # Caller NAT detection route > > route[NAT] { > > #!ifdef WITH_NAT > > force_rport(); > > if (nat_uac_test("19")) { > > if (method=="REGISTER") { > > fix_nated_register(); > > } else { > > fix_nated_contact(); > > } > > setflag(FLT_NATS); > > } > > #!endif > > return; > > } > > # RTPProxy control > > route[RTPPROXY] { > > #!ifdef WITH_NAT > > if (is_method("BYE")) { > > unforce_rtp_proxy(); > > } else if (is_method("INVITE")){ > > force_rtp_proxy(); > > } > > if (!has_totag()) add_rr_param(";nat=yes"); > > #!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] { > > #!ifdef WITH_PSTN > > # check if PSTN GW IP is defined > > if (strempty($sel(cfg_get.pstn.gw_ip))) { > > xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); > > return; > > } > > # route to PSTN dialed numbers starting with '+' or '00' > > # (international format) > > # - update the condition to match your dialing rules for PSTN routing > > if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) > > return; > > # only local users allowed to call > > if(from_uri!=myself) { > > sl_send_reply("403", "Not Allowed"); > > exit; > > } > > $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); > > route(RELAY); > > exit; > > #!endif > > return; > > } > > # XMLRPC routing > > #!ifdef WITH_XMLRPC > > route[XMLRPC] > > { > > # allow XMLRPC from localhost > > if ((method=="POST" || method=="GET") > > && (src_ip==127.0.0.1)) { > > # close connection only for xmlrpclib user agents (there is a bug in > > # xmlrpclib: it waits for EOF before interpreting the response). > > if ($hdr(User-Agent) =~ "xmlrpclib") > > set_reply_close(); > > set_reply_no_connect(); > > dispatch_rpc(); > > exit; > > } > > send_reply("403", "Forbidden"); > > exit; > > } > > #!endif > > # Sample branch router > > branch_route[BRANCH_ONE] { > > xdbg("new branch at $ru\n"); > > } > > # Sample onreply route > > onreply_route[REPLY_ONE] { > > xdbg("incoming reply\n"); > > #!ifdef WITH_NAT > > if ((isflagset(FLT_NATS) || isbflagset(FLB_NATB)) > > && status=~"(183)|(2[0-9][0-9])") { > > force_rtp_proxy(); > > } > > if (isbflagset("6")) { > > fix_nated_contact(); > > } > > #!endif > > } > > route[STATS] { > > sql_query("ca", > > "insert into statistics (time_stamp,shm_used_size,shm_real_used_size," > > "shm_max_used_size,shm_free_used_size,ul_users,ul_contacts) values ($Ts," > > "$stat(used_size),$stat(real_used_size),$stat(max_used_size)," > > "$stat(free_size),$stat(location-users),$stat(location-contacts))", > > "ra"); > > } > > # Sample failure route > > failure_route[FAIL_ONE] { > > #!ifdef WITH_NAT > > if (is_method("INVITE") > > && (isbflagset(FLB_NATB) || isflagset(FLT_NATS))) { > > unforce_rtp_proxy(); > > } > > #!endif > > if (t_is_canceled()) { > > exit; > > } > > # uncomment the following lines if you want to block client > > # redirect based on 3xx replies. > > ##if (t_check_status("3[0-9][0-9]")) { > > ##t_reply("404","Not found"); > > ## exit; > > ##} > > # uncomment the following lines if you want to redirect the failed > > # calls to a different new destination > > ##if (t_check_status("486|408")) { > > ## sethostport("192.168.2.100:5060"); > > ## append_branch(); > > ## # do not set the missed call flag again > > ## t_relay(); > > ##} > > } > > On Wed, Oct 5, 2011 at 12:12 PM, Uri Shacked <ushacked(a)gmail.com> wrote: > >> >> >> ---------- Forwarded message ---------- >> From: Uri Shacked <uri.shacked(a)gmail.com> >> Date: Wed, Oct 5, 2011 at 11:48 AM >> Subject: dialog ref Error >> To: sr-users(a)lists.sip-router.org >> >> >> Hi, >> >> i keep on getting this error: >> Oct 5 14:39:33 Kamailio2 /usr/sbin/kamailio[21568]: DEBUG: dialog >> [dlg_hash.c:247]: new dialog on hash 531 >> Oct 5 14:39:33 Kamailio2 /usr/sbin/kamailio[21568]: DEBUG: dialog >> [dlg_handlers.c:264]: route_set , contact sip:036264529@10.2.0.55:5061, >> cseq 1 and bind_addr udp:10.2.0.5:5060 >> Oct 5 14:39:33 Kamailio2 /usr/sbin/kamailio[21568]: DEBUG: dialog >> [dlg_hash.c:519]: ref dlg 0xb417f5a0 with 3 -> 3 >> Oct 5 14:39:33 Kamailio2 /usr/sbin/kamailio[21568]: DEBUG: dialog >> [dlg_hash.c:600]: unref dlg 0xb417f5a0 with 1, crt ref count: 3 >> >> anyone knows what it is? >> >> here is the dialog part in my cfg: >> >> >> *# ------ dialog params --------* >> >> modparam("dialog", "db_url", DBURL) >> >> ##modparam("dialog", "rr_param", "did") >> >> modparam("dialog", "default_timeout", 7200) >> >> modparam("dialog", "dlg_match_mode", >> 0) >> >> modparam("dialog", "detect_spirals", 1) >> >> modparam("dialog", "db_mode", 1) >> >> ##modparam("dialog", "db_fetch_rows", 100) >> >> modparam("dialog", "dlg_flag", FLT_DLG) >> >> modparam("dialog", "profiles_with_value", "test") >> >> >> >> .......... >> >> dlg_manage(); >> >> ##dlg_setflag(FLT_DLG); >> >> if (is_method("INVITE")) { >> >> $var(size) = 0; >> >> get_profile_size("test","$rU","$var(size)"); >> >> xlog("currently, the user $rU has $var(size) active calls\n"); >> >> if( $var(size) >= 5 ){ >> >> sl_send_reply("507", "Simultaneous calls limit reached"); >> >> xlog("Limit of $var(size) active calls has been reached \n"); >> >> exit;} >> >> set_dlg_profile("test","$rU");} >> >> ........ >> >> >

2 1

Dialog ref Error
by Uri Shacked 05 Oct '11

05 Oct '11

---------- Forwarded message ---------- From: Uri Shacked <uri.shacked(a)gmail.com> Date: Wed, Oct 5, 2011 at 11:48 AM Subject: dialog ref Error To: sr-users(a)lists.sip-router.org Hi, i keep on getting this error: Oct 5 14:39:33 Kamailio2 /usr/sbin/kamailio[21568]: DEBUG: dialog [dlg_hash.c:247]: new dialog on hash 531 Oct 5 14:39:33 Kamailio2 /usr/sbin/kamailio[21568]: DEBUG: dialog [dlg_handlers.c:264]: route_set , contact sip:036264529@10.2.0.55:5061, cseq 1 and bind_addr udp:10.2.0.5:5060 Oct 5 14:39:33 Kamailio2 /usr/sbin/kamailio[21568]: DEBUG: dialog [dlg_hash.c:519]: ref dlg 0xb417f5a0 with 3 -> 3 Oct 5 14:39:33 Kamailio2 /usr/sbin/kamailio[21568]: DEBUG: dialog [dlg_hash.c:600]: unref dlg 0xb417f5a0 with 1, crt ref count: 3 anyone knows what it is? here is the dialog part in my cfg: *# ------ dialog params --------* modparam("dialog", "db_url", DBURL) ##modparam("dialog", "rr_param", "did") modparam("dialog", "default_timeout", 7200) modparam("dialog", "dlg_match_mode", 0) modparam("dialog", "detect_spirals", 1) modparam("dialog", "db_mode", 1) ##modparam("dialog", "db_fetch_rows", 100) modparam("dialog", "dlg_flag", FLT_DLG) modparam("dialog", "profiles_with_value", "test") .......... dlg_manage(); ##dlg_setflag(FLT_DLG); if (is_method("INVITE")) { $var(size) = 0; get_profile_size("test","$rU","$var(size)"); xlog("currently, the user $rU has $var(size) active calls\n"); if( $var(size) >= 5 ){ sl_send_reply("507", "Simultaneous calls limit reached"); xlog("Limit of $var(size) active calls has been reached \n"); exit;} set_dlg_profile("test","$rU");} ........

2 1

kamailio sip-capture error on compile
by Jeff Anderson 05 Oct '11

05 Oct '11

I am trying to get kamailio sip-capture up and running for use with homer. I can get the service to start but i receive the following error. [root@homer kamailio]# /usr/local/sbin/kamailio -c loading modules under /usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/ 0(10208) WARNING: <core> [socket_info.c:912]: WARNING: fix_hostname: could not rev. resolve 10.40.0.22 Listening on udp: 10.40.0.22:5060 Aliases: config file ok, exiting... [root@homer kamailio]# kamctl start INFO: Starting Kamailio : INFO: started (pid: 10226) [root@homer kamailio]# [root@homer kamailio]# tail -f /var/log/messages Oct 3 05:30:17 homer /usr/local/sbin/kamailio[1554]: NOTICE: <core> [main.c:704]: Thank you for flying kamailio Oct 3 05:30:32 homer kamailio: WARNING: <core> [socket_info.c:912]: WARNING: fix_hostname: could not rev. resolve 10.40.0.22 Oct 3 05:31:59 homer kernel: device eth0 entered promiscuous mode Oct 3 05:32:47 homer kernel: device eth0 left promiscuous mode Oct 3 05:52:10 homer /usr/local/sbin/kamailio[1753]: NOTICE: <core> [main.c:704]: Thank you for flying kamailio Oct 3 05:52:12 homer kamailio: WARNING: <core> [socket_info.c:912]: WARNING: fix_hostname: could not rev. resolve 10.40.0.22 Oct 3 05:55:35 homer kernel: device eth0 entered promiscuous mode Oct 3 05:57:46 homer kernel: device eth0 left promiscuous mode Oct 3 07:05:35 homer /usr/local/sbin/kamailio[1894]: NOTICE: <core> [main.c:704]: Thank you for flying kamailio Oct 3 07:18:43 homer kamailio: WARNING: <core> [socket_info.c:912]: WARNING: fix_hostname: could not rev. resolve 10.40.0.22 ^C [root@homer kamailio]# Does anyone have any ideas on how to resolve this error? Thanks, Jeff Anderson

3 3

Installing Versign Intermediate CA Certificate In Kamailio 3.1.5
by arif tuhin 05 Oct '11

05 Oct '11

I've got trial certificate from verisign. They signed the server certificate with an intermediate ca and gave instructions for installing it into apache/IIS etc. Both issued certificate and the intermediate CA should be installed in the server. if only server certificate is installed , it will not be trusted by default. Now problem is in apache mod_ssl there is a field called ServerChainFile where the intermediate ca name should be in place, but in the tls module documentation, i find no place for intermediate ca. How should i install the intermediate ca the? Thanks in advance.

1 0

db_text
by Bruno Bresciani 04 Oct '11

04 Oct '11

Hello ALL, Someone knows why at line 149 in .../kamailio-3.1.2/modules_k/ db_text/dbt_tb.c the shm_malloc function sometimes allocates memory to dtp->prev but sometimes doesn't? I'm asking this question because when the subscriber table is being loading to shared memory in db_text module, dtp->prev is allocated but on version, dispatcher, location table it doesn't... Analysing the source code I can't see any reason to this behavior. The allocation of dtp->prev to subscriber table is generating a core at db_text module. Best Regards

2 5

Siremis CDR problems
by Klaus Darilion 04 Oct '11

04 Oct '11

Hi! When populating cdrs table from timer route I get some errors: ERROR: db_mysql [km_dbase.c:120]: driver error on query: Duplicate entry '088827BCD0738B6F(a)111.11.111.11-00E0F510042319395872000014D3-6ADB' for key 2 ERROR: <core> [db_query.c:130]: error while submitting query ERROR: sqlops [sql_api.c:217]: cannot do the query Checking the acc table for those key I found the entries in the attachment. Maybe it be that the stored procedure can not handle PRACK or reINVITEs correctly? Thanks Klaus

2 3

new wiki system
by Daniel-Constantin Mierla 04 Oct '11

04 Oct '11

Hello, I installed a new wiki site, targeting to collect documentation for upcoming v3.2.0 and further releases. You can find it at: * http://www.kamailio.org/wiki/ Hopefully, it will make the process of updating/migrating the relevant documentation from old wiki (http://www.kamailio.org/dokuwiki/) straightforward and offer a clean reference for upcoming stable version. Old wiki is in place since 2005, collected a lot of pages, but many are for very old versions of kamailio, making difficult, at least for new comers, to sort out documentation for latest stable. All of you are invited to help with this process of adding documentation for 3.2.0 on the new wiki. The old wiki will stay in place, it will not be removed. Both are based on Dokuwiki, using different themes. The new one is at the beginning, so no much content in there yet, but by the time of releasing 3.2.0, should contain sufficient documentation for get started with that version. One very demanded thing was the alphabetic index for parameters, functions and mi commands from modules. This page contains new generated indexes for version 3.2.0. More details in a separate email. If you have questions about the new wiki, don't know where to place some tutorial, feel free to ask on mailing lists. Ultimately, pages can be relocated, but try to use namespaces for related topics. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

3 4

dispatcher, LCR, carrierroute or mtree?
by Skyler 04 Oct '11

04 Oct '11

Hi all, I need to setup an inbound DID router for 8 proxies and ~300 DID's. Very simple, a call comes in from pstn > kamailio looks up DID for exact match in mysql > returns location > If db result returns noservice redirect to 1.2.3.4 else remain in call-flow and send to destination. Having a failure route capability would be nice. From what I've read on each of the 4 modules, any could do what I need though each have their own complexities. I'm still new to kamailio and hoping to utilize the group here for some advice. I'd like to keep this as simple as possible then adding/deleting DID's from the db as needed for routing. In your experience which would be the best module to use in order to achieve my goal? TIA, Skyler

4 5

Trouble Setting up Kamailio
by Peter Schrock 04 Oct '11

04 Oct '11

I am using the site: http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.1.x-from-git to load up Kamailio on my ppc using Leopard X. I managed to figure out the mysql issue and got it to install, but now there are some variances to the instruction from the stated install site and I am kind of lost. If I could get some help with the following - "7. Create MySQL database" This whole section seems confusing. It doesn't say what I am suppose to look for. I am sure what it says is all you need to do, but I am not sure if it is completed correctly or not. - "8. Edit configuration file" This section could use more specific information as to how to set up the configuration file. - "9. The init.d script" This section says I am suppose to put "*kamailio.init*" in the directory "*/etc/init.d*", then I am suppose chmod after it is copied. First of, I don't have that directory, am I suppose to create it or is it suppose to be created already? Also, it says to "*adduser*" with some options. Every time I try, I get the error message that it doesn't recognize the "*adduser*" command. Please help Peter

2 1

New to Kamailio - make problem
by Peter Schrock 03 Oct '11

03 Oct '11

I can't seem to get kamailio to "*make*". I am following the directions on the wiki on how to install. I have downloaded, compiled, and installed mysql server and every time I run to "*make all*" with kamailio, I get this error message. At first, I was thinking that *km_db_mysql.c* couldn't find the file. The header seems to be looking for *mysql.h* (at *#include <mysql/mysql.h>*). I found the file from installing mysql server and I thought I'd help * km_db_mysql.c* to find the file, but it still gives me this error. I am using a power pc running OS X.5.1. CC (gcc) [M db_mysql.so] km_db_mysql.o km_db_mysql.c:49:19: error: mysql/mysql.h: No such file or directory km_db_mysql.c: In function ‘kam_mysql_mod_init’: km_db_mysql.c:92: warning: implicit declaration of function ‘mysql_get_client_info’ km_db_mysql.c:92: warning: format ‘%s’ expects type ‘char *’, but argument 7 has type ‘int’ km_db_mysql.c:92: warning: format ‘%s’ expects type ‘char *’, but argument 6 has type ‘int’ make[1]: *** [km_db_mysql.o] Error 1 make: *** [modules] Error 1 Peter Schrock

3 3

Combining Dispatcher and NAT_Helper modules.
by Henrik Aagaard Sørensen 03 Oct '11

03 Oct '11

I'm currently having a Kamailio running as load balancer, at it works most of the time. But sometimes fails due to NAT issues. This is my config: http://pastebin.com/8F8X45xh Can someone help me combine the NAT_Helper module into this config? It's not the variables to set or RTPProxy, but in the Routing Logic.

1 0

Incoming calls question
by Javier Vidal -- Quasar 03 Oct '11

03 Oct '11

i have a question, i have to make a system to recieve several calls and the same time, and depending of destination DDI i have to rewrite the URI's IP. I am not going to have Registered users, only IP validation. The question is: What is the better way to consult thousands DDIs and get the IP from mysql? I had thought, use sqlops.so module, but i think that i could have problems with it. Another posibility is use the carrierroute and use the full pattern for the DDI. Thank for all sugestions. Javier V.

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alphabetic indexes for module parameters, functions, ...
by Daniel-Constantin Mierla 03 Oct '11

03 Oct '11

Hello, for the last versions the alphabetic indexes were not generated - there were different people doing that in the past, using various self-written tools. For upcoming version, 3.2.0, I spend some time to structure the docbook files for modules in order to be easier to get from them the names for functions, parameters and mi commands. Therefore, the new wiki system has fresh generated indexes from modules' documentations. You can find them at: * http://www.kamailio.org/wiki/alphaindexes:3.2.x:modparameters * http://www.kamailio.org/wiki/alphaindexes:3.2.x:modfunctions * http://www.kamailio.org/wiki/alphaindexes:3.2.x:micommands Being generated from docbook xml files, the indexes contain name, path to the module in source tree and module name, the last one with link to html readme file of the module published to kamailio.org site. There is no link at the moment pointing inside the html readmes to the place where the item is documented, but each readme has table of content, so it is easy to spot it. We have lots lots of modules right now, therefore it was not easy to check if some parameters, functions or mi commands were not indexed. If you find some missing, please write back here so we can fix the dockbook or the indexing tool, depending where the problem is. Hope the indexes will be helpful for many of you. Ending the email with another invitation to all of you to port (update to upcoming v3.2.0) your favorite tutorial from old wiki system (http://www.kamailio.org/dokuwiki/) and add to the new wiki (http://www.kamailio.org/wiki/). Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

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Cflag is not getting updated.
by Krishna Kurapati 03 Oct '11

03 Oct '11

Hi, I am hitting bit of a wall here. - I am setting bflags for NAT, and SIPPING before save("location") however, when I do "kamctl ul show", the Cflag is 0. I also checked see if the bflags are set or not in the register routine - they are being set. What am I missing? snippets of the config files. #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_method", "INFO") modparam("nathelper", "sipping_from", "sip:pinger@50.62.1.7") # params needed for NAT traversal in other modules modparam("nathelper|registrar|usrloc", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) # Caller NAT detection route route[NAT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); setbflag(FLB_NATB); } #!endif return; } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { xlog("L_ALERT", "REGISTER: From $fU, $fd To $tU, $td"); if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging setbflag(FLB_NATSIPPING); } # 7/28/11 - make sure only one contact location is active # at any given time # unregister("location", "$rU"); if (isbflagset(FLB_NATB)) { xlog("L_ALERT", "REGISTRAR: For $fU, $fd FLB_NATB set"); } if (isbflagset(FLB_NATSIPPING)) { xlog("L_ALERT", "REGISTRAR: For $fU, $fd FLB_NATSIPPING set"); } if (!save("location")) { xlog("L_ALERT", "REGISTRAR: From $fU, $fd Location could not be saved"); sl_reply_error(); } exit; } } The debug log snippets ================== Oct 1 15:58:30 siptest /usr/sbin/kamailio[19515]: ALERT: <script>: REGISTER: From p_krishna_test_com, 50.62.1.7 To p_krishna_test_com, 50.62.1.7 Oct 1 15:58:30 siptest /usr/sbin/kamailio[19515]: ALERT: <script>: REGISTRAR: For p_krishna_test_com, 50.62.1.7 FLB_NATB set Oct 1 15:58:30 siptest /usr/sbin/kamailio[19515]: ALERT: <script>: REGISTRAR: For p_krishna_test_com, 50.62.1.7 FLB_NATSIPPING set Oct 1 15:58:30 siptest /usr/sbin/kamailio[19515]: DEBUG: sl [sl.c:278]: reply in stateless mode (sl) Output of kamctl command ==================== $ sudo kamctl ul show Domain:: location table=512 records=1 max_slot=1 AOR:: p_krishna_test_com Contact:: sip:p_krishna_test_com@75.62.17.45:50073;transport=tcp;ob Q= Expires:: 287 Callid:: ad2d3a7bf39a4345861db1661576a7f3 Cseq:: 54856 User-agent:: QliqDesktop 1.0.0 (pjproject 1.10.0/win32) State:: CS_NEW Flags:: 0 Cflag:: 0 <================= Cflag is not being set. Socket:: tcp:10.1.0.6:5060 Methods:: 8143 Thanks Krish Kura

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Kamailio as dispatcher fails due to from domain on extension.
by Henrik Aagaard Sørensen 02 Oct '11

02 Oct '11

I'm using Kamailio as dispatcher to a FreeSwitch setup. Currently it "almost" work. I have 2 extensions connected to my FreeSwitch via Kamailio. However, I can only send calls one way because Kamailio seems to change the domain on the extension. One extension is registered with ID(a)sip.my-domain.com. This I can send calls to. The other extension is registered with ID@local-ip. This I cannot send calls to. Isn't it possible to have Kamailio register the extension on my FreeSwitch with @sip.my-domain.com? If I register both extension directly to my FreeSwitch everything works.

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Re: [SR-Users] Setting avp from load_credentials
by Alex Balashov 01 Oct '11

01 Oct '11

Alternately, if you dislike any of these paradigms or find the order of operations inconvenient, just roll your own. That's what sqlops is for. -- This message was painstakingly thumbed out on my mobile, so apologies for brevity, errors, and general sloppiness. Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ On Oct 1, 2011, at 10:43 AM, "graham(a)g-rock.net" <graham(a)g-rock.net> wrote: > I know I am Auth'ing users, but apparently need to confirm the location of www_authorize. Thanks for the pointers Alex. > > Sent from my HTC on the Now Network from Sprint! > > ----- Reply message ----- > From: "Alex Balashov" <abalashov(a)evaristesys.com> > Date: Sat, Oct 1, 2011 9:07 am > Subject: [SR-Users] Setting avp from load_credentials > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -Users Mailing List" <sr-users(a)lists.sip-router.org> > > Sure it is, but credentials only get loaded if you are doing proxy/www_authorize() beforehand. Are you? > > -- > This message was painstakingly thumbed out on my mobile, so apologies for brevity, errors, and general sloppiness. > > Alex Balashov - Principal > Evariste Systems LLC > 260 Peachtree Street NW > Suite 2200 > Atlanta, GA 30303 > Tel: +1-678-954-0670 > Fax: +1-404-961-1892 > Web: http://www.evaristesys.com/ > > On Oct 1, 2011, at 9:24 AM, Graham Wooden <graham(a)g-rock.net> wrote: > > > Hi there, > > > > Just doing a quick test and this doesn’t appear to be working. Kamailio > > 1.5.5-notls. Snippet of code: > > > > modparam("auth_db", "load_credentials", "username") > > . > > . > > . > > xlog("username is $avp(s:username)\n"); > > > > > > The result is '<null>'. Is this not available in the route{} ? > > > > Thanks, > > > > -graham > > > > > > > > > > > > _______________________________________________ > > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > > sr-users(a)lists.sip-router.org > > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > >

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Re: [SR-Users] Setting avp from load_credentials
by graham＠g-rock.net 01 Oct '11

01 Oct '11

I know I am Auth'ing users, but apparently need to confirm the location of www_authorize. Thanks for the pointers Alex. Sent from my HTC on the Now Network from Sprint! ----- Reply message ----- From: "Alex Balashov" <abalashov(a)evaristesys.com> Date: Sat, Oct 1, 2011 9:07 am Subject: [SR-Users] Setting avp from load_credentials To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -Users Mailing List" <sr-users(a)lists.sip-router.org> Sure it is, but credentials only get loaded if you are doing proxy/www_authorize() beforehand. Are you? -- This message was painstakingly thumbed out on my mobile, so apologies for brevity, errors, and general sloppiness. Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ On Oct 1, 2011, at 9:24 AM, Graham Wooden <graham(a)g-rock.net> wrote: > Hi there, > > Just doing a quick test and this doesn’t appear to be working. Kamailio > 1.5.5-notls. Snippet of code: > > modparam("auth_db", "load_credentials", "username") > . > . > . > xlog("username is $avp(s:username)\n"); > > > The result is '<null>'. Is this not available in the route{} ? > > Thanks, > > -graham > > > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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Setting avp from load_credentials
by Graham Wooden 01 Oct '11

01 Oct '11

Hi there, Just doing a quick test and this doesn¹t appear to be working. Kamailio 1.5.5-notls. Snippet of code: modparam("auth_db", "load_credentials", "username") . . . xlog("username is $avp(s:username)\n"); The result is '<null>'. Is this not available in the route{} ? Thanks, -graham

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Use of $rU
by Graham Wooden 01 Oct '11

01 Oct '11

Hi there, What is pv for the equivalent of $rU for outbound? My inbound use of $rU has the correct username. All my subscribers are multi-number trunks, so on the outbound I need to associate the subscriber. This is kamailio 1.5. Thanks, -graham

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Handling quota/call limits - kamailio 1.5
by Graham Wooden 01 Oct '11

01 Oct '11

Hi there, Finally getting around to introducing some call limits and starting with the inbound calls. However, I ran into an issue about how to get the "in use" count to be updated in my quota table. What is the best way to get s:in_use to update so it can be calculated? Here is my snippet of code: if (is_method("INVITE")) { avp_db_query("select quota, in_use from quota where username='$rU'","$avp(s:quota);$avp(s:in_use)"); if ($avp(s:in_use) >= $avp(s:quota)){ xlog ("DEBUG: Trunk $rU is over quota"); sl_send_reply("503","User over quota"); } . . } This opens up two more questions: How to decrease the call count when a BYE is received - maybe do a subtraction and the do a avp_db_query to update the table? And of course there is the BYE that won't be present (ie. When a call just rings and hangs up before it gets answered). Maybe ACC's missed_calls can somewhat be used to help determine? Thanks for any pointers and guidance. I am running Kamailio 1.5. Thanks, -graham

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SNMPStats and multiple proxies
by Antanas Masevicius 30 Sep '11

30 Sep '11

Hello, been trying to run two sip-router proxies in parallel and to feed data from them to snmpd via agentx. But it looks like snmpd deamon directly routes snmp request just to one (probably first connected) proxy. What ways this can be solved? Currenly i am thinking like running two concurrent snmpd daemons binded to specific IP addresses. Is there a better way to do it? best regards, -- Antanas Masevicius

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Re: [SR-Users] Making iPhone SIP client work with Kamailio
by Krishna Kurapati 30 Sep '11

30 Sep '11

> > Hi, > > The iPhone client, when it goes into background, can only respond to > incoming messages. > That makes the Registration expire when there is no communication from the > server before > the expire time. I could make the Registration not expire at all. Not sure > it's the only option. > > Is there a way to send a Ping message from server periodically to the > iPhone SIP client. This > will wakeup the SIP client and resend REGISTER message. > > Thanks > Krish Kura >

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Cleaning up presence info when user is deleted
by Krishna Kurapati 30 Sep '11

30 Sep '11

Hi, When a user is removed using "kamctl rm", is there way way to clean up presence information in the database (xcap, presentity, watcher and active watcher tables) and in the cache? Is there a event that gets fired when the user is removed from subscriber table which I could handle in config file and and cleanup? Thanks Krish Kura

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Core db_text module
by Bruno Bresciani 30 Sep '11

30 Sep '11

Hi All, Kamailio 3.1.2 generate a core in db_text module when subscriber table is updated and after this action some SIP message require authentication process... Someone Can Help me to understand why this core is happening? Below is part of kamailio's trace and gdb core too. Kamailio's trace: *Sep 22 16:35:54 sswpst00 /home2/local/kamailio/sbin/kamailio[2327]: DEBUG: auth_db [authorize.c:239]: realm value [192.168.166.199]* Sep 22 16:35:54 sswpst00 /home2/local/kamailio/sbin/kamailio[2327]: DEBUG: auth [api.c:95]: auth: digest-algo: MD5 parsed value: 1 *Sep 22 16:35:54 sswpst00 /home2/local/kamailio/sbin/kamailio[2327]: DEBUG: db_text [dbt_file.c:78]: [subscriber] was updated* Sep 22 16:35:54 sswpst00 /home2/local/kamailio/sbin/kamailio[2324]: ALERT: <core> [main.c:741]: child process 2327 exited by a signal 11 Sep 22 16:35:54 sswpst00 /home2/local/kamailio/sbin/kamailio[2324]: ALERT: <core> [main.c:744]: core was generated Sep 22 16:35:54 sswpst00 /home2/local/kamailio/sbin/kamailio[2324]: INFO: <core> [main.c:756]: INFO: terminating due to SIGCHLD Sep 22 16:35:54 sswpst00 /home2/local/kamailio/sbin/kamailio[2335]: INFO: <core> [main.c:807]: INFO: signal 15 received c Sep 22 16:35:54 sswpst00 /home2/local/kamailio/sbin/kamailio[2335]: DEBUG: <core> [main.c:818]: Memory status (pkg): gdb core trace: Core was generated by `/home2/local/kamailio/sbin/kamailio -P /var/run/kamailio.pid'. Program terminated with signal 11, Segmentation fault. *#0 0x00f21603 in dbt_db_del_table (_dc=0xb6071e60, _s=0xbf837040, sync=0) at dbt_lib.c:238* 238 dbt_lib.c: Arquivo ou diretório não encontrado. in dbt_lib.c Best Regards

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Re: [SR-Users] simple question but can't find the answer
by Daniel-Constantin Mierla 30 Sep '11

30 Sep '11

Hello, On 9/30/11 12:11 AM, JJ Zeng wrote: > > Thanks Daniel. I followed your advice and install kamailio 3.1. Now > I get a weird problem: it works 1 out 5 times for the relaying. > try to set like tcp_connection_lifetime=3604 global parameter, or to a value higher than the longer registration time. > > The t_relay is where the the problem is, because I check the response > code and route(LOCATION) always succeeds, which means the entries > exist in "location". > > If I don't put set_forward_no_connect, then t_relay will take a long > time until my client time out. If I put it, then the client will get > 477 right away. > > BTW, the original config didn't work, either. That's why I started to > strip down some irrelevant stuff. > What errors you got with the default config? Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Dec 5-8, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

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Re: [SR-Users] Placing ds_select_dst in kamailio.cfg, where?
by Daniel-Constantin Mierla 29 Sep '11

29 Sep '11

Hello, please keep the mainling list cc-ed. Calling ds_select_dst() first in the main route block is not really a good place. First you have to handle sanity checks, within dialog requests a.s.o. Of course it is a matter of your purpose, but as general hint, selecting the destination should be done just (or as close as possible) before relaying. If you take default config and you have only load balancing, then you can replace location lookup part with load balancing. Other parts may be stripped as well, just giving some direction here. Cheers, Daniel On 9/28/11 11:11 PM, Henrik Aagaard Sørensen wrote: > That did the job. > > I just placed the ds_select_dst-code at the very first thing after route{ > > Thank you for your help! > > 2011/9/28 Daniel-Constantin Mierla <miconda(a)gmail.com > <mailto:miconda@gmail.com>> > > Hello, > > the devel version of dispatcher module README has a different > config example, maybe it helps you more: > > http://kamailio.org/docs/modules/devel/modules_k/dispatcher.html#id2522847 > > I am considering to update it again soon to be a bit more complete > and follow the structure of the default config file shipped with > kamailio. > > Cheers, > Daniel > > > On 9/28/11 6:30 PM, Henrik Aagaard Sørensen wrote: >> Hi Carsten. >> >> The configuration works. It's actually where (which line) in the >> configuration I should put: >> ds_select_dst("1", "0"); >> >> :) I'm not strong in Kamailio and just want it to load balance >> between Freswitch servers. So I have a basic installation of >> Kamailio, without MySQL, NAT, Auth etc. >> >> On Wed, Sep 28, 2011 at 5:09 PM, Carsten Bock >> <carsten(a)ng-voice.com <mailto:carsten@ng-voice.com>> wrote: >> >> Hi, >> >> When you've got problems with placing "ds_select_dst()" in your >> config, you should do the following: >> - Open an Editor of your choice >> - go to the line, where you want to place it >> - type: ds_select_dst("1", "0"); >> - save & exit >> >> Just kidding! You should do the following with your config: >> kamailio -f <path-to-your-config> -c >> >> What error is displayed? If none, then check your syslog for >> errors. >> Could you post your config? It is better to put a full path >> in the >> config to the dispatcher-file... otherwise it might be >> dependent of, >> where you are running the init.d-skript/kamailio.... >> >> I am using dispatcher in Kamailio 1.5, 3.1 and 3.2, >> everythings works >> fine with me. >> >> Kind regards, >> Carsten >> >> 2011/9/28 Henrik Aagaard Sørensen >> <henrikaagaardsorensen(a)gmail.com >> <mailto:henrikaagaardsorensen@gmail.com>>: >> > I've installed 2 Freeswitch on different servers and would >> like to have >> > Kamailio load balance between them. >> > >> > I've installed Kamailio following: >> > >> http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.1.x-from-git >> > >> > I've created dispatcher.list and added: >> > >> > loadmodule "dispatcher.so" >> > >> > modparam("dispatcher", "list_file", "../etc/dispatcher.list") >> > >> > in the kamailio.cfg. >> > >> > But looking at: >> > >> http://kamailio.org/docs/modules/3.1.x/modules_k/dispatcher.html >> > I'm having difficulties to placing: >> > >> > ds_select_dst("1", "0"); >> > >> > in the kamailio.cfg that was created originally. >> > >> > Can anyone help me? >> > >> > _______________________________________________ >> > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users >> mailing list >> > sr-users(a)lists.sip-router.org >> <mailto:sr-users@lists.sip-router.org> >> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> > >> > >> >> >> >> -- >> Carsten Bock >> CEO (Geschäftsführer) >> >> ng-voice GmbH i. Gr. >> Schomburgstr. 80 >> D-22767 Hamburg / Germany >> >> http://www.ng-voice.com >> mailto:carsten@ng-voice.com <mailto:carsten@ng-voice.com> >> >> Mobile +49 179 2021244 <tel:%2B49%20179%202021244> >> Office +49 40 34927219 <tel:%2B49%2040%2034927219> >> Fax +49 40 34927220 <tel:%2B49%2040%2034927220> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users >> mailing list >> sr-users(a)lists.sip-router.org >> <mailto:sr-users@lists.sip-router.org> >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > -- > Daniel-Constantin Mierla --http://www.asipto.com > Kamailio Advanced Training, Oct 10-13, Berlin:http://asipto.com/u/kat > http://linkedin.com/in/miconda -- http://twitter.com/miconda > > -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

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Re: [SR-Users] Logmessage when Dispatcher-Module sets a dispatcher to probing
by Timo Klecker 29 Sep '11

29 Sep '11

Hey again, Btw: I am using Kamailio 3.1.5 and found the event_route in 3.2.x Dispatcher documentation. Is there a possibility in 3.1.x? Greetings Timo Von: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] Im Auftrag von Timo Klecker Gesendet: Donnerstag, 29. September 2011 11:10 An: 'kamailio users' Betreff: [SR-Users] Logmessage when Dispatcher-Module sets a dispatcher to probing Hey, I'd like to have a log-entry in the kamailio logfile when #ds_probind_threshhold OPTIONS pings fail to reach a dispatcher (408 localy generated after #fr_timer ms). Is it possible to do this without changing code? Greetings Timo

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simple question but can't find the answer
by JJ Zeng 29 Sep '11

29 Sep '11

Hi, I am really new to ser(just installed it today). My situation is here: Both A and B make tcp connection to ser proxy server. If A sends "Invite" to B, ser would look up the tcp connection of B, and relay the sip request. If B is not connected, then 480 is returned to A. When B sends back the response, ser would relay it to A. It looks really simple, but I just can't figure out how to do it. Thanks. JJ

2 3

Logmessage when Dispatcher-Module sets a dispatcher to probing
by Timo Klecker 29 Sep '11

29 Sep '11

Hey, I'd like to have a log-entry in the kamailio logfile when #ds_probind_threshhold OPTIONS pings fail to reach a dispatcher (408 localy generated after #fr_timer ms). Is it possible to do this without changing code? Greetings Timo

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OpenIMS
by Carlos Ruiz Diaz 29 Sep '11

29 Sep '11

Hello list. I'm trying to test/implement http://www.openimscore.org/ to setup an IMS scenario for development purposes. The project seems quite inactive to me and I wonder if there is any other similar project or this one just died and is no longer maintained. The documentation didn't specify if there is any kind of compatibility with Kamailio and based on what was written, only SER core architecture is supported. Any help will be appreciated. Regards. Carlos.

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documentation update for 3.2.0 and wiki system
by Daniel-Constantin Mierla 28 Sep '11

28 Sep '11

Hello, in preparation for release 3.2.0, I was checking our wiki (on kamailio.org), which is quite rich in information, but some tutorials are for older versions. Maybe it is also a good time to do some restructuring along with the update of the useful tutorials to be valid for 3.2.0. For restructuring, the main goal is to provide some ways to let people find the documentation they need (and it is available there) easier. Some ideas, introducing categorization/tags for specific topics (e.g., radius, ldap, presence, asterisk, ...), introducing some index pages to link tutorials per version, etc. More suggestions are welcome here, let us know what will make your life easier in browsing the docs. On another hand, currently wiki is based on dokuwiki, which from my point of view did good job so far, but maybe we are missing another excellent wiki system that we should use, because provides more tools for this kind of purpose. One thing I was missing, was export to pdf, which I may be possible right now with dokuwiki plugins -- or even better, we can get something to export to docbook xml format with dokuwiki or other wiki system. In the same kind of subject, what about using git to store (or at least have a replica of) the wiki pages? IIRC, I saw somewhere that even for dokuwiki is possible to configure to use a git repository to store the wiki content instead of local CVS. This will allow people to mirror very easy the content. I could spot from time to time web spiders downloading the wiki. Comments or other suggestions to make the access to documentation easier? Thanks, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

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Placing ds_select_dst in kamailio.cfg, where?
by Henrik Aagaard Sørensen 28 Sep '11

28 Sep '11

I've installed 2 Freeswitch on different servers and would like to have Kamailio load balance between them. I've installed Kamailio following: http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.1.x-from-git I've created dispatcher.list and added: loadmodule "dispatcher.so" modparam("dispatcher", "list_file", "../etc/dispatcher.list") in the kamailio.cfg. But looking at: http://kamailio.org/docs/modules/3.1.x/modules_k/dispatcher.html I'm having difficulties to placing: ds_select_dst("1", "0"); in the kamailio.cfg that was created originally. Can anyone help me?

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Sip-router gentoo ebuild v3.1.5
by caio 28 Sep '11

28 Sep '11

Hello, Just want to announce the availability of an ebuild for version 3.1.5 (kamailio and ser flavours) on Gentoo Portage Overlays. Here the git link: * http://git.overlays.gentoo.org/gitweb/?p=user/caio.git;a=tree;f=net-misc/si… The unofficial overlays portage website: * http://gpo.zugaina.org/net-misc/sip-router Important, this is not an official ebuild, and neither it's not at official gentoo portage tree. You can use *layman* to add this overlay. Also at pkg/ser/gentoo path of sip-router project you can find the latest version. If you need any guidance to probe this ebuild under your gentoo linux do not hesitate to contact me or provide some feedback here at the lists. Best regards, Claudio

2 2

Accounting only the 2nd branch of missed serial forked call
by Ozren Lapcevic 27 Sep '11

27 Sep '11

Hi, I'm having some problems accounting missed serial forked calls to mysql database. I have following setup. Each user can have up to two contacts: telephone number (routed to asterisk) and SIP URI. Users can specify which contact has higher priority - which one should ring first. There is also SEMS voicemail which is forked as 3rd serial call leg if there is no answer at first two contacts. For example, I have two users: oz(a)abc.hr and pero(a)abc.hr. pero(a)abc.hr also has set telephone number as alternative number if he is not reachable at sip:pero@abc.hr. Moreover, pero(a)abc.hr has voicemail turned on. When oz(a)abc.hr calls pero(a)abc.hr, first pero(a)abc.hr's SIP client rings, then if there is no answer and after the timeout telephone number rings and finally, if there is no answer at telephone and after the timeout INVITE is forked to SEMS. There are two interesting scenarios accounting-wise which can happened: 1. oz(a)abc.hr calls pero(a)abc.hr, there are no answers and call is forked to voicemail. 2. oz(a)abc.hr calls pero(a)abc.hr, there is no answer at SIP client, but pero answers call at telephone. When scenario 1 happens, I want to have only one log (row) in missed_calls table. When scenario 2 happens, I don't want to have a log in missed_calls table. To accomplish this,* I want to log only the 2nd branch of the forked call. However, there is either a bug in acc module or I'm doing something wrong, and I can't get Kamailio to log only the 2nd branch*. I think that I am setting the FLT_ACCMISSED flag correctly - after the 2nd branch is handled and prior to calling the RELAY route. Logs show that FLT_ACCMISSED flag is set prior to calling t_relay(), and there are no errors in debug log. I am using $ru = "something" to rewrite URI prior to forking request. I can easily set up logging of every call (two missed calls for serially forked call to two locations) by setting FLT_ACCMISSED flag for each INVITE. I can set up logging of every call's 1st branch, by reseting FLT_ACCMISSED flag when handling 2nd branch of the call. Interestingly, logging of only the 2nd branch of the serial forked call works when there is no forking to voicemail! Any ideas how to solve this problem? Bellow are important parts of my config file. I'm running kamailio 3.1.4. Cheers Ozren # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 0) modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) modparam("acc", "log_extra","src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;dst_user=$tU;dst_domain=$td;src_ip=$si") #!endif ... # Main SIP request routing logic # - processing of any incoming SIP request starts with this route route { # per request initial checks route(REQINIT); if (src_ip != ****) { # NAT detection route(NAT); } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); if ( is_method("INVITE") ) { route(DBALIASES); #check for user defined forking priorities and timers route(FORK); } # user location service route(LOCATION); route(RELAY); } #check for user defined forking priorities and timers route[FORK]{ sql_query("con", "select * from usr_pref_custom where uuid='$tu'", "pref"); $avp(uuid)=$dbr(pref=>[0,0]); $avp(email)=$dbr(pref=>[0,1]); $avp(prio1)=$dbr(pref=>[0,2]); $avp(prio2)=$dbr(pref=>[0,3]); $avp(timer1)=$dbr(pref=>[0,5]); $avp(timer2)=$dbr(pref=>[0,6]); if (strlen($avp(prio1))>5) { # user has multiple contacts, do serial forking setflag(FLT_USRPREF); # set counter if (!$avp(prio)) { $avp(prio) = 1; } # overwrite request URI with highest priority contact if ($avp(prio1) =~ "^sip:00") { $ru = $avp(prio1) + "@host"; xlog("L_INFO","PRIO 1 is tel number, RURI set: $ru"); } else { $ru = $avp(prio1); xlog("L_INFO","PRIO 1 is SIP URI, RURI set: $ru"); } } } route[RELAY] { #!ifdef WITH_NAT if (check_route_param("nat=yes")) { setbflag(FLB_NATB); } if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) { route(RTPPROXY); } #!endif /* example how to enable some additional event routes */ if (is_method("INVITE")) { t_on_reply("REPLY_ONE"); t_on_failure("FAIL_ONE"); #if users have priorities set, use FAIL_FORK failure route if ( isflagset(FLT_USRPREF) ) { t_on_failure("FAIL_FORK"); } } if (isflagset(FLT_ACCMISSED)) xlog("L_INFO","RELAY, $rm $ru, ACCMISSED FLAG IS SET"); else xlog("L_INFO","RELAY, $rm $ru, ACCMISSED FLAG IS NOT SET"); if (!t_relay()) { sl_reply_error(); } 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()) { xlog("L_INFO","WITHINDLG, loose_route()"); if (is_method("BYE")) { xlog("L_INFO","WITHINDLG, BYE, DO ACCOUNTING"); setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... 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() ) { # 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; } } # USER location service route[LOCATION] { #skip if $ru is telephone number if ($ru =~ "^sip:00") { xlog("L_INFO","SKIP lookup..."); } else { if (!lookup("location")) { switch ($rc) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } } # when routing via usrloc, log the missed calls also, but only if user doesn't have prios set if ( is_method("INVITE") && !(isflagset(FLT_USRPREF))) { setflag(FLT_ACCMISSED); } } # Failure route for forked calls failure_route[FAIL_FORK] { #!ifdef WITH_NAT if (is_method("INVITE") && (isbflagset(FLB_NATB) || isflagset(FLT_NATS))) { unforce_rtp_proxy(); } #!endif if ($avp(prio) >= 1) { $avp(prio) = $avp(prio) + 1; # handle 2nd branch if ( ($avp(prio) == 2) && ( isflagset(FLT_USRPREF) )) { t_on_failure("FAIL_FORK"); if ($avp(prio2) =~ "^sip:00") { xlog("L_INFO","FAIL FORK, PRIO 2 is tel number"); $ru = $avp(prio2) + "@host"; } else { xlog("L_INFO","FAIL FORK, PRIO 2 is SIP URI"); $ru = $avp(prio2); route(LOCATION); } setflag(FLT_ACCMISSED); } else { $avp(prio) = 0; $ru = $(avp(uuid)); rewritehostport("host:port"); xlog("L_INFO","FAIL FORK, VOICEMAIL email:$avp(email), ru:$ru, br: $br"); append_hf("P-App-Name: voicemail\r\n"); append_hf("P-App-Param: Email-Address=$avp(email)\r\n"); } route(RELAY); } if (t_is_canceled()) { exit; } }

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Kamailio with IMS extensions build
by loic.trottin＠orange.com 27 Sep '11

27 Sep '11

Hi all, I try to compile Kamailio with IMS extensions from OpenIMSCore available in carstenbock/ims branch in the GIT repository. I have some problems with the modules "Client_Ro" and "pcscf" that do not compile: CC (gcc) [M Client_Ro.so] ims_ro.o ims_ro.c: In function 'send_ccr_interim': ims_ro.c:594: error: 'AVP_EPC_User_Equipment_Info_Type_MAC' undeclared (first use in this function) ims_ro.c:594: error: (Each undeclared identifier is reported only once ims_ro.c:594: error: for each function it appears in.) ims_ro.c: In function 'send_ccr_stop': ims_ro.c:749: error: 'AVP_EPC_User_Equipment_Info_Type_MAC' undeclared (first use in this function) ims_ro.c: In function 'Ro_Send_CCR': ims_ro.c:831: error: 'struct dlg_binds' has no member named 'get_dlg' ims_ro.c:835: error: 'struct dlg_binds' has no member named 'unref_dlg' ims_ro.c:886: error: 'AVP_EPC_User_Equipment_Info_Type_MAC' undeclared (first use in this function) CC (gcc) [M pcscf.so] pcc_avp.o pcc_avp.c:1459: error: 'AVP_IMS_Subscription_Id_Type_E164' undeclared (first use in this function) pcc_avp.c:1459: error: (Each undeclared identifier is reported only once pcc_avp.c:1459: error: for each function it appears in.) pcc_avp.c:1461: error: 'AVP_IMS_Subscription_Id_Type_SIP_URI' undeclared (first use in this function) Does anybody have encountered these problems? Is there a stable version tagged of the branch carstenbock/ims available in the GIT repository ? Best regards, Loïc. ******************************************************************************** IMPORTANT.Les informations contenues dans ce message electronique y compris les fichiers attaches sont strictement confidentielles et peuvent etre protegees par la loi. Ce message electronique est destine exclusivement au(x) destinataire(s) mentionne(s) ci-dessus. Si vous avez recu ce message par erreur ou s il ne vous est pas destine, veuillez immediatement le signaler a l expediteur et effacer ce message et tous les fichiers eventuellement attaches. Toute lecture, exploitation ou transmission des informations contenues dans ce message est interdite. Tout message electronique est susceptible d alteration. A ce titre, le Groupe France Telecom decline toute responsabilite notamment s il a ete altere, deforme ou falsifie. De meme, il appartient au destinataire de s assurer de l absence de tout virus. IMPORTANT.This e-mail message and any attachments are strictly confidential and may be protected by law. This message is intended only for the named recipient(s) above. If you have received this message in error, or are not the named recipient(s), please immediately notify the sender and delete this e-mail message. Any unauthorized view, usage or disclosure ofthis message is prohibited. Since e-mail messages may not be reliable, France Telecom Group shall not be liable for any message if modified, changed or falsified. Additionally the recipient should ensure they are actually virus free. ********************************************************************************

3 3

need little help to setting up pike module
by dirtytoni＠gmx.de 27 Sep '11

27 Sep '11

Hi to all on this list =) jUSt a tiny user question aboUt pike module(at least kamailio doc). are these values for sampling_time_unit seconds or minutes ? it's not in the doc. same 4 remove_latency. thanks for helping me out sorry to bother you best regards Toni -- Empfehlen Sie GMX DSL Ihren Freunden und Bekannten und wir belohnen Sie mit bis zu 50,- Euro! https://freundschaftswerbung.gmx.de

2 1

Account 403 to RADIUS
by Mino Haluz 26 Sep '11

26 Sep '11

Hi, I have this code: if ( is_user_in("From", "blocked") && is_method("INVITE")) { xlog("L_INFO", "XLOG: [number_and_ruri_checks] NOTICE: Account ($fu) to ($ru) is blocked"); sl_send_reply("403", " Account blocked "); exit; } I would like to account also this answer to Radius, now I have it only in application xlog. Thanks, Mino

3 5

Migration from v3.1.x to 3.2.0
by Daniel-Constantin Mierla 26 Sep '11

26 Sep '11

Hello, a kind reminder for everyone to add to the wiki page hints for migration from v3.1.x to v3.2.0. Especially to developers that know what kind of major changes they did to existing modules in 3.1.x. The wiki is open for anyone: * http://sip-router.org/wiki/install/3.1.x-to-3.2.x I just updated the section related to database structure changes. If you spot errors/missing parts, go ahead and fix it. Thanks, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

Re: [SR-Users] I am getting this error in the log When REGISTER message is received from jitsi
by Daniel-Constantin Mierla 23 Sep '11

23 Sep '11

On 9/22/11 9:04 PM, Krishna Kurapati wrote: > Thanks Daniel, > > Here it is... > > [...] It looked ok for my eye. Can you run kamailio with higher debug level (debug=4 in config file) and send the log messages related to parsing the URI for AoR? Cheers, Daniel > > > On Thu, Sep 22, 2011 at 1:44 PM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > Hello, > > can you grab such REGISTER with ngrep (or wireshark) and past it here? > > Cheers, > Daniel > > > On 9/22/11 8:07 PM, Krishna Kurapati wrote: >> Sep 22 09:49:53 siptest /usr/sbin/kamailio[3160]: ERROR: >> registrar [common.c:75]: failed to parse Address of Record >> Sep 22 09:49:53 siptest /usr/sbin/kamailio[3160]: ERROR: >> registrar [save.c:822]: failed to extract Address Of Record >> >> Is there a problem when AoR contains "_" >> >> for example is AoR krish_kura(a)sip.org <mailto:krish_kura@sip.org> >> is not allowed? >> >> Thanks >> Krish Kura >> >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > -- > Daniel-Constantin Mierla --http://www.asipto.com > Kamailio Advanced Training, Oct 10-13, Berlin:http://asipto.com/u/kat > http://linkedin.com/in/miconda -- http://twitter.com/miconda > > -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

2 4

Re: [SR-Users] CANCEL is not processed in onsend_route
by Daniel-Constantin Mierla 23 Sep '11

23 Sep '11

Hello, On 9/15/11 4:02 PM, Timo Klecker wrote: > Hi Daniel, > > I managed to get a log with debug 4 from one call. I had to take out all > IP-Addresses and Usernames, hope you can use it. the log confirmed what I expected: Sep 15 15:32:33 vux896 /sbin/kamailio[31594]: DEBUG: siptrace [siptrace.c:954]: no uas msg, local transaction Sep 15 15:32:33 vux896 /sbin/kamailio[31594]: DEBUG: tm [t_fwd.c:1242]: DEBUG: e2e_cancel: e2e cancel proceeding Because a new CANCEL is generated from scratch, incoming message is lost and no longer available to check the sip trace flag. Now, the question is how to decide whether the cancel is to be siptraced. I can lookup for incoming CANCEL and check if it has the siptrace flag set, but there are CANCELs in case of local timeout or parallel forking, so I think it is better to actually lookup the INVITE transaction and if the siptrace flag is set for it, record the cancel. If there is no INVITE transaction anymore, the cancel is forwarded stateless and can be captured in onsend route, although it should be normally discarded if it is known that the proxy works in statefull mode only. Any other opinions? Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

2 2

carrierroute module
by Mino Haluz 23 Sep '11

23 Sep '11

Hi, I have a problem with configuring the carrierroute module, I do not clearly understand how it should work from the documentation. What I would like to do is that I have 2 gateways (A.A.A.A and B.B.B.B), lots of scan_prefixes with different rewrite prefixes. The traffic should be sent to the 1 gateway, and if it fails (5xx,4xx codes) it should go to the another. I have only one SIP domain (domain.com). The question is, how the tables (domain_name, carrierroute, carrierfailroute) should be filled? Can I achieve this scenario with the carrierroute module ? Thanks. Mino

1 0

XCAP over SIP
by Krishna Kurapati 23 Sep '11

23 Sep '11

Hi, The document http://kb.asipto.com/kamailio:presence:k31-made-simple describes that XCAP can be sent over SIP instead of over HTTP. I tried to do however, the server is unable to update presence rules. I was using PUBLISH method. I was using content-type = text... Has anyone used SIP to send over XCAP? Appreciate if someone could share a link or configuration changes required. Thanks Krish

3 3

I am getting this error in the log When REGISTER message is received from jitsi
by Krishna Kurapati 22 Sep '11

22 Sep '11

Sep 22 09:49:53 siptest /usr/sbin/kamailio[3160]: ERROR: registrar [common.c:75]: failed to parse Address of Record Sep 22 09:49:53 siptest /usr/sbin/kamailio[3160]: ERROR: registrar [save.c:822]: failed to extract Address Of Record Is there a problem when AoR contains "_" for example is AoR krish_kura(a)sip.org is not allowed? Thanks Krish Kura

2 1

perl module compilation on centos
by Daniel-Constantin Mierla 22 Sep '11

22 Sep '11

Hello, while building the v3.1.5 RPMs, the perl module failed to compile on centos 5 and 6, with following error: error: `PL_unitcheckav' undeclared (first use in this function ) so its packaging was disabled. It works fine on opensuse and fedora. Someone on twitter tried with the package perl-ExtUtils-Embed (like in fedora) and worked. However, OpenSUSE Build System fails to find that package. Is anyone here with latest centos 5 or 6 that can try compilation of perl module in v3.1.5 and let me know the result? Few months ago, it didn't fail for v3.1.4. Thanks, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

4 5

Re: [SR-Users] Kamailio and FreeSWITCH realtime integration, tutorial?
by Henrik Aagaard Sørensen 22 Sep '11

22 Sep '11

But FreeSwitch don't use MySQL like Asterisk? So adapting isn't that easy or...??? On 21/09/2011, at 16.18, David <david(a)styleflare.com> wrote: You can adapt pretty easily. On 9/21/11 3:46 PM, Henrik Aagaard Sørensen wrote: Does anyone know if there somewhere exists a tutorial about Kamailio and FreeSWITCH realtime integration? I have Googled a lot and found: http://kb.asipto.com/kamailio:index and http://kb.asipto.com/freeswitch:index On the same site there is a tutorial for Kamailio and Asterisk realtime integration, but not for FreeSWITCH. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

2 2

Re: [SR-Users] Problem with REGISTER retransmissions
by Daniel-Constantin Mierla 22 Sep '11

22 Sep '11

Hello, I tried to reproduce with a very simple config to handle register requests. So I sent a REGISTER with expires 60 (very much the same as the one in the trace you sent), waited a while, then sent two REGISTERs having the CSeq incremented. All went fine. I tested with master and 3.1 branches (so the last is actually 3.1.5). Can you upgrade to 3.1.5 first and try again? If still fails, then update config to print the return code for save() function, something like: $var(r) = save("location"); xlog("code returned by save is $var(r)\n"); if($var(r) < 0) send_reply("500", "Failure"); Let me know the value of the returned code for the register that does not have a contact in reply. Cheers, Daniel On 9/20/11 2:10 PM, Alejandro Mingo wrote: > > Hi Daniel-Constantin, > > I send you the PCAP and LOG files corresponding to one of these > situations. As you can see in the LOG file, one thread (5444) handles > the first REGISTER, other thread (5447) handles the second REGISTER > (retransmission) and a third thread (5451) indicates that the binding > has expired, six seconds after the arrival of both REGISTERs. > > Thanks again. > > Alejandro. > > > > El 20/09/2011 10:54, Daniel-Constantin Mierla escribió: >> Hello, >> >> On 9/20/11 10:13 AM, Alejandro Mingo wrote: >>> Hi, >>> >>> Since we installed last version of Kamailio (3.1.4) >> >> the last version is now 3.1.5, but this is not really the most >> important aspect. However, it is recommended to upgrade, there were >> some fixed to registration handling when register requests have >> multiple contacts. >>> we have been experiencing a big problem with REGISTER >>> retransmissions. When the server receives a retransmitted REGISTER >>> it removes the binding and the UAC remains unregistered until next >>> refreshing period. I include a PCAP capture that shows this behaviour. >>> >>> I am not 100% sure that this problem never happened with older >>> versions. >>> >>> Has anybody experienced the same problem? >> I will try to reproduce and investigate myself. Anyhow, meanwhile >> maybe you can provide more hints faster. Do you have error messages >> in the syslog? Can you run with debug=4 in config file and send the >> output in syslog for such a case? -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

2 3

Kamailio 3.1.5 RPMS
by Daniel-Constantin Mierla 22 Sep '11

22 Sep '11

Hello, just to let you know that the RPMs of Kamailio 3.1.5 were generated for CentOS, Fedora, OpenSUSE and RedHat. This time CentOS RPMs are packaged for both versions 5 and 6. Links are available on our wiki at: * http://www.kamailio.org/dokuwiki/doku.php/packages:rpms Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

Kamailio and FreeSWITCH realtime integration, tutorial?
by Henrik Aagaard Sørensen 22 Sep '11

22 Sep '11

Does anyone know if there somewhere exists a tutorial about Kamailio and FreeSWITCH realtime integration? I have Googled a lot and found: http://kb.asipto.com/kamailio:index and http://kb.asipto.com/freeswitch:index On the same site there is a tutorial for Kamailio and Asterisk realtime integration, but not for FreeSWITCH.

4 4

Re: [SR-Users] Dialog module showing calls that have already been terminated
by Timo Reimann 21 Sep '11

21 Sep '11

Hey Phillip, On 20.09.2011 13:48, Phillman25 Kyriacou wrote: > Thanks for your email. > > Yes dlg_manage(); has to now be called on INVITE and BYE/CANCEL messages. > Where would i have to call loose_route()? Only on INVITE? On *all* in-dialog requests, i.e., all requests which contain a To tag. This may include Re-INVITEs too (but not initial INVITEs). > My configuration did not change between 3.1.2 and 3.1.5. > > Call flow example: > ============== > > Cisco PGW ===> Kamailio 3.1.5 ===> VOIP PROVIDER or ASTERISK PABX > > > The below is my configuration. Let's take a look at it: > #!KAMAILIO > # [...] > if(is_method("BYE|CANCEL")) > > { > > dlg_manage(); > > > > # per request initial checks > route(REQINIT); > > # NAT detection > route(NAT); > > # handle requests within SIP dialogs > route(WITHINDLG); [...] > # 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()) { [...] So route[WITHINDLG] contains the logic to track in-dialog requests. However, you seem to call that route only from BYE and CANCEL requests, missing out ACKs, which is likely the reason why things go wrong. So make sure you run all in-dialog requests through that route, and you should be fine (hopefully). Cheers, --Timo

2 8

Kamailio Development Training, San Francisco Area, October 2011
by Daniel-Constantin Mierla 21 Sep '11

21 Sep '11

Hello, since I am attending GSoC'11 Mentors Summit and come to Bay Area again, it is time to bring the developer training in USA, therefore I am planning to organize a two-day seminar focused on developing Kamailio extensions in C. Among topics to be approached: - internal architecture - SIP parser - memory manager - locking manager - database API - config file language interpreter - RPC interface - pseudo-variables and transformations framework - module interface – write your own extensions in C as modules - inter-module APIs - tm, presence, a.s.o. - documentation docbook format This kind of training follows the pattern from the past, trying to keep a low attendee fee, targeting just to cover the expenses with the organization. I am discussing with several people for a good location at a convenient price, however, I am open for new alternatives, including companies/institutes that want to host the event at no cost in exchange of sending few attendees free of charge. Contact me if you can help here. Training is planned for October 24-25, right now the estimated fee per participant is 350USD (it will not exceed 400USD). The policy to accept people is first come, first served. You can register at this time to reserve your seat(s), emailing to me or to registration(a)kamailio.org. Number of seats is usually 10 to 20, last one organized in February in Barcelona filled up quickly. Once there will be enough interested people and a location selected, I will have more precise details regarding the participation to this seminar and I will send updates. Looking forward to meeting many of you and get new US developers on board of the project in the near future. Cheers, Daniel -- http://www.kamailio.org/w/daniel-constantin-mierla/

1 0

how to remove media description from sdp?
by Juha Heinanen 21 Sep '11

21 Sep '11

trying to hack around the possible fix_nated_sdp("1") issue that i outlined in previous message, i thought to remove video media description from sdp. in sdpops module there is sdp_with_media(type), but for some reason no sdp_remove_media(type) function. is there some other means to get rid of sdp video media description? -- juha

3 3

Digest authentication
by Morten Isaksen 20 Sep '11

20 Sep '11

Hi, Sorry for the slightly off topic question, but I am trying to debug a buggy PBX and want to check if the digest information is correct. I have this perl script. #!/usr/bin/perl use strict; use Digest::MD5 qw(md5 md5_hex md5_base64); my $uri = 'sip:sip1.uni-tel.dk'; my $method = "SIP/2.0"; my $username = "69102208"; my $password = "password"; my $nonce = "4e78d2b0a72196d9123be7c88d093bd74c77f35e"; my $realm = "sip1.uni-tel.dk"; my $a1 = "$username:$realm:$password"; my $ha1 = md5_hex($a1); my $a2 = "$method:$uri"; my $ha2 = md5_hex($a2); print "A1 : $a1\n"; print "A2 : $a2\n"; my $response = md5_hex("$ha1:$nonce:$ha2"); print "\nRES : $response\n\n"; I registered with a softphone and entered the above values from the Authentication header, but I cannot get the response to match. Any help would be appreciated. -- Morten Isaksen

2 2

wt_timer, delayed replies and sendto error
by Andreas Granig 20 Sep '11

20 Sep '11

Hi, Another funny thing I encountered lately is this: Client sends INVITE via kamailio (via dispatcher) to sems, gets back 100/200, sends another couple of INVITE with same CSeq (looks like retransmissions), gets back more 200, then sends ACKs for each of the 200 it got. Looks like a quite broken client to me, but anyways. The interesting thing is that after >5000ms (which is bigger then tm.wt_timer), another 200 is sent from sems to kamailio to the client. I see the 200 coming to kamailio in a network trace, but not an according log message in the kamailio logs. What I see though is this: Sep 14 13:51:26 sp2 /usr/sbin/kamailio[15950]: ERROR: <core> [udp_server.c:586]: ERROR: udp_send: sendto(sock,0x97ec10,1055,0,1.2.3.4:58620,16): Invalid argument(22) Sep 14 13:51:26 sp2 /usr/sbin/kamailio[15950]: ERROR: <core> [forward.h:149]: msg_send: ERROR: udp_send failed The 1.2.3.4:58620 is obviously taken from the Vias, which looks like this: Via: SIP/2.0/UDP 127.0.0.1:5060;rport=5060;branch=xyz Via: SIP/2.0/UDP 10.1.1.9:58620;received=1.2.3.4;rport=58620;branch=abc The top-most Via is itself, the second is the one which reflects the destination address in the error message above. So I'm wondering whether there is some issue with finding the proper socket after wt_timer is expired (note that the leg client<->kamailio uses a different socket (with public ip) than kamailio<->sems (which uses 127.0.0.1)). Or any other idea where this error could come from? It's actually an educated guess that this could be related to wt_timer, but I don't know what else it could be. Andreas

2 6

Re: [SR-Users] Dialog module showing calls that have already been terminated
by Timo Reimann 20 Sep '11

20 Sep '11

Hey Phillip, you probably forgot to keep the mailing list in CC, I'll do that for you. On 19.09.2011 08:43, Phillman25 Kyriacou wrote: > You are right. > > I added the function dlg_manage(); when there is a BYE or CANCEL and now > its working perfectly. Before i had this function defined only on > INVITES. The strange thing is that it was working normally before when > dlg_manage(); was defined only for INVITE messages in the route block. So you are saying that before, you called dlg_manage() on INVITEs only but now, you need that function on both INVITE and BYE/CANCEL messages? That'd still qualify as a bug as the trigger to tracking the dialog (i.e., dlg_manage()) is required only once. After that, the dialog module makes sure that all dialog-related messages are properly processed. The only thing that's necessary for sequential requests to be tracked correctly is a call to loose_route() on such messages. If you are sure that your Kamailio configuration didn't change between 3.1.2 and 3.1.5 w.r.t. dialog tracking or anything that could affect it, I'd be willing to take a closer look at this issue (giving you can provide me with more information on your scenario, like call flow examples or similar). Cheers, --Timo > The calls that were suppose to be terminated before where in state 3 to > answer your question. However, i still see calls in state 3 now but they > are valid. > > Regards > Phillip > > On Fri, Sep 16, 2011 at 8:01 PM, Timo Reimann <timo.reimann(a)1und1.de > <mailto:timo.reimann@1und1.de>> wrote: > > Hey Phillip, > > after looking closer at what has been reported to me initially, your > case may be different after all. > > Could you possibly use "kamctl fifo dlg_list" to check what state the > majority of dialogs that should be terminated in your opinion are in? > Being given the reference counter should be useful in debugging this as > well. > > > Thanks, > > --Timo > > > > On 16.09.2011 14:22, Phillman25 Kyriacou wrote: > > On Fri, Sep 16, 2011 at 3:09 PM, Timo Reimann > <timo.reimann(a)1und1.de <mailto:timo.reimann@1und1.de> > > <mailto:timo.reimann@1und1.de <mailto:timo.reimann@1und1.de>>> wrote: > > > > Hey Phillip, > > > > > > On 16.09.2011 13:35, Phillman25 Kyriacou wrote: > > > Hello > > > > > > I have been facing an issue where the dialog module is showing > > calls as > > > being active when in fact the call has already been > completed long ago > > > and this is giving wrong number of concurrent calls on our > SNMP work > > > station (CACTI) when polling the data from Kamailio. I > realized this > > > only started occurring after i upgraded from 3.1.2 to 3.1.5, has > > anyone > > > experienced the same issue? > > > > I was *just* being notified of issues concerning dialogs not being > > deleted. Working on this right now to report back soon. > > > > Thanks for the note! > > > > > > Cheers, > > > > --Timo

2 1

Add client certificate checking against a CRL in opensips
by tomsc 20 Sep '11

20 Sep '11

Hi everyone, I'd like to check that a client certificat is revoked or not against a crl. Actually, opensips use context SSL_CTX. How can I do with this context? I do this change to load the crl : load_crl(SSL_CTX * ctx, char *filename) { LM_DBG("entered load crl\n"); X509_STORE *pStore = SSL_CTX_get_cert_store(ctx); X509_LOOKUP* plookup = X509_STORE_add_lookup(pStore, X509_LOOKUP_file()); if (!X509_load_crl_file(plookup, filename, X509_FILETYPE_PEM)){ LM_ERR("unable to load certificate file '%s'\n", filename); return -1; } LM_DBG("'%s' successfuly loaded\n", filename); return 0; } But I dont know how to add flags, is it the right solution? Thanks all -- View this message in context: http://old.nabble.com/Add-client-certificate-checking-against-a-CRL-in-open… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

2 1

Lazy evaluation in if expressions?
by Morten Isaksen 20 Sep '11

20 Sep '11

Does Kamailio use lazy evaluation in if expressions? Like if ($var(a) == 2 && $var(b) ==3) { ... } then $var(b) == 3 is only evaluated if $var(a) == 2 is true -- Morten Isaksen

3 2

directed media problem when video is included in sdp
by Juha Heinanen 20 Sep '11

20 Sep '11

when sems receives voice call invite from ua behind nat that has direction:active in its sdp, like this: Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): foo.bar 0 0 IN IP4 192.168.1. 68 Session Name (s): - Connection Information (c): IN IP4 192.168.1.68 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 8500 RTP/AVP 18 4 112 100 101 0 8 106 Media Attribute (a): sendrecv Media Attribute (a): rtpmap:18 G729/8000 Media Attribute (a): rtpmap:4 G723/8000 Media Attribute (a): rtpmap:112 AMR/8000 Media Attribute (a): rtpmap:100 EG711U/8000 Media Attribute (a): rtpmap:101 EG711A/8000 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:106 telephone-event/8000 Media Attribute (a): direction:active sems includes direction:passive in 200 OK and starts to correctly send rtp packets to the ip address/port of received rtp packets. however, then sems receives a voice+video call invite like this: Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): foo.bar 0 0 IN IP4 192.168.1.68 Session Name (s): - Connection Information (c): IN IP4 192.168.1.68 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 8500 RTP/AVP 18 4 112 100 101 0 8 106 Media Attribute (a): sendrecv Media Attribute (a): rtpmap:18 G729/8000 Media Attribute (a): rtpmap:4 G723/8000 Media Attribute (a): rtpmap:112 AMR/8000 Media Attribute (a): rtpmap:100 EG711U/8000 Media Attribute (a): rtpmap:101 EG711A/8000 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:106 telephone-event/8000 Media Description, name and address (m): video 8600 RTP/AVP 34 Bandwidth Information (b): AS:300 Media Attribute (a): sendrecv Media Attribute (a): rtpmap:34 H263/90000 Media Attribute (a): direction:active it does not include direction:passive in 200 OK and keeps on sending rtp packets to 192.168.1.68 even after receiving first voice rtp packet from public ip of the ua. is this a bug in sems or is it a bug in sr nathelper_k fix_nated_sdp("1") implementation, i.e., does that function place direction:active line to only one place, when it should place it in two places (in both voice and video media descriptions)? -- juha

3 4

Module db_text
by Bruno Bresciani 20 Sep '11

20 Sep '11

Hi All, I'm starting kamailio version 3.1.2 with db_text module but when it tries to load route table, the following error is generated: * "ERROR: <core> [daemonize.c:273]: Main process exited before writing to pipe" * Someone Can Help me to understand why this error is happening? Below is part of kamaio's trace. Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_base.c:208]: _tbc->name: route Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_base.c:209]: _tbc->nrcols: 9 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_base.c:210]: _nc: 8 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_base.c:238]: new res with 8 cols Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_res.c:69]: new res with 8 cols Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: <core> [db_res.c:118]: allocate 28 bytes for result set at 0x82e23f0 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: <core> [db_res.c:155]: allocate 32 bytes for result names at 0x82e3798 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: <core> [db_res.c:165]: allocate 32 bytes for result types at 0x82e37c0 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_api.c:80]: allocate 8 bytes for RES_NAMES[0] at 0x82e2438 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_api.c:80]: allocate 8 bytes for RES_NAMES[1] at 0x82e2390 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_api.c:80]: allocate 8 bytes for RES_NAMES[2] at 0x82e23d0 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_api.c:80]: allocate 8 bytes for RES_NAMES[3] at 0x82e23e0 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_api.c:80]: allocate 8 bytes for RES_NAMES[4] at 0x82e37e8 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_api.c:80]: allocate 8 bytes for RES_NAMES[5] at 0x82e37f8 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_api.c:80]: allocate 8 bytes for RES_NAMES[6] at 0x82e3808 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: db_text [dbt_api.c:80]: allocate 8 bytes for RES_NAMES[7] at 0x82e3818 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: <core> [db_res.c:184]: allocate 8 bytes for rows at 0x82e3828 Sep 19 16:13:01 sswpst00 /usr/local/sbin/kamailio[25661]: DEBUG: <core> [db_row.c:119]: allocate 160 bytes for row values at 0x82e3838 Sep 19 16:13:01 sswpst00 kamailio: ERROR: <core> [daemonize.c:273]: Main process exited before writing to pipe Best Regards

2 1

Re: [SR-Users] rtpproxy VS Turn
by Mikael Williams 19 Sep '11

19 Sep '11

> > ICE is not supported as I select the Media Proxy (which is TURN) manually > in some Nat cases...Even note that I do not have peer to peer calls between > 2 endpoints, just have VOIP to PSTN calls...All my VOIP customers supports > STUN/TURN but without ICE > > One more thing: I would like to ask how rtp proxy help to bypass NAT else > than working as relay server...Does it have any other job like find the NAT > type or check blocked port or something? > > Regards > > > On Fri, Sep 16, 2011 at 5:50 PM, Klaus Darilion < > klaus.mailinglists(a)pernau.at> wrote: > >> STUN alone is not 100% sufficient, only ICE+TURN is reliable. Thus, only >> if all your clients support ICE+TURN then you can drop rtpproxy. >> >> klaus >> >> >> On 16.09.2011 16:47, Mikael Williams wrote: >> >>> Dear All, >>> >>> I would like to ask please if and in case I'm using STUN/TURN for Nat >>> traversal to drp the rtpproxy from my solution as NAT traversal and >>> relay server as I'm using SYUN for NAT traversal and TURN as relay in >>> case of Symmetric NAT >>> >>> Regards >>> >>> >>> ______________________________**_________________ >>> 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<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> >>> >> >

2 1

Re: [SR-Users] sipp to kamailio doesn't match ACK [SOLVED]
by Uri Shacked 19 Sep '11

19 Sep '11

Hi, I got the same problem. Do you have the XML SIPp file for example? 10x Uri

1 0

Dialog module showing calls that have already been terminated
by Phillman25 Kyriacou 16 Sep '11

16 Sep '11

Hello I have been facing an issue where the dialog module is showing calls as being active when in fact the call has already been completed long ago and this is giving wrong number of concurrent calls on our SNMP work station (CACTI) when polling the data from Kamailio. I realized this only started occurring after i upgraded from 3.1.2 to 3.1.5, has anyone experienced the same issue? my config: ======== loadmodule "dialog.so" . . . . # ----------------- DIALOG MODULE PARAMETERS----------------------------------# modparam("dialog", "enable_stats", 1) modparam("dialog", "hash_size", 4096) modparam("dialog", "rr_param", "did") modparam("dialog", "dlg_flag", 4) modparam("dialog", "timeout_avp", "$avp(i:10)") modparam("dialog", "dlg_extra_hdrs", "NULL") modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "detect_spirals", 1) modparam("dialog", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("dialog", "db_mode", 1) modparam("dialog", "db_update_period", 60) modparam("dialog", "db_fetch_rows", 500) modparam("dialog", "table_name", "dialog") modparam("dialog", "from_uri_column", "from_uri") modparam("dialog", "from_tag_column", "from_tag") modparam("dialog", "to_uri_column", "to_uri") modparam("dialog", "to_tag_column", "to_tag") modparam("dialog", "h_id_column", "hash_id") modparam("dialog", "h_entry_column", "hash_entry") modparam("dialog", "state_column", "state") modparam("dialog", "start_time_column", "start_time") modparam("dialog", "timeout_column", "timeout") modparam("dialog", "sflags_column", "sflags") modparam("dialog", "bridge_controller", "sip:controller@kamailio.org") modparam("dialog", "default_timeout", 7200) . . . . route { # MANAGE ALL DIALOGS #=================================================== if (is_method("INVITE")) { if(is_method("INVITE") && !has_totag()) { $dlg_ctx(timeout_route) = 12; $dlg_ctx(timeout_bye) = 1; } dlg_manage(); } Thanks Phillip

2 3

kamailio debian/ubuntu support
by Jon Bonilla 16 Sep '11

16 Sep '11

Hi all with the release of kamailio 3.1.5, I've done a small change to the debian repositories: Nightly builds of kamailio_3.0 branch have been dropped and the repository has been erased. There is no development in this branch and it has no sense to keep nightly builds for it. Those (few) people using kamailio_3.0-nightly repositories are advised to move their sources.list to the stable 3.0 repository of upgrade to 3.1 or master branches. kamailio 3.0 will continue to be packaged if any new stable version is released for etch, lenny, squeeze and lucid. For 3.1 branch, nightly builds and stable repos will be provided for lenny, lucid and squeeze When the time for 3.2 (master branch at the moment) comes, I'll add Wheezy support. Not sure about Ubuntu distro support. Should we add support to Ubuntu 11.10 or wait until the lts 12.04 is released? Any opinions welcomed cheers, Jon

4 4

rtpproxy VS Turn
by Mikael Williams 16 Sep '11

16 Sep '11

Dear All, I would like to ask please if and in case I'm using STUN/TURN for Nat traversal to drp the rtpproxy from my solution as NAT traversal and relay server as I'm using SYUN for NAT traversal and TURN as relay in case of Symmetric NAT Regards

2 1

Communication from IPv6 to Ipv4
by Loïc Fofana 16 Sep '11

16 Sep '11

My architecture is described below: TDM device----------- Call server[IPv4]---------------[IPv4]Kamailio[IPv6]------------[IPv6]Asterisk---------[IPv6]Linphone * * * * * * * * * * * * Media Server Rtpproxy Media Server and Rtpptoxy carry the media. Call Server carry only signaling. I have succeeded to establish audio communication between an IPv6 domain & an IPv4 domain i.e. : from an TDM device to a Linphone. However, When i try to establish a call from (IPv6 to IPv4), i.e. from Linphone to TDM device, the signaling is OK, but there's no voice. Because, Rtpproxy forward the media to the Call server instead the Media Server. Hence, there can't be voice. I have been stucking on this for a couple of months. I will be very glad if somebody have any idea. best regards!

2 1

Re: [SR-Users] Kamailio terminating after upgrading from 3.1.2 to 3.1.4
by Phillman25 Kyriacou 16 Sep '11

16 Sep '11

Dear Marius Thank you for your prompt reply. I apologise, As i am a newbie could you please explain how to produce the below output that you require? Thanking you in advance for your help! Regards Phillip > Hello > Hello. You should have a coredump . Can you please send the trace (bt full) as seen with gdb ? Thanks Marius > Im facing a big problem after upgrading from version 3.1.2 to 3.1.4. > The kamailio process keeps on terminating every now and then, this is > what i was able to retrieve from my logs. > > > Sep 14 12:52:31 SipProxy1 kernel: [71448.062089] __ratelimit: 15 > callbacks suppressed > Sep 14 12:52:31 SipProxy1 kernel: [71448.062096] kamailio[13963]: > segfault at 14 ip 0816452a sp bfedc940 error 4 in kamailio[8048000+1d9000] > > > Does anyone have an idea why this is happening? > > Your help is much appreciated! > > Regards > Phillip

5 9

(no subject)
by Loïc Fofana 16 Sep '11

16 Sep '11

This is my scenario: Call server -------------(IPV4)Kamailio(IPV6)----------Asterisk(IPV6)------Linphone(IPV6) * * * Media server

2 1

One way communications in Kamailio 3.1.4
by CaptWho 16 Sep '11

16 Sep '11

I'm running 3.1.4 on centos and I'm having some trouble with voice only going one way. Both extensions will ring each other, but after they connect, voice will only travel in one direction. One extension hears fine, but can't talk. I've totally opened up the firewalls (including port 5060) and I'm still having the trouble. I'm not behind NAT. I've tried it using X-Lite, VoIP phones and ATAs in a number of combinations. The problem is across the board. X-Lite sometimes automatically disconnects after 32 seconds. Does anyone have any suggestions on where to look? ANY help appreciated. This has been going on for weeks. -- View this message in context: http://old.nabble.com/One-way-communications-in-Kamailio-3.1.4-tp32460597p3… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

3 3

Open source HLR (Home Location Register).
by Henrik Aagaard Sørensen 16 Sep '11

16 Sep '11

I know this is the mailing list for Kamailio, but hopefully someone can help me out as it is "kinda" related to VoIP etc. Does anyone know any open source / free HLR implementation?

4 3

Siremis site access from Romania
by Dani Popa 16 Sep '11

16 Sep '11

Hi, There is any reasons why any linked Siremis site is not accessible from Romania ? i tried to access http://siremis.asipto.com/ and other sites and they are not accesible. Thanks, Dani Popa

4 6

Re: [SR-Users] planning v3.1.5
by Vitaliy Aleksandrov 16 Sep '11

16 Sep '11

There is a bug in UAC module. uac_req_send doesn't restore avps. I tried to backport your patch to 3.1 but with no success. Could you add this patch to a new stable release ?

2 9

DNS support
by CT Radu 15 Sep '11

15 Sep '11

Dear sirs and madams, I was wondering what is the current support for DNS fallback in the 3.1.x branch . Lets assume we have a new dialog started and the name in the To field needs to be resolved via a NAPTR. Kamailio sends a NAPTR request, gets 3 SRVs , each with a number of A or AAAA records. The best/most suitable SRV is selected, however no response is returned from any of the addresses in the A records. Now, we should go the second SRV and try the A records in there and so on. My question is how much from this kind of dns fallback algorithm is supported ? (I hope my explanation wasn't too confusing) Thank you, Costin Radu

2 1

OT: anyone in Bay Area
by Daniel-Constantin Mierla 14 Sep '11

14 Sep '11

Hello, anyone here living/working in San Francisco/Bay Area that has a bit of time to assist me with some stuff? Reply to me for follow up. Thanks, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

Kamailio v3.1.5 Released
by Daniel-Constantin Mierla 14 Sep '11

14 Sep '11

Hello, Kamailio SIP Server v3.1.5 stable release is out. This is a maintenance release of latest stable branch, 3.1, that includes fixes since release of v3.1.4. There is no change to database schema or configuration language structure. Deployments running previous v3.1.x versions are strongly recommended to be upgraded to v3.1.5. For more details about version 3.1.5, visit: http://www.kamailio.org/w/2011/09/kamailio-v3-1-5-released/ Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

preparing release of v3.1.5
by Daniel-Constantin Mierla 14 Sep '11

14 Sep '11

Hello, today starting with 14:00 I will be preparing the release of v3.1.5. If any devel has commits to do in branch 3.1, please do it before to avoid conflicts. Thanks, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 1

Invitation to connect on LinkedIn
by Dani Popa 14 Sep '11

14 Sep '11

SIP, I'd like to add you to my professional network on LinkedIn. - Dani Dani Popa Technical Manager at NECC Telecom Romania Confirm that you know Dani Popa: https://www.linkedin.com/e/-szw6fj-gskbz4q0-1q/isd/4204604361/O7PsXc0-/?hs=… -- You are receiving Invitation to Connect emails. Click to unsubscribe: http://www.linkedin.com/e/-szw6fj-gskbz4q0-1q/c6YlT3v6LB4U4BZW-5Wc20tVAQF44… (c) 2011 LinkedIn Corporation. 2029 Stierlin Ct, Mountain View, CA 94043, USA.

1 0

Kamailio terminating after upgrading from 3.1.2 to 3.1.4
by Phillman25 Kyriacou 14 Sep '11

14 Sep '11

Hello Im facing a big problem after upgrading from version 3.1.2 to 3.1.4. The kamailio process keeps on terminating every now and then, this is what i was able to retrieve from my logs. Sep 14 12:52:31 SipProxy1 kernel: [71448.062089] __ratelimit: 15 callbacks suppressed Sep 14 12:52:31 SipProxy1 kernel: [71448.062096] kamailio[13963]: segfault at 14 ip 0816452a sp bfedc940 error 4 in kamailio[8048000+1d9000] Sep 14 12:52:31 SipProxy1 /usr/local/sbin/kamailio[13991]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 49 Sep 14 12:52:31 SipProxy1 /usr/local/sbin/kamailio[13907]: ALERT: <core> [main.c:741]: child process 13963 exited by a signal 11 Sep 14 12:52:31 SipProxy1 /usr/local/sbin/kamailio[13907]: ALERT: <core> [main.c:744]: core was generated Sep 14 12:52:31 SipProxy1 /usr/local/sbin/kamailio[13907]: INFO: <core> [main.c:756]: INFO: terminating due to SIGCHLD Does anyone have an idea why this is happening? Your help is much appreciated! Regards Phillip

2 1

parse errors in rr-callback of path module
by Andreas Granig 14 Sep '11

14 Sep '11

Hi, path module calls register_rrcb when "use_received" is set to 1 to extract the "received=" param and set its content as dst-uri if present. Now I've noticed that when e.g. a message has a pre-loaded route like "Route: <sip:mydomain.com;lr>", then kamailio (at least 3.1.3) spits the following error message: /usr/sbin/kamailio[23953]: ERROR: <core> [parser/parse_param.c:481]: parse_params(): Invalid character, ; expected /usr/sbin/kamailio[23953]: ERROR: path [path.c:213]: failed to parse route parametes I was trying to reproduce this but without much success, it sometimes happens and sometimes not, but I can't yet put the finger on the reason. Any ideas? Andreas

3 8

planning v3.1.5
by Daniel-Constantin Mierla 14 Sep '11

14 Sep '11

Hello, it's being quite some time since we released 3.1.4, so I am thinking of packaging soon v3.1.5 out of stable branch 3.1. My plan is to release it Wednesday, September 14, 2011. Any other opinions? Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

6 6

Does xcap server of kamailio support partial update of xcap document?
by laura testi 13 Sep '11

13 Sep '11

Hi, I am using the XCAP server of kamailio for contact list and presence rules. It seems it only support to update the whole xcap doc of the user but not the partial update. Can anyone tell me if the kamailio's xcap server support the partial update of a user's doc? for example, to remove a contact from the contact list or add a contact to the contact list without PUT or DELETE the whole xcap document of this user? If yes, what's the syntax of the HTTP request? Thank you very much in advanced Best Regards, Laura

3 12

max_contacts
by Rafael Rincon 13 Sep '11

13 Sep '11

Hello Guys, Is there any chance to modify at running time (in a register event) the max_contacts parameter? I am running OpenSER version 1.4 . Regards, Rafael Rincon rrincon(a)ipcomnetwork.com +13054772902 x 111

4 7

cr default route
by caio 13 Sep '11

13 Sep '11

Hello, The documentation of carrierroute says "If flags and mask are not zero, and no match to the message flags is possible, no routing will be done." Then is there any route entry which can act as default gw (like an always-match entry)? For example this 'carrierroute' table: +---------+--------+-------------+-------+------+------+ | carrier | domain | scan_prefix | flags | mask | +---------+--------+-------------+-------+------+------+ 1 | 1 | 1 | 156 | 500 | 0 | 2 | 1 | 1 | | 0 | 0 | 3 | 1 | 2 | | 0 | 0 | +---------+--------+-------------+-------+------+- Here if ruri is 156xx, and there is no flag set, then the cr module does not find a match because of the non-zero value in the flag column (entry 1). But I want to route this call via carrier1 domain1 and scan_prefix empty-value entry (entry 2). How can I set a default route entry here? Is possible with cr? Now I only find it possible via a pstn.def_gw_ip variable. Thank you, Claudio

2 3

wildcarded sip contacts
by Antanas Masevicius 13 Sep '11

13 Sep '11

Hello everyone, been looking for a way to do sip wildcarded contact matching where contact could look like <sip:2222..@33.33.33.33:5060>. Current usrloc module does not implement any such matching, but it seems there is a new rfc 5947 issued regarding this problem. Maybe someone is already working on it? What options do i have in sip-router / kamailio besides LCR based ones ? I want to do local contact matching as fast as possible and without a need to reload lcr table. Antanas Masevicius NTT

1 0

Re: [SR-Users] Fwd: configuration
by Daniel-Constantin Mierla 13 Sep '11

13 Sep '11

Hello, On 9/12/11 4:14 PM, neili ma wrote: > Hello, > > Thanks a lot for your advice, I found and solved the problem thanks to > the debugger, actually the problem was with > > modparam("auth_db", > "db_url","mysql://openser:openserrw@localhost/openser") > > I just had to change localhost to my locationserver IP and the problem > was solved :), thanks again. good that you find it. Don't forget to cc the mailing list always so people reading the thread (archive later) see the conclusion -- it can help many others running on same situation. Cheers, Daniel > > Sincerely, > Neili > > > 2011/9/12 Daniel-Constantin Mierla <miconda(a)gmail.com > <mailto:miconda@gmail.com>> > > Hello, > > you have another group of actions that can send 401 Unauthorized: > > > if (!check_to()) { > sl_send_reply("401", "Unauthorized"); > exit; > } > > You can either run with debug=4 and watch the syslog to see some > hints in messages where it fires this 401, or use debugger module > with cfgtrace option set and you see the execution path of the > config in the logs. > > If you send here the ngrep of such REGISTER and the reply we can > spot what could be the situation for 401. > > Cheers, > Daniel > > > On 9/11/11 9:18 PM, neili ma wrote: >> Hello sr-users, >> >> I'm new using kamailio open source, and I would like to ask for help. >> >> My version is 3.1.4 and I am trying to configure 3 servers, one >> with proxy, one with registrar and one with location server. >> >> My problem is when I try to send a register message it returns an >> error "401 Unauthorized", I know the error is in the fuction >> www_authorize("10.254.239.7", "subscriber") but I don't know how >> to solve it, for me it seems correct the parameters. >> >> Below I have attatched the proxy and registrar relevant route >> code for my problem, and the IPs for proxy an registrar are: >> 10.254.239.7-----------10.254.239.8 >> >> If someone can help me, I will be gratefull. Thanks in advance. >> >> ####REGISTRAR CODE >> >> route{ >> >> # initial sanity checks -- too long messages >> if (msg:len >= 2048 ) { >> sl_send_reply("513", "Message too big"); >> exit; >> }; >> >> if (!uri==myself) { >> sl_send_reply("400", "Bad destination, only >> register messages de$ >> exit; >> }; >> >> # the only valid message for the registrar is the register >> message >> # other messages are answered with an error message >> if (uri==myself) { >> >> if (method=="REGISTER") { >> >> sl_send_reply("100", "Trying"); >> >> if (!www_authorize("10.254.239.7", "subscriber")) { >> www_challenge("10.254.239.7", "1"); >> exit; >> }; >> >> if (!check_to()) { >> sl_send_reply("401", "Unauthorized"); >> exit; >> }; >> >> consume_credentials(); >> if(!save("location")) { >> sl_reply_error(); >> }; >> exit; >> } else { >> sl_send_reply("403", "Forbidden"); >> exit; >> }; >> }; >> } >> >> #########PROXY CODE >> >> if (method=="ACK") { >> route(1); >> exit; >> } if (method=="INVITE") { >> route(3); >> exit; >> } else if (method=="REGISTER") { >> route(2); >> exit; >> }; >> >> lookup("aliases"); >> if (!uri==myself) { >> route(1); >> }; >> >> route[2] { >> #rewrinting the host allows the registrar to know that the >> message is de$ >> >> rewritehost("10.254.239.8"); >> if(!t_relay()) { >> sl_reply_error(); >> }; >> exit; >> } >> >> >> >> >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > -- > Daniel-Constantin Mierla --http://www.asipto.com > Kamailio Advanced Training, Oct 10-13, Berlin:http://asipto.com/u/kat > http://linkedin.com/in/miconda -- http://twitter.com/miconda > > -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

CANCEL is not processed in onsend_route
by Timo Klecker 13 Sep '11

13 Sep '11

Hello all, I want to log the incoming and outgoing CANCEL on my Kamailio. I tried with a call of sip_trace in the default route setflag(22); if (method == "CANCEL") { sip_trace(); if (t_check_trans()) { t_relay(); exit; } } This works great, I get the incoming CANCEL in the Database. But the outgoing CANCEL is not logged with flag set. So I tried to use onsend_route with the following lines: onsend_route { xlog("L_NOTICE", " SENDING from $si \n $mb \n "); if ( method == "CANCEL" ) sip_trace(); } I am using Kamailio Version 3.1.0 (with Patch for Siptrace in onsend_route by Daniel ). But the CANCEL is not processed in the onsend_route at all. I do not even see the xlog-line. But tcpdump gives me the CANCEL outgoing. Following callflow: INVITE > 100 < > INVITE < 100 < 180 180 < CANCEL> > CANCEL I am missing the last CANCEL. Why is the CANCEL not logged with flag 22 set? Why is the CANCEL not processed in the onsend_route? Greetings Timo

3 4

question bout UA and BOX in same NAT
by MingHon 12 Sep '11

12 Sep '11

Hello List, Can i have UA in the same nat with the box and also UA in different nat at the same time? My box is currrently working behind nat with ip 192.168.2.3 and advertised_address is set to sip.mydomain.com is it possible to append the via header replace the sip.mydomain.com to 192.168.2.3 ? Thanks, -- Regards, MingHon

2 4

no publish from pua/pua_mi to outbound proxy
by Juha Heinanen 12 Sep '11

12 Sep '11

i started to test sr presence and for some reason sr does not send out any publish request when i issue one via pua_mi. i have in route [xmlrpc_requests]: xlog("L_INFO", "Handling XMLRPC $rm from <$si> with body <$(rb{re.subst,/[\n\r\t]//g})>\n"); set_reply_no_connect(); set_reply_close(); dispatch_rpc(); syslog shows that sr receives xmlrpc request: Aug 29 19:42:13 sip /usr/sbin/pres-serv[2118]: INFO: Handling XMLRPC POST from <127.0.0.1> with body <<?xml version='1.0'?><methodCall><methodName>mi</methodName><params><param><value><string>pua_publish</string></value></param><param><value><string>sip:jh@vm.test.fi</string></value></param><param><value><int>7776000</int></value></param><param><value><string>message-summary</string></value></param><param><value><string>application/simple-message-summary</string></value></param><param><value><string>.</string></value></param><param><value><string>a.1313821931.6178.2.0</string></value></param><param><value><string>P-Flags: 0</string></value></param><param><value><string>Messages-Waiting:yesVoice-Message: 1/1 (0/0)</string></value></param></params></methodCall>> and wireshark shows that sr xmlrpc server responds to it with 200 ok. but after than nothing happens, i.e., according to wireshark no publish request is sent out. i have pua and pua_mi modules loaded and outbound proxy defined: modparam("pua", "outbound_proxy", "sip:127.0.0.1:5082") there is no error messages in syslog. any idea why sr does not send publish to outbound proxy? this used to work with opensips. -- juha

2 13

Nat problems
by David Zambrano 12 Sep '11

12 Sep '11

Hi Ive been trying to configure kamailio as a loadbalancer between two transcoders that afterwards send all calls to a softswitch but I currently have two problems. My first and most important one is a NAT problem or so I think. All calls in UDP work perfectly but when dealing with calls in TCP only outgoing calls work, no incoming calls work. After doing some traces on the network I realized that the transcoder is trying to reach the router in front of the softphone and skipping the Kamailio loadbalancer. The call never reaches the softphone so the phonecall never completes. In UDP exactly the same thing happens, the call skips the Kamailio loadbalancer but the call completes perfectly fine. The other detail has to do with the actual loadbalancing and failover. I was using the weight distribution and it works great except its sending the same CSeq to both transcoders and then the softswitch is receiving it twice and an error pops up at the softphone Ive been using saying theres a duplicate CSeq. I was using the forward() command after the dispatcher arguments. I changed it to the t_relay() and it works as in it doesnt send the same CSeq twice but instead a unique one to each transcoder so the Softswitch stopped winning but then the failover doesnt work. Whenever it one goes down its till trying to reach it. Im using roundrobin in the dispatcher config because I noticed it doesnt send stuff to both transcoders but only to the transcoder its going to actually use. I started to read that for failover I need the user_avp module which I didnt install but then I wonder how come failover worked before? Thanks for your help Cheers David

3 7

Fwd: configuration
by neili ma 12 Sep '11

12 Sep '11

Hello sr-users, I'm new using kamailio open source, and I would like to ask for help. My version is 3.1.4 and I am trying to configure 3 servers, one with proxy, one with registrar and one with location server. My problem is when I try to send a register message it returns an error "401 Unauthorized", I know the error is in the fuction www_authorize("10.254.239.7", "subscriber") but I don't know how to solve it, for me it seems correct the parameters. Below I have attatched the proxy and registrar relevant route code for my problem, and the IPs for proxy an registrar are: 10.254.239.7-----------10.254.239.8 If someone can help me, I will be gratefull. Thanks in advance. ####REGISTRAR CODE route{ # initial sanity checks -- too long messages if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; if (!uri==myself) { sl_send_reply("400", "Bad destination, only register messages de$ exit; }; # the only valid message for the registrar is the register message # other messages are answered with an error message if (uri==myself) { if (method=="REGISTER") { sl_send_reply("100", "Trying"); if (!www_authorize("10.254.239.7", "subscriber")) { www_challenge("10.254.239.7", "1"); exit; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); exit; }; consume_credentials(); if(!save("location")) { sl_reply_error(); }; exit; } else { sl_send_reply("403", "Forbidden"); exit; }; }; } #########PROXY CODE if (method=="ACK") { route(1); exit; } if (method=="INVITE") { route(3); exit; } else if (method=="REGISTER") { route(2); exit; }; lookup("aliases"); if (!uri==myself) { route(1); }; route[2] { #rewrinting the host allows the registrar to know that the message is de$ rewritehost("10.254.239.8"); if(!t_relay()) { sl_reply_error(); }; exit; }

2 1

Re: [SR-Users] Kamailio 3.0.4 is crashed
by Daniel-Constantin Mierla 12 Sep '11

12 Sep '11

Hello, try to keep the mailing list cc-ed -- others may be able to help you faster or this discussion will be helpful for different people. The username_spec and password_spec were removed indeed, the first was useless since the username was taken from auth header and the second became parameter to pv_www_authenticate, see: http://kamailio.org/docs/modules/3.1.x/modules/auth.html#pv_www_authenticate You simply have to use something like: if (!pv_www_authenticate("$td", "$var(password)", "0")) { www_challenge("$td", "0"); exit; } Cheers, Daniel On 9/10/11 12:51 PM, Khoa Phuong Viet Ngo wrote: > Hi Daniel, > > Thanks Daniel for your answer. > > It's very helpful for me :). > > Actually i tried with using kamailio 3.1.4 (lastest kamailio) but i > got the problem with auth module and parameter with auth module. > Because some parameters at auth module have changed. > http://kamailio.org/docs/modules/stable/modules/auth.html [3.1.4] > > But in 3.0.4 kamailio, i am using auth module with these parameter and > it had error with new kamailio (3.1.4) . :( > > kamailio.cfg > > ######################################################## > loadmodule auth.so > > # ----- auth params ----- > modparam("auth", "username_spec", "$var(username)") > modparam("auth", "password_spec", "$var(password)") > modparam("auth", "calculate_ha1", 1) > > route[AUTH] { > $var(utype) = 0; > #!ifdef WITH_AUTH > if (is_method("REGISTER")) > { > if(is_present_hf("Authorization")) > { > # authenticate the REGISTER requests (uncomment to enable > auth) > if(!route(GETPASSWD)) > { > www_challenge("", "0"); > exit; > } > if(!pv_www_authorize("")) > { > www_challenge(""/*realm*/,"0"/*qop*/); > exit; > } > } else { > www_challenge("", "0"); > exit; > } > } else { > $var(srcgrp) = allow_source_address_group(); > if ($var(srcgrp) > 0) { > # src ip allowed > $var(utype) = 1; > if($var(srcgrp)!=100) > { > $rd = $Ri; > $fs = "udp:" + $Ri + ":5062"; > } > return 1; > } > # authenticate if from local subscriber (uncomment to enable auth) > if (from_uri==myself) > { > if(is_present_hf("Proxy-Authorization")) > { > # authenticate the non-REGISTER requests (uncomment to > enable auth) > if(!route(GETPASSWD)) > { > proxy_challenge("", "0"); > exit; > } > if(!pv_proxy_authorize("")) > { > proxy_challenge(""/*realm*/,"0"/*qop*/); > exit; > } > } else { > proxy_challenge("", "0"); > exit; > } > $var(utype) = 2; > consume_credentials(); > # caller authenticated > } > } > #!endif > return; > } > > ##################################################### > > Error logs: > > Sep 6 19:37:01 CorePBX kamailio: ERROR: <core> [modparam.c:150]: > set_mod_param_regex: parameter <username_spec> not found in module <auth> > Sep 6 19:37:01 CorePBX kamailio: : <core> [cfg.y:3419]: parse error > in config file /usr/local/etc/kamailio/kamailio.cfg, line 260, column > 51: Can't set module parameter > Sep 6 19:37:01 CorePBX kamailio: ERROR: <core> [modparam.c:150]: > set_mod_param_regex: parameter <password_spec> not found in module <auth> > Sep 6 19:37:01 CorePBX kamailio: : <core> [cfg.y:3419]: parse error > in config file /usr/local/etc/kamailio/kamailio.cfg, line 261, column > 51: Can't set module parameter > Sep 6 19:37:01 CorePBX kamailio: ERROR: <core> [modparam.c:150]: > set_mod_param_regex: parameter <calculate_ha1> not found in module <auth> > Sep 6 19:37:01 CorePBX kamailio: : <core> [cfg.y:3419]: parse error > in config file /usr/local/etc/kamailio/kamailio.cfg, line 262, column > 36: Can't set module parameter > Sep 6 19:37:01 CorePBX kamailio: : <core> [cfg.y:3419]: parse error > in config file /usr/local/etc/kamailio/kamailio.cfg, line 562, column > 27: unknown command, missing loadmodule? > Sep 6 19:37:01 CorePBX kamailio: : <core> [cfg.y:3419]: parse error > in config file /usr/local/etc/kamailio/kamailio.cfg, line 594, column > 30: unknown command, missing loadmodule? > Sep 6 19:37:39 CorePBX kamailio: ERROR: <core> [modparam.c:150]: > set_mod_param_regex: parameter <username_spec> not found in module <auth> > Sep 6 19:37:39 CorePBX kamailio: : <core> [cfg.y:3419]: parse error > in config file /usr/local/etc/kamailio/kamailio.cfg, line 260, column > 51: Can't set module parameter > Sep 6 19:37:39 CorePBX kamailio: ERROR: <core> [modparam.c:150]: > set_mod_param_regex: parameter <password_spec> not found in module <auth> > Sep 6 19:37:39 CorePBX kamailio: : <core> [cfg.y:3419]: parse error > in config file /usr/local/etc/kamailio/kamailio.cfg, line 261, column > 51: Can't set module parameter > Sep 6 19:37:39 CorePBX kamailio: ERROR: <core> [modparam.c:150]: > set_mod_param_regex: parameter <calculate_ha1> not found in module <auth> > > ######################################################## > > I appreciate all your help on this case. > > Thanks Daniel, > KhoaNVP > > > > > On Sat, Sep 10, 2011 at 1:58 PM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > Hello, > > it seems like a buffer overflow happened previously, since the > crash is in malloc. Do you get any error messages is syslog? Can > you recompile with QM_DBG_MALLOC like explained in the next link > and try again watching the syslog messages? > http://www.kamailio.org/dokuwiki/doku.php/troubleshooting:memory > > On another hand, 3.0.4 is quite old, why are not using 3.1.4, it > is far more updated than 3.0 series -- we are just to release > first in 3.2 series. Upgrading from 3.0 to 3.1 is rather trivial. > > Cheers, > Daniel > > > On 9/9/11 8:38 PM, KhoaNVP wrote: > > Hi everyone, > > I hope you can help me with this issues. > I'm working with kamailio 3.0.4 and when i implement the call > via kamailio > after 1-->3 mins , kamailio is crashed. > > It's my debug from core log. > > Core was generated by `kamailio -E -ddd'. > Program terminated with signal 11, Segmentation fault. > #0 fm_malloc (qm=0x88c860, size=<value optimized out>) at > mem/f_malloc.c:382 > 382 *f=frag->u.nxt_free; > (gdb) qt > Undefined command: "qt". Try "help". > (gdb) bt > #0 fm_malloc (qm=0x88c860, size=<value optimized out>) at > mem/f_malloc.c:382 > #1 0x000000000041fd27 in dup_lump_list_r (l=0x9275f8, > dir=LD_AFTER, > error=0x7fff23bd661c) at data_lump.c:453 > #2 0x000000000041ffa9 in dup_lump_list_r (l=0x927640, > dir=LD_AFTER, > error=0x7fff23bd666c) at data_lump.c:476 > #3 0x000000000041ff6d in dup_lump_list_r (l=0x926db0, > dir=<value optimized > out>, error=0x7fff23bd66b4) at data_lump.c:465 > #4 0x0000000000420058 in dup_lump_list (l=0x88c860) at > data_lump.c:508 > #5 0x00002b8b2efb858d in prepare_new_uac (t=0x2b8b3218ffa8, > i_req=0x926ef0, > branch=0, uri=0x927150, path=0x9274d0, next_hop=<value > optimized out>, > fsocket=0x0, snd_flags=0 '\000', fproto=0, flags=0) at > t_fwd.c:242 > #6 0x00002b8b2efb94c7 in add_uac (t=0x2b8b3218ffa8, > request=0x926ef0, > uri=0x927150, next_hop=0x7, path=0x9274d0, proxy=0x0, fsocket=0x0, > snd_flags=0 '\000', proto=0, flags=0) at t_fwd.c:679 > #7 0x00002b8b2efbce7a in t_forward_nonack (t=0x2b8b3218ffa8, > p_msg=0x926ef0, proxy=0x0, proto=0) at t_fwd.c:1340 > #8 0x00002b8b2efb4996 in t_relay_to (p_msg=0x926ef0, > proxy=0x0, proto=0, > replicate=0) at t_funcs.c:358 > #9 0x0000000000413489 in do_action (h=0x7fff23bd71e0, a=0x8be788, > msg=0x926ef0) at action.c:860 > #10 0x0000000000415e03 in run_actions (h=0x7fff23bd71e0, > a=0x8be788, > msg=0x926ef0) at action.c:1315 > #11 0x00000000004161e7 in run_actions_safe (h=0x7fff23bd8bb0, > a=0x38, > msg=0x88ca08) at action.c:1370 > #12 0x000000000049696b in rval_get_int (h=0x7fff23bd8bb0, > msg=0x88ca08, > i=0x7fff23bd75dc, rv=0x7, cache=0x7) at rvalue.c:897 > #13 0x000000000049b7fc in rval_expr_eval_int (h=0x7fff23bd8bb0, > msg=0x926ef0, res=0x7fff23bd75dc, rve=0x8be878) at rvalue.c:1841 > #14 0x000000000049b8ad in rval_expr_eval_int (h=0x7fff23bd8bb0, > msg=0x926ef0, res=0x7fff23bd79d8, rve=0x8bef48) at rvalue.c:1848 > #15 0x0000000000411d49 in do_action (h=0x7fff23bd8bb0, a=0x8bf798, > msg=0x926ef0) at action.c:834 > #16 0x0000000000415e03 in run_actions (h=0x7fff23bd8bb0, > a=0x8be608, > msg=0x926ef0) at action.c:1315 > #17 0x0000000000413084 in do_action (h=0x7fff23bd8bb0, a=0x8f6060, > msg=0x926ef0) at action.c:479 > #18 0x0000000000415e03 in run_actions (h=0x7fff23bd8bb0, > a=0x8f6060, > msg=0x926ef0) at action.c:1315 > #19 0x0000000000411dab in do_action (h=0x7fff23bd8bb0, a=0x8f62d0, > msg=0x926ef0) at action.c:853 > #20 0x0000000000415e03 in run_actions (h=0x7fff23bd8bb0, > a=0x8f62d0, > msg=0x926ef0) at action.c:1315 > #21 0x0000000000413084 in do_action (h=0x7fff23bd8bb0, a=0x8bd698, > msg=0x926ef0) at action.c:479 > #22 0x0000000000415e03 in run_actions (h=0x7fff23bd8bb0, > a=0x8bd398, > msg=0x926ef0) at action.c:1315 > #23 0x0000000000411dab in do_action (h=0x7fff23bd8bb0, a=0x8bd788, > msg=0x926ef0) at action.c:853 > #24 0x0000000000415e03 in run_actions (h=0x7fff23bd8bb0, > a=0x8a1340, > msg=0x926ef0) at action.c:1315 > #25 0x0000000000416174 in run_top_route (a=0x8a1340, > msg=0x926ef0, c=<value > optimized out>) at action.c:1391 > #26 0x000000000048009c in receive_msg (buf=0x57e779 "DEBUG", > len=<value > optimized out>, rcv_info=0x7fff23bd8e20) at receive.c:196 > #27 0x000000000050698b in udp_rcv_loop () at udp_server.c:520 > #28 0x0000000000455e2f in main_loop () at main.c:1447 > #29 0x0000000000456f32 in main (argc=<value optimized out>, > argv=0x7fff23bd90e8) at main.c:2251 > (gdb) > > > Thanks with all the help, > Khoa > > > -- > Daniel-Constantin Mierla -- http://www.asipto.com > Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat > http://linkedin.com/in/miconda -- http://twitter.com/miconda > > -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

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Kamailio 3.0.4 is crashed
by KhoaNVP 10 Sep '11

10 Sep '11

Hi everyone, I hope you can help me with this issues. I'm working with kamailio 3.0.4 and when i implement the call via kamailio after 1-->3 mins , kamailio is crashed. It's my debug from core log. Core was generated by `kamailio -E -ddd'. Program terminated with signal 11, Segmentation fault. #0 fm_malloc (qm=0x88c860, size=<value optimized out>) at mem/f_malloc.c:382 382 *f=frag->u.nxt_free; (gdb) qt Undefined command: "qt". Try "help". (gdb) bt #0 fm_malloc (qm=0x88c860, size=<value optimized out>) at mem/f_malloc.c:382 #1 0x000000000041fd27 in dup_lump_list_r (l=0x9275f8, dir=LD_AFTER, error=0x7fff23bd661c) at data_lump.c:453 #2 0x000000000041ffa9 in dup_lump_list_r (l=0x927640, dir=LD_AFTER, error=0x7fff23bd666c) at data_lump.c:476 #3 0x000000000041ff6d in dup_lump_list_r (l=0x926db0, dir=<value optimized out>, error=0x7fff23bd66b4) at data_lump.c:465 #4 0x0000000000420058 in dup_lump_list (l=0x88c860) at data_lump.c:508 #5 0x00002b8b2efb858d in prepare_new_uac (t=0x2b8b3218ffa8, i_req=0x926ef0, branch=0, uri=0x927150, path=0x9274d0, next_hop=<value optimized out>, fsocket=0x0, snd_flags=0 '\000', fproto=0, flags=0) at t_fwd.c:242 #6 0x00002b8b2efb94c7 in add_uac (t=0x2b8b3218ffa8, request=0x926ef0, uri=0x927150, next_hop=0x7, path=0x9274d0, proxy=0x0, fsocket=0x0, snd_flags=0 '\000', proto=0, flags=0) at t_fwd.c:679 #7 0x00002b8b2efbce7a in t_forward_nonack (t=0x2b8b3218ffa8, p_msg=0x926ef0, proxy=0x0, proto=0) at t_fwd.c:1340 #8 0x00002b8b2efb4996 in t_relay_to (p_msg=0x926ef0, proxy=0x0, proto=0, replicate=0) at t_funcs.c:358 #9 0x0000000000413489 in do_action (h=0x7fff23bd71e0, a=0x8be788, msg=0x926ef0) at action.c:860 #10 0x0000000000415e03 in run_actions (h=0x7fff23bd71e0, a=0x8be788, msg=0x926ef0) at action.c:1315 #11 0x00000000004161e7 in run_actions_safe (h=0x7fff23bd8bb0, a=0x38, msg=0x88ca08) at action.c:1370 #12 0x000000000049696b in rval_get_int (h=0x7fff23bd8bb0, msg=0x88ca08, i=0x7fff23bd75dc, rv=0x7, cache=0x7) at rvalue.c:897 #13 0x000000000049b7fc in rval_expr_eval_int (h=0x7fff23bd8bb0, msg=0x926ef0, res=0x7fff23bd75dc, rve=0x8be878) at rvalue.c:1841 #14 0x000000000049b8ad in rval_expr_eval_int (h=0x7fff23bd8bb0, msg=0x926ef0, res=0x7fff23bd79d8, rve=0x8bef48) at rvalue.c:1848 #15 0x0000000000411d49 in do_action (h=0x7fff23bd8bb0, a=0x8bf798, msg=0x926ef0) at action.c:834 #16 0x0000000000415e03 in run_actions (h=0x7fff23bd8bb0, a=0x8be608, msg=0x926ef0) at action.c:1315 #17 0x0000000000413084 in do_action (h=0x7fff23bd8bb0, a=0x8f6060, msg=0x926ef0) at action.c:479 #18 0x0000000000415e03 in run_actions (h=0x7fff23bd8bb0, a=0x8f6060, msg=0x926ef0) at action.c:1315 #19 0x0000000000411dab in do_action (h=0x7fff23bd8bb0, a=0x8f62d0, msg=0x926ef0) at action.c:853 #20 0x0000000000415e03 in run_actions (h=0x7fff23bd8bb0, a=0x8f62d0, msg=0x926ef0) at action.c:1315 #21 0x0000000000413084 in do_action (h=0x7fff23bd8bb0, a=0x8bd698, msg=0x926ef0) at action.c:479 #22 0x0000000000415e03 in run_actions (h=0x7fff23bd8bb0, a=0x8bd398, msg=0x926ef0) at action.c:1315 #23 0x0000000000411dab in do_action (h=0x7fff23bd8bb0, a=0x8bd788, msg=0x926ef0) at action.c:853 #24 0x0000000000415e03 in run_actions (h=0x7fff23bd8bb0, a=0x8a1340, msg=0x926ef0) at action.c:1315 #25 0x0000000000416174 in run_top_route (a=0x8a1340, msg=0x926ef0, c=<value optimized out>) at action.c:1391 #26 0x000000000048009c in receive_msg (buf=0x57e779 "DEBUG", len=<value optimized out>, rcv_info=0x7fff23bd8e20) at receive.c:196 #27 0x000000000050698b in udp_rcv_loop () at udp_server.c:520 #28 0x0000000000455e2f in main_loop () at main.c:1447 #29 0x0000000000456f32 in main (argc=<value optimized out>, argv=0x7fff23bd90e8) at main.c:2251 (gdb) Thanks with all the help, Khoa -- View this message in context: http://old.nabble.com/Kamailio-3.0.4-is-crashed-tp32433892p32433892.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

2 1

Error reading AVP
by Carlos Ruiz Diaz 09 Sep '11

09 Sep '11

Hello. I'm currently developing an OpenSER module which does some business logic for the company I work for. For compatibility reasons we are working with openser-1.2.2-tls and we haven't been able to upgrade to newer version yet. In one block of the code I have to read an avp which is written inside the openser.cfg script using: OPENSER: avp_copy("sip:123@123.123.123:5060", "$avp(s:selectedroute)/g"); C CODE: if (is_response && is_invite_seq) { struct usr_avp* destavp = NULL; int_str destval; const str selected_route = { "selectedroute", 13 }; int attempts = 1, max_attempts = 3; for ( destavp = search_first_avp(AVP_NAME_STR | AVP_VAL_STR, (int_str) selected_route, NULL, 0); destavp != NULL; destavp = search_next_avp(destavp, NULL) ) { get_avp_val(destavp, &destval); } } The problem is that "selectedroute" avp cannot be found so I can't read its value. This happens sporadically and most of the time the code works just fine. I also checked for the list of available AVPs using get_avp_list() but it reports an empty list. Is there another way to read AVPs or is there something I'm missing which is causing this problem? Any help will be appreciated. Thanks in advance! Carlos.

2 2

10 Years SER Conference - Presentations
by Daniel-Constantin Mierla 08 Sep '11

08 Sep '11

Hello, the presentations from 10 Years SER Conference I received so far were uploaded at: * http://www.kamailio.org/events/2011-10yearsSER/ Expect the rest during the next days. The recording of the talk about Gemeinschaft 4 by Stefan Wintermeyer is available on youtube: * http://www.youtube.com/watch?v=mbGn5N_y4rw Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

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Re: [SR-Users] Kamailio + rtpproxy talking to multiple carrier gateways - Solution
by Sarat C. Vemuri 07 Sep '11

07 Sep '11

I think I have this working. For anyone else trying to do similar thing (or placing Kamailio behind FW/NAT), here is my cookbook. We are trying to setup the following: [UAS] <------> (PubIP)[FireWall/One-to-OneNAT]<-------->(PrivIP)[Kamailio/RTPProxy](PrivIP)<---------->(PrivIP2)[UAC] UAC/UAS cannot reach each other directly. For simplicity I am running RTPProxy on same machines as Kamailio. But this can be changed. Versions/Patches/fixes you need if you are running 3.1.2 : (Trunk/3.2 has these): a. Fixed forward.c for mhome to work properly (included in 3.1.4). b. rr module from trunk to be able to use 2 strings for record_route_preset c. Patched rtpproxy module to actually use the forced ip supplied to force_rtp_proxy. Patch at the end of this message d. rtpproxy version 1.2.1 What to do: 1. Setup FW/NAT rules to map public IP to PrivateIP and allow communications to/from UAS 2. Setup RTPProxy in "bridged" mode (e.g -F -l 10.10.5.90/10.10.5.90 -s udp:10.10.5.90:9005) listening on PrivateIP. 3. Setup the config file to: a. Setup modparams to make the above rtpproxy available. Use set_rtp_proxy_set if you have more than one instance. b. Do the following where you are processing Outbound *initial* invite (i.e. check totag) (Call from UAC to UAS) set_advertised_address("YourPublicIP "); if(!has_totag()) { record_route_preset("YourPublicIP:5060;r2=on;nat=yes","YourPrivateIp:5060;r2=on;nat=yes"); } force_rtp_proxy("ocfaei"," YourPublicIP "); c. Do the following where you are processing Inbound invite (Call from UAS to UAC) if(!has_totag()) { record_route_preset("YourPrivateIP:5060;r2=on;nat=yes", "YourPublicIp:5060;r2=on;nat=yes"); } force_rtp_proxy("ocfaei"); d. Do the following where you are processing replies (180/183/2XX) For Inbound Calls force_rtp_proxy("ocfa","YourPublicIP"); For Outbound Calls force_rtp_proxy("ocfa"); e. Unforce RTPProxy on BYE or Failures. Do not use record_route in the above scenario. If you have it now, remove it or make sure it doesn't get called for this specific case. Explanation: In #b, you are fixing the Via headers to use public IP, setting Record-Route headers so that ACK/RE-INVITE can find way back and are force fixing SDP with public IP for outbound calls. In #c, You are doing same thing except no need to fix Via and the route set is reversed. No need to force fix SDP either since it is figured out automatically. In #d, You are force fixing the SDP to use the public IP for inbound or usual for outbound so that the audio stream from UAS can be relayed. The r2=on parameter is required (as well as enabling double rr, which is default) so that the proxy can deal with the route sets properly Hope the above helps others. SV. Rtpproxy Path 2226,2233c2226 < LM_DBG("Str2 is %s\n", str2); < if(str2) { < newip.s = str2; < LM_DBG("NewIp is %s\n", newip.s); < } < else { < newip.s = (argc < 2) ? str2 : argv[1]; < } --- > newip.s = (argc < 2) ? str2 : argv[1];

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Re: [SR-Users] Kamailio + rtpproxy talking to multiple carrier gateways
by Sarat C. Vemuri 07 Sep '11

07 Sep '11

Ovidiu, Thanks for your time. The "fixes" I pulled in is the latest rr module only. I didn't see anything in that diff to make this work. Could you elaborate a little? However, I was able to get "Outbound" working properly by adding ";r2=on" to the record_route_preset IPs. In going through the source code, I noticed that this is what regular record_route uses to enable double rr and it would automatically take both headers out if it sees that. That seems to have worked. Any pitfals with this? I am still working on "Inbound". For some reason my carrier GW keeps resending invites even after receiving ACK. I need to see if it is an issue with the carrier. Thanks SV. -----Original Message----- From: Ovidiu Sas [mailto:osas@voipembedded.com] Sent: Tuesday, September 06, 2011 9:50 AM To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List Subject: Re: [SR-Users] Kamailio + rtpproxy talking to multiple carrier gateways It seems that something is miss configured on your server. The fixes that I made in the trunk (and you pulled in your local 3.1 repo) were designed to handle the scenario that you are trying to implement. The ACK should be handled properly and routed to the upstream carrier (following the same path as the initial INVITE). Regards, Ovidiu Sas On Mon, Sep 5, 2011 at 5:07 PM, Sarat C. Vemuri <sarat.vemuri(a)fthco.com> wrote: > Again, I apologize for this clumsy way of replying. > > Ovidiu, > > Thanks for the pointer on set_advertised_address. I had to patch rtpproxy module (and rr module for the two parameters to request_route_preset) since I am running 3.1. > > However, I still have a problem with ACKs after following what you suggested. > > INVITE from Internal to Carrier routes properly (two Request-Route headers, one internal IP and other public IP). On 200 OK, the carrier GW properly copies the route set in to Route header. Now the route contains two entries, the public IP and the private IP of Kamailio. The Internal UAC then sends the ACK back to Kamailio. Everything is fine till this point. Now, Kamailio removes the top entry, which is the private IP and then promptly sends the ACK to the public IP of itself!. Of course, that doesn't go anywhere. > > How do I remove the "public IP" entry from the route set before forwarding the reply to Internal UAC? Is there another way to deal with this? I've tried to set an alias= core parameter with the public IP, but doesn't seem to have any effect. The public IP is not reachable from internal network. > > Thanks for your help > > SV. > ------------------------------ > > Message: 3 > Date: Sat, 3 Sep 2011 16:44:22 -0700 > From: Ovidiu Sas <osas(a)voipembedded.com> > Subject: Re: [SR-Users] Kamailio + rtpproxy talking to multiple > carrier gateways - some via Firewall/NAT > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: > > <CAND0Lkt_dpcTm2WKMywMhX6rdsX1ia0r=LYrzB1WfX3oN32Wrg(a)mail.gmail.com> > Content-Type: text/plain; charset=windows-1252 > > It is feasable to what you want: kamailio behind NAT proxying traffic > from/to public internet to/from private network. > You will need to properly craft the INVITE and use proper record route headers. > Use set_advertised_address when needed: > http://kamailio.org/dokuwiki/doku.php/core-cookbook:3.1.x#set_advertis > ed_address Also, use record_route_preset (note that there are two > parameters): > http://kamailio.org/docs/modules/devel/modules_k/rr.html#id2547566 > > That should do it. You don't need any patches for rtpproxy. > Just use force_rtpp_proxy (and force the IP address): > http://kamailio.org/docs/modules/devel/modules/rtpproxy.html#id2546034 > > > Note: Make sure that you understand how in-dialog requests are routed > by a proxy in order to know how to properly handle the initial INVITE. > > > Regards, > Ovidiu Sas > > On Sat, Sep 3, 2011 at 2:53 PM, Sarat C. Vemuri <sarat.vemuri(a)fthco.com> wrote: >> We are trying to configure Kamailio ?(3.1.x) as a ?boarder proxy? >> where it acts as the front for various carrier gateways so that >> internal UACs and UASs are unaware of the carrier gateways. >> >> >> >> Let me try to present a clear picture of our setup. >> >> 1.?????? Kamailio has several NICs (physical or vlan).? Each on a >> different subnet. One subnet is internal/has routes for internal.? >> Other subnets are private connections to carriers or a ?route to public Internet. >> >> 2.?????? All of these subnets are non-routable from Internet. In >> addition , the carrier private connections are not routable internally. >> >> 3.?????? Connection to public internet is via a FW/NAT (one-to-one >> NAT), which maps to one of the interfaces. >> >> 4.?????? All internal? UAC/UAS connect on the internal subnet. >> >> 5.?????? We are using RTPProxy? (at least one instance per carrier) >> to relay media between internal and carrier subnets >> >> >> >> We are able to make this setup up work great except for one >> scenario.? One of the carriers is only reachable via public >> Internet.? Due to security requirements, our Kamailio cannot have a >> public IP address and must use FW/NAT. I guess this scenario is >> ??Proxy behind NAT? and not really encouraged. But I would like to >> see if there is a way to make this work.? We cannot use the >> ?advertised_address? since it changes the IP for every ?route?. >> >> >> >> We were able to get this mostly ?working by doing the following >> >> 1.?????? mhomed=1 >> >> 2.?????? Small patch in the rtpproxy module so that ?force_rtp_proxy >> actually uses the IP address passed (public IP). >> >> 3.?????? Using request_route_preset(?publicIP?) >> >> >> >> The above ?mostly? works.? By that I mean, the INVITE transaction is >> properly passed between internal UAS and carrier SBC and the call is setup. >> However, further transactions (BYE/re-INVITE) etc do not work >> properly. So, far-end hangups are not ?working etc. >> >> >> >> I?ve searched various archives of this and other SER lists looking to >> see if anyone was able to get this scenario working, but couldn?t >> find a definitive answer.? Most of them point to using ?advertised_address?. >> >> >> >> So, and ideas on how to make ?Proxy behind NAT? work without using >> advertised_address?? Am I wasting my time? >> >> >> >> Thanks in advance for any help you can offer. >> >> >> >> SV. >> >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing >> list sr-users(a)lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing > list sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >

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permanent usrloc entries in Kamailio
by Klaus Darilion 07 Sep '11

07 Sep '11

Hi! Recently I had a problem with permanent usrloc entries when upgrading a server from SER 0.9 to Kamailio 3.1. Obviously the problem was that the permanent entries had "flag=128" and now expires=0 is used, see old thread: http://old.nabble.com/usrloc-permanent-records-and-strptime-td17668417.html The strange thing is, that even setting the expiry to 2099-0...... did not worked and Kamailio deleted the "expired" entries. Only setting it to "1970-01-01 01:00:00" worked. Any ideas what could been wrong? Further I wonder why Kamailio uses 01:00:00 for the permanent entry? The server is currently running in timezon CEST (UTC+2), thus IMO 02:00:00 would be more logical. Above thread also mentions problems with day-light-saving changes. Are there really problems? How do you solve those problems? I think the best workaround is to set system time always to UTC. Thanks Klaus

2 2

Siremis User Role permissions
by Camila Troncoso 07 Sep '11

07 Sep '11

Hi, I have a question about the assignment of Role permissions to limit some Users privileges in Siremis. I defined a user , and then a Role denying some actions, but I couldn’t deny the action from de SER menu . The Role permissions only show the modules on Application Menu from Administration Tab . What if I want to deny access to some items inside SER Menu , like modifying the ACL services or /and Rounting services? Regards, *Camila Troncoso **|* Ingeniero de Desarrollo VoissNet S.A. *|*ctroncoso(a)redvoiss.net Santiago - Chile *|* +56 2 2408535 www.redvoiss.net

2 1

Check disconnected users
by Anto 06 Sep '11

06 Sep '11

Hello I want to store in a file access and disconnections to the system. To access (registrar), I thought scheduling a perl script and executed with the perl module, the problem I find with disconnections. Since the user can be disconnected for various reasons, had planned to do a script that runs an SQL table location and according to the expires field, see if disconnection has occurred or not. Would have to run this script in cron every X minutes. Is there a simpler way to check this or have Kamailio some function for it?. Thank you. Best regards

2 2

Re: [SR-Users] Kamailio + rtpproxy talking to multiple carrier gateways
by Sarat C. Vemuri 06 Sep '11

06 Sep '11

Again, I apologize for this clumsy way of replying. Ovidiu, Thanks for the pointer on set_advertised_address. I had to patch rtpproxy module (and rr module for the two parameters to request_route_preset) since I am running 3.1. However, I still have a problem with ACKs after following what you suggested. INVITE from Internal to Carrier routes properly (two Request-Route headers, one internal IP and other public IP). On 200 OK, the carrier GW properly copies the route set in to Route header. Now the route contains two entries, the public IP and the private IP of Kamailio. The Internal UAC then sends the ACK back to Kamailio. Everything is fine till this point. Now, Kamailio removes the top entry, which is the private IP and then promptly sends the ACK to the public IP of itself!. Of course, that doesn't go anywhere. How do I remove the "public IP" entry from the route set before forwarding the reply to Internal UAC? Is there another way to deal with this? I've tried to set an alias= core parameter with the public IP, but doesn't seem to have any effect. The public IP is not reachable from internal network. Thanks for your help SV. ------------------------------ Message: 3 Date: Sat, 3 Sep 2011 16:44:22 -0700 From: Ovidiu Sas <osas(a)voipembedded.com> Subject: Re: [SR-Users] Kamailio + rtpproxy talking to multiple carrier gateways - some via Firewall/NAT To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Message-ID: <CAND0Lkt_dpcTm2WKMywMhX6rdsX1ia0r=LYrzB1WfX3oN32Wrg(a)mail.gmail.com> Content-Type: text/plain; charset=windows-1252 It is feasable to what you want: kamailio behind NAT proxying traffic from/to public internet to/from private network. You will need to properly craft the INVITE and use proper record route headers. Use set_advertised_address when needed: http://kamailio.org/dokuwiki/doku.php/core-cookbook:3.1.x#set_advertised_ad… Also, use record_route_preset (note that there are two parameters): http://kamailio.org/docs/modules/devel/modules_k/rr.html#id2547566 That should do it. You don't need any patches for rtpproxy. Just use force_rtpp_proxy (and force the IP address): http://kamailio.org/docs/modules/devel/modules/rtpproxy.html#id2546034 Note: Make sure that you understand how in-dialog requests are routed by a proxy in order to know how to properly handle the initial INVITE. Regards, Ovidiu Sas On Sat, Sep 3, 2011 at 2:53 PM, Sarat C. Vemuri <sarat.vemuri(a)fthco.com> wrote: > We are trying to configure Kamailio ?(3.1.x) as a ?boarder proxy? where it > acts as the front for various carrier gateways so that internal UACs and > UASs are unaware of the carrier gateways. > > > > Let me try to present a clear picture of our setup. > > 1.?????? Kamailio has several NICs (physical or vlan).? Each on a different > subnet. One subnet is internal/has routes for internal.? Other subnets are > private connections to carriers or a ?route to public Internet. > > 2.?????? All of these subnets are non-routable from Internet. In addition , > the carrier private connections are not routable internally. > > 3.?????? Connection to public internet is via a FW/NAT (one-to-one NAT), > which maps to one of the interfaces. > > 4.?????? All internal? UAC/UAS connect on the internal subnet. > > 5.?????? We are using RTPProxy? (at least one instance per carrier) to relay > media between internal and carrier subnets > > > > We are able to make this setup up work great except for one scenario.? One > of the carriers is only reachable via public Internet.? Due to security > requirements, our Kamailio cannot have a public IP address and must use > FW/NAT. I guess this scenario is ??Proxy behind NAT? and not really > encouraged. But I would like to see if there is a way to make this work.? We > cannot use the ?advertised_address? since it changes the IP for every > ?route?. > > > > We were able to get this mostly ?working by doing the following > > 1.?????? mhomed=1 > > 2.?????? Small patch in the rtpproxy module so that ?force_rtp_proxy > actually uses the IP address passed (public IP). > > 3.?????? Using request_route_preset(?publicIP?) > > > > The above ?mostly? works.? By that I mean, the INVITE transaction is > properly passed between internal UAS and carrier SBC and the call is setup. > However, further transactions (BYE/re-INVITE) etc do not work properly. So, > far-end hangups are not ?working etc. > > > > I?ve searched various archives of this and other SER lists looking to see if > anyone was able to get this scenario working, but couldn?t find a definitive > answer.? Most of them point to using ?advertised_address?. > > > > So, and ideas on how to make ?Proxy behind NAT? work without using > advertised_address?? Am I wasting my time? > > > > Thanks in advance for any help you can offer. > > > > SV. > > > > _______________________________________________ > 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 > >

3 2

KAMAILIO RTP PROXY
by Phillman25 Kyriacou 06 Sep '11

06 Sep '11

Hi I want to use Kamailio to handle both signalling and media always. My scenario is the following PSTN==>KAMAILIO==>ASTERISK. I have a dialplan that points to ASTERISK when a call comes into Kamailio. Signalling is fine, however i cant seem to get the Media to work, i have installed the RTPPROXY as suggested by RTPPROXY module and have implemented it into the routing logic, however i have one way audio. The PSTN side cannot hear anything. I dont use the Kamailio for sip registrations or for users behind NAT. I just use a simple dialplan for forwarding SIP messages to our certain VOIP carriers, however some of our carriers request that we use both signalling and media from the same IP thats why i need to implement this scenario. Is this possible? Thanking you in advance for your help! Phillip

2 1

db_mysql "Commands out of sync; you can't run this command now "
by MÉSZÁROS Mihály 06 Sep '11

06 Sep '11

Hi, I am using the git version of kamailio, and i am experiencing problem This message is repeated in log with all kind of mysql query (insert/delete/select) Aug 27 22:04:05 hal /usr/sbin/kamailio[25681]: ERROR: db_mysql [km_dbase.c:124]: driver error on query: Commands out of sync; you can't run this command now Aug 27 22:04:05 hal /usr/sbin/kamailio[25681]: ERROR: <core> [db_query.c:103]: error while submitting query Aug 27 22:04:05 hal /usr/sbin/kamailio[25681]: ERROR: usrloc [udomain.c:570]: db_query failed Aug 27 22:04:05 hal /usr/sbin/kamailio[25681]: ERROR: db_mysql [km_dbase.c:124]: driver error on query: Commands out of sync; you can't run this command now Aug 27 22:04:05 hal /usr/sbin/kamailio[25681]: ERROR: <core> [db_query.c:186]: error while submitting query Aug 27 22:04:05 hal /usr/sbin/kamailio[25681]: ERROR: usrloc [ucontact.c:513]: inserting contact in db failed Aug 27 22:04:05 hal /usr/sbin/kamailio[25681]: ERROR: usrloc [urecord.c:474]: failed to insert in database Aug 27 22:04:05 hal /usr/sbin/kamailio[25681]: ERROR: registrar [save.c:430]: failed to insert contact Aug 27 22:04:05 hal /usr/sbin/kamailio[25681]: ERROR: db_mysql [km_dbase.c:124]: driver error on query: Commands out of sync; you can't run this command now Aug 27 22:04:05 hal /usr/sbin/kamailio[25681]: ERROR: <core> [db_query.c:242]: error while submitting query Aug 27 22:04:05 hal /usr/sbin/kamailio[25681]: ERROR: usrloc [urecord.c:426]: failed to delete from database Aug 27 22:04:05 hal /usr/sbin/kamailio[25681]: ERROR: usrloc [udomain.c:896]: DB delete failed I am using stable debian so db_mysql.so linked with libmysqlclient.so.16 I read mysql doc and it is saying about this error the following: "Commands out of sync; you can't run this command now" http://dev.mysql.com/doc/refman/5.0/en/commands-out-of-sync.html So this problem cause it that not freed the last result. Only one assumption: Could it related to this change: http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=commitdiff;h=df… Any help appreciated. Regards, Misi

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Re: [SR-Users] Kamailio + rtpproxy talking to multiple carrier gateways
by Sarat C. Vemuri 05 Sep '11

05 Sep '11

Alex, Sorry about this clumsy way of replying, but I am stuck with Outlook 2010 and Exchange. You stated the problem correctly. Carrier GWs (UAS) and internal clients (UAC) cannot reach each other directly (TCP/UDP). They have to go through Kamailio and rtpproxy. I am goint to reply to Ovidiu as well with some more details. Thanks SV. ------------------------------ Message: 2 Date: Sat, 03 Sep 2011 19:12:38 -0400 From: Alex Balashov <abalashov(a)evaristesys.com> Subject: Re: [SR-Users] Kamailio + rtpproxy talking to multiple carrier gateways - some via Firewall/NAT To: sr-users(a)lists.sip-router.org Message-ID: <4E62B466.20808(a)evaristesys.com> Content-Type: text/plain; charset=windows-1252; format=flowed On 09/03/2011 05:53 PM, Sarat C. Vemuri wrote: > 3.Using request_route_preset(?publicIP?) > > The above ?mostly? works. By that I mean, the INVITE transaction is > properly passed between internal UAS and carrier SBC and the call is > setup. However, further transactions (BYE/re-INVITE) etc do not work > properly. So, far-end hangups are not working etc. So in other words, there is no interface to which Kamailio could be bound on a network that both the UAC and the UAS have direct network and transport-layer reachability to? If I understand correctly, it sounds like the fundamental problem here is that the Record-Route URI introduced by the proxy in handling the initial INVITE request has to be conserved in all subsequent requests and replies. It is a requirement to use the same route set in sequential (in-dialog) requests such as reinvites or BYEs. It is also mandatory for the UAS to copy the received Record-Route into the final reply (i.e. 200 OK). The problem is that you then have to choose the network address in the Record-Route. If either the UAC or the UAS cannot directly reach it, your problem arises. Is this an accurate statement of the problem? I wanted to verify before continuing. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

1 0

SIP Analysis & Testing tools
by Vladimir Broz 05 Sep '11

05 Sep '11

Hi, for sake of Vaclav's courage, let me introduce the "SIP Analysis & Testing tools". This is a set of very powerful utilities that may help you to do SIP call flows analysis. It can do SIP dialogs and transactions matching and analysis in large pcaps, create pretty call flow diagrams, check and verify SIP message content (headers, parameters,..) and many more.... It's available on: https://github.com/vkubart/sipat For more details, please ask Vaclav Kubart (vaclav.kubart(a)iptel.org). SIP Analysis & Testing tools ============================ This project offers set of tools for testing SIP. Dependencies (ubuntu packages) ============ libosip2-dev libpcap-dev libxml2-dev xsltproc cfanal ====== Call flow analyzer able to draw call flows from PCAP and verify callflows against given definition. Usage: cfanal -r captured-traffic.pcap -a 10.38.14.73=uac -a 10.38.14.74:5062=uas -print-cf drawing.cf cf2svg drawing.cf drawing.svg svg2png drawing.svg drawing.png callflow_diagrams ================= XSL template for painting call flow diagrams. ..... Thank you Vaclav a lot! -Vlada

4 4

question
by Allen D. Ford 05 Sep '11

05 Sep '11

Where are the instructions for this gui Im trying to setup a sip load balancer and rtp proxy for nating "Success is not measured by the positions you reach in life but by the obstacles you overcome -B.T.W" [Description: Description: Description: Description: Description: cyford]<http://www.cyfordtechnologies.com/>Director Of Technology Allen D. Ford 4290-106 Bells Ferry Road, Suite 20 , Kennesaw , GA 30144 Tel: 404-424-8270: Fax: 404-424-8271 Email: allen(a)cyfordtechnologies.com<mailto:allen@cyfordtechnologies.com> Web Address: http://www.cyfordtechnologies.com <http://www.cyfordtechnologies.com/> LinkedIn: http://www.linkedin.com/in/cyford FaceBook: http://www.facebook.com/profile.php?id=100002260975450 NOTICE: This email message is for the sole use of the intended recipient(s) and may contain confidential and privileged information. Any unauthorized review, use, disclosure, or distribution is prohibited. If you are not the intended recipient, please destroy all copies of the original message and notify the sender so that our address record can be corrected. Thank you

2 1

10 Years SER - Awards
by Daniel-Constantin Mierla 05 Sep '11

05 Sep '11

Hello, a surprise part of 10 Years SER Conference was granting *special* awards to the people attending the event that had a major contribution in the management and development of SER/Kamailio to date. Most of them are well known in the community, but not all of them. Hope you like them, it proved to be a very entertaining moment. Here is the pdf with the awards: * http://asipto.com/u/48 The plan is to extend it with the other people in our community as ideas with award titles pop up. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

10 Years SER - Follow UP
by Daniel-Constantin Mierla 05 Sep '11

05 Sep '11

Hello, first I want to thank to all participants at 10 Years SER event in Berlin last Friday. We spent great time, at least I did! We accepted 50 visiting participants to match the maximum capacity of the conference room, being full booked about 2 weeks before. Just few people didn't show up and didn't inform us at all in advance, which was a pity over all since several persons that registered later were not accepted, thus some of them could have attended. Folks at Fokus working with SER/Kamailio joined us and we had probably one of the very few occasions so far to get so dense knowledge of SIP/VoIP in the same place, making a day of interesting presentations and open discussions. Accompanied by nice weather in Berlin, the evening at Playa Paradiso on river Spree offered a great place for continuing the technical and mufti-cultural debates (I could count attendees from about 15 countries (including Brazil!)). Many thanks again to our sponsors for free drinks and food at this event: Sipgate (http://www.sipgate.de), Amooma (http://www.amooma.de), NG-Voice (http://www.ng-voice.com), Asipto (http://www.asipto.com), Frafos (http://www.frafos.com) and Fokus Institute (http://www.fokus.fraunhofer.de). I will upload the presentations soon and send an update here -- there were 15 talks, covering from technical concepts to business matters and open source philosophy. Therefore if you were a speaker and see this message, please send the pdf slides to me. Do not forget, you can join the next party for this celebration this week in Vienna, details at: * http://sip-router.org/10-years-ser-vienna/ Meanwhile, be sure you start/continue testing the upcoming release v3.2.0, we have to be sure it is stable to release it and start a new development cycle in the near future, to be able to present new features at the 11th celebration :-) . Here is a step by step guide to pull the code from GIT and install: * http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-devel-from-git Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

2 1

tls + tcp problems
by MÉSZÁROS Mihály 05 Sep '11

05 Sep '11

Hi, I am experiencing two problem: First: I am experiencing that tls read buffer overflow, if i use default value of tcp_rd_buf_size If i doubled the size of read buffer and i use tcp_rd_buf_size=8192 then the problem seems to be solved. Second: I am experiencing that TLS connection is dropped by sip proxy very frequently! And i think the tcp connection shouldn't dropped! So my guess is that TLS communication is not restarting the timeout counters! So in every 2 minutes the sip-router is restarting the TCP connection with tcp RST. like this: Sep 5 11:28:48 hal /usr/sbin/kamailio[8129]: DEBUG: <core> [tcp_main.c:4304]: tcp_main: entering timer for 0xac537c74 (ticks=446299809, ti meout=446299809 (0 s), wr_timeout=0 (240541717 s)), write queue: 0 bytes Sep 5 11:28:48 hal /usr/sbin/kamailio[8129]: DEBUG: <core> [tcp_main.c:4346]: tcp_main: timeout for 0xac537c74 Sep 5 11:28:48 hal /usr/sbin/kamailio[8129]: DEBUG: <core> [io_wait.h:617]: DBG: io_watch_del (0x824cf80, 48, -1, 0x10) fd_no=41 called Sep 5 11:28:48 hal /usr/sbin/kamailio[8129]: DEBUG: tls [tls_server.c:597]: Closing SSL connection 0xac4a0374 Sep 5 11:28:48 hal /usr/sbin/kamailio[8129]: DEBUG: <core> [ip_addr.c:247]: tcpconn_new: new tcp connection: 10.10.10.10 Sep 5 11:28:48 hal /usr/sbin/kamailio[8129]: DEBUG: <core> [tcp_main.c:1081]: tcpconn_new: on port 36142, type 3 Sep 5 11:28:48 hal /usr/sbin/kamailio[8129]: DEBUG: <core> [tcp_main.c:1382]: tcpconn_add: hashes: 2473:3781:3367, 5 This second problem is not solved and very urgent for me. Any help suggestion is highly appreciated! Thanks, Misi

2 1

Carrier module for kamailio
by Mino Haluz 05 Sep '11

05 Sep '11

Hi, I need a module which could allow me to send traffic to various carriers and it has to support some important features. So some basic ones: - possibility to re-route the call in case the original route fails - peak/offpeak conditions (time-based) - route traffic according to prefix I found LCR and carrierroute module, but it does not have peak/offpeak feature. Correct me if I am wrong..

2 1

new developer: Sven Knoblich
by Henning Westerholt 05 Sep '11

05 Sep '11

Hi all, I'm happy to announce a new developer for the sip-router project: Sven Knoblich. Sven is an experienced C/C++ developer in our offices in Karlsruhe and since about a year part of the team that develops our VoIP backend services. Past contributions of him were the CDR based accounting extensions in the acc module and several other bugfixes and extensions for the dialog module. He'll maintain in the future this new acc and dialog functionality and also other code contributed from us. Best regards, Henning -- Henning Westerholt - Head of IT Operations Internet Access & Communications 1&1 Internet AG, Brauerstraße 48, 76135 Karlsruhe, Germany

3 2

Pass all rtp to RTPProxy
by Stas Bakulin 05 Sep '11

05 Sep '11

Hello! What is it right way to pass all rtp traffic through RTPProxy? How this can be configured in Kamailio? Stas Bakulin ---------------------------------------------------------------------- ogion(a)kvant-telecom.ru | www.kvant-telecom.ru Tel: +7 473 233 0330 (128)

2 1

Kamailio + rtpproxy talking to multiple carrier gateways - some via Firewall/NAT
by Sarat C. Vemuri 03 Sep '11

03 Sep '11

We are trying to configure Kamailio (3.1.x) as a "boarder proxy" where it acts as the front for various carrier gateways so that internal UACs and UASs are unaware of the carrier gateways. Let me try to present a clear picture of our setup. 1. Kamailio has several NICs (physical or vlan). Each on a different subnet. One subnet is internal/has routes for internal. Other subnets are private connections to carriers or a route to public Internet. 2. All of these subnets are non-routable from Internet. In addition , the carrier private connections are not routable internally. 3. Connection to public internet is via a FW/NAT (one-to-one NAT), which maps to one of the interfaces. 4. All internal UAC/UAS connect on the internal subnet. 5. We are using RTPProxy (at least one instance per carrier) to relay media between internal and carrier subnets We are able to make this setup up work great except for one scenario. One of the carriers is only reachable via public Internet. Due to security requirements, our Kamailio cannot have a public IP address and must use FW/NAT. I guess this scenario is "Proxy behind NAT" and not really encouraged. But I would like to see if there is a way to make this work. We cannot use the "advertised_address" since it changes the IP for every "route". We were able to get this mostly working by doing the following 1. mhomed=1 2. Small patch in the rtpproxy module so that force_rtp_proxy actually uses the IP address passed (public IP). 3. Using request_route_preset("publicIP") The above "mostly" works. By that I mean, the INVITE transaction is properly passed between internal UAS and carrier SBC and the call is setup. However, further transactions (BYE/re-INVITE) etc do not work properly. So, far-end hangups are not working etc. I've searched various archives of this and other SER lists looking to see if anyone was able to get this scenario working, but couldn't find a definitive answer. Most of them point to using "advertised_address". So, and ideas on how to make "Proxy behind NAT" work without using advertised_address? Am I wasting my time? Thanks in advance for any help you can offer. SV.

3 2

10 years of SER - statistics
by Daniel-Constantin Mierla 31 Aug '11

31 Aug '11

Hello, it is now one month till the 10 years SER event in Berlin. I am thinking to collect few statistics about usage of SER-based code at this moment (SER, OpenSER, Kamailio, OpenIMSCore ...) and present them during the conference and publish on the web. Everything is going to be like a combined report, not individual listing. Therefore, if you want to participate, your name or company won't be mentioned. Here are the stats I thought of: - type of usage: production, evaluation (testing), research - number of subscribers (phone lines) - percentage of phones behind NAT - number of calls per month - number of call minutes per month I am not looking for exact numbers, but rough estimation (e.g., about 10 000 phone lines, ...). Of course, some of the metrics don't apply always (e.g., if you do termination routed through SIP server, you don't have subscribers, but just calls traffic). Those doing deployments, can make one summary of statistics for all instances. You may send other statistics you think they worth publishing. Again, this is voluntary, naming is not required. I hope only those giving real number will write back -- I will try anyhow to figure out if someone is just dumping fantasy numbers. If privacy is really a big concern, from case to case, everything can be done under NDA. Thanks, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

2 3

Resend INVITE in case of failure
by Mino Haluz 31 Aug '11

31 Aug '11

Hi, I would like to resend the INVITE to another gateway if the original fails (503, 500, etc.). Is there any module or mechanism which could do this ? Mino

4 3

Celebrating 10 years SER SIP Router in Vienna: 8th September 2011
by Klaus Darilion 31 Aug '11

31 Aug '11

Hi! If you haven't noticed yet, SER (the mother of the SIP proxy projects Openser, Kamailio, sip-router, opensips, ....) is celebrating their 10th year. There will be a main event happening in Berlin (http://sip-router.org/10-years-ser/). For those who can not travel to this event, there will be some smaller events around the world. In Vienna, this will happen on the 8th September at 7pm in the evening, organized by Andreas Granig and myself. We do not plan to have a big event with speaker, it is just an informal come together to meet with other SER-interested[1] guys, having some techie talks, maybe getting some business contacts and drinking some beers. Drinks will be sponsored by http://www.ipcom.at/ and http://www.sipwise.com/. The event will happen in the city center of Vienna. The location is not fixed yet, it depends on how many people are going to attend. Thus, please let us know if you going to attend (send us a short e-mail) and spread this e-mail to interested people you know. Cheers Andreas and Klaus [1] User's of all SER-flavors are welcome

1 1

Kamailio setup help - needed in Hamburg / Germany
by Rene Bormann 30 Aug '11

30 Aug '11

Hello SR List, our background is classic telephony with E1 interfaces, active telephony-cards etc. We now got our telephone-plattform transformed for SIP, but all behind a firewall. Actually we receive calls from some carriers throu the firewall on our Asterisk, which has E1 links to the IVR -Plattform. Now we want to eliminate Asterisk as kinda internal PSTN gateway and direcly route the calls throu Kamailio to our IVR services. So we thought, hey Kamailio can't be that complicated to setup, but now we have to admit - that we are kinda "LOST". We are located in Hamburg / Germany and need HELP. Is on this list anyboy - who can help us, maybe from around Hamburg? Get us on the kamailio train or maybe help us on a continous bases? Thanks from Hamburg, René

2 1

Kamailio 3.1.1 snmpstat help!!
by Stephen Dodge (Bistech) 30 Aug '11

30 Aug '11

Hello, I am having difficulty running the snmpstats module. We are running CentOS 5.6 and Kamailio version 3.1.1 (i386/linux). I've tried a fresh install of Kamailio, installing net-snmp from source but I can't seem to get it running - I've seen a couple of posts regarding the issue but nothing seems to work for us!! The issue I get when I start Kamailio is: 0(27523) ERROR: <core> [sr_module.c:523]: ERROR: load_module: could not open module </usr/local/lib/kamailio/modules_k/snmpstats.so>: /usr/lib/libnetsnmpagent.so.10: undefined symbol: hosts_ctl Any help would be much appreciated. Also, if anyone has any suggestions on how best to get either Solarwinds or PRTG to monitor our Kamailio servers that would be good. Thanks. Stephen Dodge Visit www.bistech.co.uk<http://www.bistech.co.uk/> for news and information Regards Stephen Dodge Visit www.bistech.co.uk<http://www.bistech.co.uk/> for news and information ________________________________ Information in this message, including any attachments, is confidential to the person to whom it is addressed and may be legally privileged. If you are not the intended recipient please notify the sender and delete the message from your system. Please note that Bistech Group plc, Bistech plc, Bisnet Limited and the sender do not accept any responsibility for viruses. It is your responsibility to check the e-mail and any attachments for viruses. Calls may be monitored and recorded.

2 4

registered("location") not working
by Skyler 30 Aug '11

30 Aug '11

Hi all, I am attempting to only reply to NOTIFY when the request is originating from a user/device that is authenticated on kamailio already. There are 2 ways I've tried (below) but neither will work. Any idea on what I am doing wrong or suggestions on how to do it right? TIA, Skyler if(is_method("NOTIFY") && is_registered("location")) { route(ROUTE_NOTIFY); } OR .... if(is_method("NOTIFY")) { # We only reply OK if coming from authenticated users or trusted peers if(!registered("location") || !from_any_gw("$avp(s:ip)")) { sl_send_reply("503", "Service Unavailable"); exit; } xlog("L_INFO", "Friendly Peer Poke - M=$rm F=$fu R=$ru\n"); sl_send_reply("200", "OK"); exit; }

2 3

SEMS 1.4.2 released
by Stefan Sayer 29 Aug '11

29 Aug '11

(apologies if you receive multiple copies of this mail) Hello, the SIP Express Media Server version 1.4.2 is available. Users of SEMS 1.4 versions and anyone else is recommended to upgrade to that version. Please find below the relevant changelog, the source at ftp://ftp.iptel.org/pub/sems/sems-1.4.2.tar.gz and Debian/Ubuntu packages as always at OBS https://build.opensuse.org/project/show?project=home%3Ateam-sems Many thanks to everyone who contributed with testing, bug fixes, documentation and feedback. * SEMS 1.4.2 release * auth'ed BYE (wait_for_bye_transaction) * fixes SIP auth for qop header format * xmlrpc: fix busyloop with keep-alive * a few minor SST issues * builds on Ubuntu 11.4 (build-deps) * ivr: release GIL on blocking file I/O * SBC: fix codec filter for unnamed payloads<96 * fixed DSM variables to outgoing call * some examples and documentation added This release introduces a new configuration option wait_for_bye_transaction=yes which enables authentication of BYE requests sent by SEMS, which is disabled by default. If this option is not enabled, SEMS behaves like in 1.4.1. With Best Regards Stefan Sayer for the SEMS community -- frafos.com

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Re: [SR-Users] Kamailio as simple SIP proxy between client and SIP server.
by Edson - Lists 29 Aug '11

29 Aug '11

Hi, Henrik... Please, never, ever, forget to copy the list in a thread initiated there... other may be "curious" on it... ;) About your configuration, it seems that you just need a 'simple forward' server. It's very simple to accomplish... Make a simple forward configuration script (you can start with the provided one and cut all registration and whistles). Just be aware of all security issues binded to this choice. Ok... you still can load any of the control modules (pike, p.ex.) and have some prevention, but even so it's not your best choice. Edson. Em 29/08/2011 12:13, Henrik Aagaard Sørensen escreveu: > Hi Edson. > > I do know the different definitions as SIP-Proxy, SIP-Server etc. (I hope). > > That's why I want Kamailio to act as a SIP-Proxy, not a SIP-Server, as > it does currently. > > As I've understood Kamailio and SIP-Router is pretty much the same > thing. Going to sip-router.org <http://sip-router.org> for downloads > also refers to Kamailio. > > So shouldn't it be possible to use Kamailio as SIP-Proxy? Or have I > misunderstood that. > > I do not want my Kamailio to take registrations etc. It should just > forward SIP messages between client and SIP-Server, as I'm not allowed > to register towards my current SIP-Server with my current IP. But my > server, where the Kamailio is running, is allowed with it's IP. > > 2011/8/29 Edson - Lists <4lists(a)gmail.com <mailto:4lists@gmail.com>> > > Hi, Henrik... > > Sip-Router is "just" a SIP Proxy... but a dam good one... > > You have to take a step back... You are mixing concepts when > affirming that SR is playing SIP-Server roles... Please take a > better look on the definitions (SIP-Proxy, SIP-Server, Registrar, > etc) and find wich system play each role. > > In your actual scenario they are mixed concepts. > > Draw your target scenario with all the alternatives. You will find > many of them. Only them you will be able to point the roles each > component can and will have to play in the chosen scenario. Each of > this "play alternatives" (roles played by each component) will give > You Pros and Cons (security, scalability, velocity, etc, etc, etc) ... > > Make Your choice, try to implement it and this list will be able to > help you when problems on SR (SER or KAMAILIO flavors) configuration > come to play. > > Edson. > > Em 29/08/2011 11:46, Henrik Aagaard Sørensen escreveu: > > This is my current setup, which works: > SIP client <-> SIP server. > > As I've currently moved to another country I would like to be > able to do > the following setup: > SIP client <-> Kamailio SIP proxy <-> SIP server. > > As the SIP server only accept packets from specific IP's. > > Is it possible? > > I've managed to setup a simple Kamailio installation on my > server, which > does have a valid IP for the SIP server. > With the Kamailio installation I've also enabled NAT with RTPProxy. > > My problem is now, that the Kamailio installation doesn't work as a > simple proxy, but actually takes registrations etc. and acts as a > "full-blown" SIP server. > > It should only take SIP messages and transfer them between > clients and > server. > > Any ideas? > > > _________________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users > mailing list > sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> > http://lists.sip-router.org/__cgi-bin/mailman/listinfo/sr-__users <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> > >

1 0

Kamailio as simple SIP proxy between client and SIP server.
by Henrik Aagaard Sørensen 29 Aug '11

29 Aug '11

This is my current setup, which works: SIP client <-> SIP server. As I've currently moved to another country I would like to be able to do the following setup: SIP client <-> Kamailio SIP proxy <-> SIP server. As the SIP server only accept packets from specific IP's. Is it possible? I've managed to setup a simple Kamailio installation on my server, which does have a valid IP for the SIP server. With the Kamailio installation I've also enabled NAT with RTPProxy. My problem is now, that the Kamailio installation doesn't work as a simple proxy, but actually takes registrations etc. and acts as a "full-blown" SIP server. It should only take SIP messages and transfer them between clients and server. Any ideas?

2 1

Hunting Groups In SIP-Router
by afshin afzali 29 Aug '11

29 Aug '11

Hi Guys, I'm newbie in SER. I need to implement a hunting group service in SER. After some digging in the modules I could only find AVPOPS which seems to be able to use. I'll appreciate every advise on this. BEST, -- afshin

2 1

enum_pv_query
by Andrew Melton 28 Aug '11

28 Aug '11

I would like to construct an enum query using more than one pseudo variable. The result will be a non-e.164 string which is ok for this application. Ideally, it would be something like: enum_pv_query("$rU + $rp"); This would result in the username from the R-URI appended to the port of the R-URI. However, this doen't seem to work and I wind up with 1.2.1.2.5.5.5.2.1.2.1.$rp.e164.arpa. Instead of using the $rp pseudo variable, the string "$rp" is used to form the query. It should be: 1.2.1.2.5.5.5.2.1.2.1.*0.6.0.5*.e164.arpa. where the port 5060 is inserted between the R-URI username and the domain suffix. Is there a different or better way to use the variables within enum_pv_query to construct a query? Thanks.

4 3

Persistent dialog
by Jon Bonilla 27 Aug '11

27 Aug '11

Hi all I'm playing with the dialog module to count the calls and limit the number of calls per subscriber. After configured the proxy I've tested a call and I realize that the dialog list is not empty after the call ends. Here's a sample call log and the dialog list while the call was being stablished and 3 minutes after that: root@proxy1a:~# grep "fbrzjkplmvlauty@quenya" /var/log/kamailio.log New request - M=INVITE Dialog set mark TOTAL Proxy authentication failed New request - M=INVITE Dialog set mark TOTAL Dialog set mark user and userout to 'd5a78dafcb93' Dialog set mark type to outgoing Call to PSTN Dialog set mark peer and peerout to '4' Relaying request Request leaving server NAT-Reply - S=100 - Connecting NAT-Reply - S=488 - Not Acceptable Here Failure route for PSTN call Unset dialog mark peer and peerout '4' Filter reply code No failover routing needed for this response code New request - M=ACK While the call was being stablished or just after it was dropped (quickly) sercmd> dlg.list hash:3837:638039689 state:1 timestart:0 timeout:0 callid:fbrzjkplmvlauty@quenya from_tag:inobo to_tag: from_uri:sip:123456789@labsip.com to_uri:sip:0043987654321@labsip.com caller_contact:sip:123456789-sbhn1naovm6bc@111.111.111.164:5060;transport=udp caller_cseq:149 caller_route_set:<sip:111.111.111.135;r2=on;lr=on;ftag=inobo;nat=yes>,<sip:111.111.111.100;r2=on;lr=on;ftag=inobo;nat=yes> callee_contact: callee_cseq: callee_route_set: caller_bind_addr:udp:111.111.111.138:5062 callee_bind_addr: hash:3837:638039690 state:5 timestart:0 timeout:0 callid:fbrzjkplmvlauty@quenya from_tag:inobo to_tag: from_uri:sip:123456789@labsip.com to_uri:sip:0043987654321@labsip.com caller_contact:sip:123456789-sbhn1naovm6bc@111.111.111.164:5060;transport=udp caller_cseq:150 caller_route_set: <sip:111.111.111.135;r2=on;lr=on;ftag=inobo;nat=yes>,<sip:111.111.111.100;r2=on;lr=on;ftag=inobo;nat=yes> callee_contact: callee_cseq: callee_route_set: caller_bind_addr:udp:111.111.111.138:5062 callee_bind_addr: After a couple of minutes: hash:3837:638039689 state:1 timestart:0 timeout:0 callid:fbrzjkplmvlauty@quenya from_tag:inobo to_tag: from_uri:sip:123456789@labsip.com to_uri:sip:0043987654321@labsip.com caller_contact:sip:123456789-sbhn1naovm6bc@111.111.111.164:5060;transport=udp caller_cseq:149 caller_route_set: <sip:111.111.111.135;r2=on;lr=on;ftag=inobo;nat=yes>,<sip:111.111.111.100;r2=on;lr=on;ftag=inobo;nat=yes> callee_contact: callee_cseq: callee_route_set: caller_bind_addr:udp:111.111.111.138:5062 callee_bind_addr: sercmd> dlg.profile_get_size total Profile:total => profile:total value: count:1 sercmd> dlg.profile_get_size user Profile:user => profile:user value: count:0 sercmd> dlg.profile_get_size userout Profile:userout => profile:userout value: count:0 sercmd> dlg.profile_get_size peer Profile:peer => profile:peer value: count:0 sercmd> dlg.profile_get_size peerout Profile:peerout => profile:peerout value: count:0 Obviously, there's no such a call. But my dialog profile TOTOAL say's there's one, which is listed by sercmd. The dialog is persistent, I guess it will be destroyed after 12 hours, but I'd like to know why it's still there if I received a negative response and the ACK for it. Thanks, Jon

3 13

insert_hf in branch_route
by Gustavo Garcia Bernardo 26 Aug '11

26 Aug '11

Hi, I'm trying to add some headers in a branch_route making use of insert_hf method: insert_hf("Route: <sip:10.10.10.10:5080;lr=on>\r\n"); But the forwarded message doesn't include that new header. According to the doc, insert_hf can be used in a branch_route, Am I missing something? I'm using Kamailio 3.1. G. ________________________________ Este mensaje se dirige exclusivamente a su destinatario. Puede consultar nuestra pol?tica de env?o y recepci?n de correo electr?nico en el enlace situado m?s abajo. This message is intended exclusively for its addressee. We only send and receive email on the basis of the terms set out at. http://www.tid.es/ES/PAGINAS/disclaimer.aspx

2 2

Re: [SR-Users] How to get mem_summary (SIGUSR1) on a worker process?
by Andrade Ricardo (CI/AFU1) 26 Aug '11

26 Aug '11

Hi, I'd just like to say that I found the answer to my own question in the documentation: With 3.x versions, you have to load ctl and cfg_rpc modules and execute: sercmd cfg.set_now_int core mem_dump_pkg <pid> http://www.kamailio.org/dokuwiki/doku.php/troubleshooting:memory I did also write a patch to suppress the full memory dump, because I am only interested in the first lines of the status (total usage). This way I the server doesn't get blocked - full memory dump usually takes a VERY long time. Patch is attached and I'd be glad if something similar would be integrated in the trunk. I've introduced a mem_summary option "16" to control that behavior. Best regards, Ricardo.

2 2

issue with tm callbacks / premature cancel
by Andrew Pogrebennyk 26 Aug '11

26 Aug '11

I've got stuck today trying to implement some sequential forking scenario. When the call comes destined to the local user, I'm trying to reach it sequentially through two alternative extensions. For each extension, I'm calling t_set_fr with a timeout of 15sec. They reply only 180 and do not pick up the phone. After that the call should go to the "last resort" PSTN number. I'm calling then t_set_fr with a timeout of 30sec. But immediately after sending INVITE to PSTN gw Kamailio generates cancel and sends 408 Request Timeout to the initial INVITE. In the debug log I'm seeing this: Aug 1 17:34:43 localhost /usr/local/sbin/kamailio[28488]: DEBUG: tm [t_hooks.c:288]: DBG: trans=0xb60f543c, callback type 1048576, id 0 entered Aug 1 17:34:43 localhost /usr/local/sbin/kamailio[28488]: DEBUG: tm [t_reply.c:1634]: DEBUG: relay_reply: branch=1, save=1, relay=-1 Aug 1 17:34:43 localhost /usr/local/sbin/kamailio[28488]: DEBUG: tm [t_cancel.c:328]: DEBUG: cancel_branch: sending cancel... Does it look like something familiar to you? Kamailio version: 3.1.4 Config of tm module: modparam("tm", "fr_timer", 20000) modparam("tm", "fr_inv_timer", 120000) modparam("tm", "restart_fr_on_each_reply", 1) The rest of config is really simple, I can upload it somewhere if it's necessary. -- Sincerely, Andrew Pogrebennyk

2 5

Problems with pseudo-variables
by Rocío Ruz 26 Aug '11

26 Aug '11

Hello, I am using kamailio ims extensions and I am observing some problems with pseudo-variable handling. On the s-cscf the pseudo-variables are not 'translated' into the real values. When using xlog like this: xlog("L_ERR", " found pai: $ai time [$Tf] \n"); I am finding : tail -f /var/log/syslog | grep found invlcscs /usr/sbin/kamailio[21689]: ERROR: <script>: found pai: $ai time [$Tf] However on the p-cscf the same case is correctly treated: xlog("L_ERR", " found pai: $ai time [$Tf] \n"); invlcpcs /usr/sbin/kamailio[20540]: ERROR: <script>: found pai: sip:0188888881@open-ims.test;transport=UDP time [Fri Aug 19 14:05:44 2011] I am using 3.1 kamalio version and I have added pv and xlog modules on the s-cscf. Why is the S-CSCF unable to read pseudo-variables?Could anybody please help me? Thank you. RR.

2 1

Status of t_relay()
by Fabian Borot 25 Aug '11

25 Aug '11

Hi, If t_relay() fails, is it possible to retrieve the status/reason on a variable or something? I am using t_grep_status("code") and trying to catch each case that I am interested but I was wondering if there was a better way. One could do something like this: If (result > 400) { do action 1 } else { do action 2 } txs a lot fborot

3 5

report
by zmailer＠nic.funet.fi 25 Aug '11

25 Aug '11

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: Dear user serusers(a)lists.iptel.org, mail server administrator of lists.iptel.org would like to let you know the following, We have detected that your e-mail account has been used to send a huge amount of junk email messages during this week. Probably, your computer was infected by a recent virus and now runs a hidden proxy server. [...] Content analysis details: (14.0 points, 5.0 required) pts rule name description ---- ---------------------- -------------------------------------------------- 0.5 FSL_XM_419 Old OE version in X-Mailer only seen in 419 spam 0.4 RCVD_IN_XBL RBL: Received via a relay in Spamhaus XBL [87.235.117.21 listed in zen.spamhaus.org] 0.0 DEAR_EMAIL BODY: Message contains Dear email address 0.8 BAYES_50 BODY: Bayes spam probability is 40 to 60% [score: 0.5000] 1.4 RCVD_IN_BRBL_LASTEXT RBL: RCVD_IN_BRBL_LASTEXT [87.235.117.21 listed in bb.barracudacentral.org] 0.3 FSL_UA FSL_UA 0.8 RDNS_NONE Delivered to internal network by a host with no rDNS 2.6 MSOE_MID_WRONG_CASE MSOE_MID_WRONG_CASE 1.2 TO_NO_BRKTS_MSFT To: misformatted and supposed Microsoft tool 1.9 FORGED_MUA_OUTLOOK Forged mail pretending to be from MS Outlook 2.5 DOS_OE_TO_MX Delivered direct to MX with OE headers 1.5 TO_NO_BRKTS_DIRECT To: misformatted and direct-to-MX 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.

1 0

Dumping Online Users stops at 2000 records
by Sebastian Damm 25 Aug '11

25 Aug '11

Hi, we have a script running on our Kamailio servers which dumps all online users with kamctl ul show, parses it and then sends it to database. (For explanation, we started doing it this way because we noticed at least with an older (open)ser release that SER would completely stop working for a couple of seconds while writing registered users to database every 5 minutes. When dumping it, we don't see any disruption.) With Kamailio 1.5 this works, with Kamailio 3.1 it works in the beginning, but when you pass the 2000 online user mark, it just stops working. I've started the Kamailio completely empty, then let users register against it. I did a "kamctl ul show" every 5 seconds, this is what the results were: Domain:: location table=512 records=82 max_slot=2 Domain:: location table=512 records=412 max_slot=4 Domain:: location table=512 records=640 max_slot=5 Domain:: location table=512 records=973 max_slot=7 Domain:: location table=512 records=1209 max_slot=8 Domain:: location table=512 records=1454 max_slot=11 Domain:: location table=512 records=1667 max_slot=12 Domain:: location table=512 records=1907 max_slot=12 500 command 'ul_dump' failed 500 command 'ul_dump' failed 500 command 'ul_dump' failed 500 command 'ul_dump' failed 500 command 'ul_dump' failed 500 command 'ul_dump' failed Is there a hard limit for dumping users through the fifo? Why? Or should we do it differently? Best Regards, Sebastian -- Dipl.-Inf. Sebastian Damm - VoIP-Engineer - damm(a)sipgate.de sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois Steuernummer: 106/5724/7147, Umsatzsteuer-ID: DE219349391 www.sipgate.de - www.sipgate.at - www.sipgate.co.uk

4 5

Segmentation Fault in ACC module
by Bruno Bresciani 25 Aug '11

25 Aug '11

Hi all, ACC module is generating segmentation fault at acc_db_request function in kamailio 3.1.2 with database postgres. When module try to access the struct db_func_t acc_dbf.use_table (line 389) the child proccess that execute this operation is exited by a signal 11. Someone know why this segmentation fault is gererating? How I fix this problem? Best Regards

2 1

Kamailio 3.1.1 snmpstat help!!
by Stephen Dodge (Bistech) 25 Aug '11

25 Aug '11

Hello, I am having difficulty running the snmpstats module. We are running CentOS 5.6 and Kamailio version 3.1.1 (i386/linux). I've tried a fresh install of Kamailio, installing net-snmp from source but I can't seem to get it running - I've seen a couple of posts regarding the issue but nothing seems to work for us!! The issue I get when I start Kamailio is: 0(27523) ERROR: <core> [sr_module.c:523]: ERROR: load_module: could not open module </usr/local/lib/kamailio/modules_k/snmpstats.so>: /usr/lib/libnetsnmpagent.so.10: undefined symbol: hosts_ctl Any help would be much appreciated. Also, if anyone has any suggestions on how best to get either Solarwinds or PRTG to monitor our Kamailio servers that would be good. Thanks. Stephen Dodge Visit www.bistech.co.uk<http://www.bistech.co.uk/> for news and information ________________________________ Information in this message, including any attachments, is confidential to the person to whom it is addressed and may be legally privileged. If you are not the intended recipient please notify the sender and delete the message from your system. Please note that Bistech Group plc, Bistech plc, Bisnet Limited and the sender do not accept any responsibility for viruses. It is your responsibility to check the e-mail and any attachments for viruses. Calls may be monitored and recorded.

1 0

ACC with MySqsl problem after installing Siremis
by אורי שקד - חטיבת הנדסה ורשת - Uri Shacked 25 Aug '11

25 Aug '11

Hi, I managed to setup the ACC module with MySql. Now, after installing Siremis, the ACC table and the Missed_calls table are empty and do not populae the data from the call I make. Any ideas why? BR, Uri

4 8

Re: [SR-Users] ACC with MySqsl problem after installing Siremis
by Daniel-Constantin Mierla 25 Aug '11

25 Aug '11

On 8/25/11 1:29 PM, אורי שקד - חטיבת הנדסה ורשת - Uri Shacked wrote: > > I managed… the softphones clients were configured with “multicast” so > they where “talking behind my back” J…. I should have noticed it earlier… > > Thanks, and sorry…. > Don't worry, it happens many time to all of us to overlook the simplest things. Whenever the solution is found it is good to keep the mailing list cc-ed, so people that will read in the future the archive for similar issues, can get the resolution to this discussion. Cheers, Daniel > ------------------------------------------------------------------------ > > *From:*Daniel-Constantin Mierla [mailto:miconda@gmail.com] > *Sent:* Thursday, August 25, 2011 2:23 PM > *To:* SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List > *Cc:* אורי שקד- חטיבת הנדסה ורשת- Uri Shacked > *Subject:* Re: [SR-Users] ACC with MySqsl problem after installing Siremis > > Hello, > > do you get any error message in syslog from acc module. Depending on > your Linux OS, the syslog file should be /var/log/syslog or > /var/log/messages. > > Since you have acc to syslog enabled as well, do you see the ACC > records for syslog written? > > Cheers, > Daniel > > On 8/22/11 2:50 PM, אורי שקד- חטיבת הנדסה ורשת- Uri Shacked wrote: > > Hi, > > I checked and redo the changes… > > It doesn’t work. > > My config file is attached. > > I checked the DB tables and columns to see it is OK and it is….. > > Any Idea what else? > > Thanks, > > Uri > > ------------------------------------------------------------------------ > > *From:*Elena-Ramona Modroiu [mailto:ramona@asipto.com] > *Sent:* Monday, August 22, 2011 11:57 AM > *To:* SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List > *Cc:* אורי שקד- חטיבת הנדסה ורשת- Uri Shacked > *Subject:* Re: [SR-Users] ACC with MySqsl problem after installing Siremis > > Hi, > > if you installed Siremis 2.0 via the wizard, the structures of acc and > missed_calls tables were updated, so you have to change the parameters > of acc module in kamailio.cfg, like: > > http://kb.asipto.com/siremis:install20:accounting#config_file > > If you want to see the sql used to create the new acc and missed_calls > tables, see: > > http://siremis.git.sourceforge.net/git/gitweb.cgi?p=siremis/siremis;a=blob;… > > Regards, > Ramona > > > On 8/21/11 10:58 AM, אורי שקד- חטיבת הנדסה ורשת- Uri Shacked wrote: > > Hi, > > I managed to setup the ACC module with MySql. > > Now, after installing Siremis, the ACC table and the Missed_calls > table are empty and do not populae the data from the call I make. > > Any ideas why? > > BR, > > Uri > > > <http://robustus.impactia.com/AnalyticServer/redirect?cid=50f7cf7d3be2418b&m…><http://robustus.impactia.com/AnalyticServer/redirect?cid=50f7cf7d3be2418b&m…><http://robustus.impactia.com/AnalyticServer/redirect?cid=50f7cf7d3be2418b&m…><http://robustus.impactia.com/AnalyticServer/redirect?cid=50f7cf7d3be2418b&m…> > > ------------------------------------------------------------------------ > > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > > > -- > Daniel-Constantin Mierla --http://www.asipto.com > Kamailio Advanced Training, Oct 10-13,Berlin:http://asipto.com/u/kat > http://linkedin.com/in/miconda -- http://twitter.com/miconda -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

Float Comparison
by Brandon Armstead 25 Aug '11

25 Aug '11

Hello, Is there anything special that needs to be done for float comparison? For example: if([5.5 >= 4.3]) .... or if(5.5 > 4.3) .... The conditional does not seem to be coming back as true like it should? This is simply an example... the precision in the actual case is different (longer decimal places). Thank you for all and any help in advance!

4 9

[regression] Message flags set in branch_route not saved into transaction
by Alex Hermann 25 Aug '11

25 Aug '11

Hello, It seems kamailio 3 does not save message flags set in branch route into the transaction. In reply_route and failure_route the flag set in branch_route is unset. In 1.4 this used to work. I would like to get that behaviour back, is this possible? Testscenario: route { setflag(0); t_on_branch("1"); t_on_reply("1"); t_on_failure("1"); xlog("L_NOTICE", "[$rm] Request before relay: $mF"); t_relay(); } branch_route[1] { xlog("L_NOTICE", "[$rm] Branch begin: $mF"); setflag(8); xlog("L_NOTICE", "[$rm] Branch end: $mF"); } onreply_route[1] { xlog("L_NOTICE", "[$rm] Reply ($rs) begin: $mF"); setflag(4); xlog("L_NOTICE", "[$rm] Reply end: $mF"); } failure_route[1] { xlog("L_NOTICE", "[$rm] Failure ($T_reply_code): $mF"); } Log output on 3.x: [INVITE] Request begin: 00000000 [INVITE] Request before relay: 00000001 [INVITE] Branch begin: 00000001 [INVITE] Branch end: 00000101 [INVITE] Reply (100) begin: 00000001 [INVITE] Reply end: 00000011 [INVITE] Failure (408): 00000011 Log output on 1.4: [INVITE] Request begin: 00000000 [INVITE] Request before relay: 00000001 [INVITE] Branch begin: 00000001 [INVITE] Branch end: 00000101 [INVITE] Reply (100) begin: 00000101 [INVITE] Reply end: 00000111 [INVITE] Failure (408) begin: 00000111 -- Greetings, Alex Hermann

2 5

Kamailio Freezes and no calls are processed
by Omar 25 Aug '11

25 Aug '11

The only difference is we have added some AVPs variables to process, and kamailio stops processing new calls, is not regular, but seems is related to the number of calls received. No additional calls can be processed after certain time, or maybe some amount of calls. we were unable to isolate the problem, but kamailio.fifo is removed, which never happened before. No core dump created. did anybody have a similar issue. Omar

4 5

add carrierroute module
by אורי שקד - חטיבת הנדסה ורשת - Uri Shacked 25 Aug '11

25 Aug '11

Hi, I am trying to add the carrierroute module. (I installed Kamailio 3.1.4 with mysql). I add to the cfg file the ‘loadmodule “carrierroute.so”’ and all the modparams needed. When I started Kamailio I encountered errors. One of them was that the carrierroute.so could not be found…. So, can I add this module without the need to reinstall everything again? Please advice, Thanks, Uri

3 2

SIP to SMS/HTTP
by Mark Anthony C. Delfin 24 Aug '11

24 Aug '11

Hello list, I am not sure if this has been ask before but is there a way that Kamailio can sip received SIP method MESSAGES and then call a URL (Kannel URL). Thank you. Regards, Mark Anthony C. Delfin

3 2

WARNING: timer: add_timeout: 0 expire timer added
by Alex Hermann 23 Aug '11

23 Aug '11

Hello, I get this warning message when a request is relayed over tcp: WARNING: <core> [timer_funcs.h:119]: WARNING: timer: add_timeout: 0 expire timer added The warning is displayed after send_route() has run. (relevant) settings: modparam("tm", "fr_timer", 500) modparam("tm", "fr_inv_timer", 80000) modparam("tm", "wt_timer", 20000) Every other timer related setting is at its default. Just before t_relay() is called, fr timer is set with: t_set_fr($avp(ringtime) * 1000); where $avp(ringtime) = 80. An identical setup with udp does not show the warning. Where does this warning come from and how do i suppress it? -- Greetings, Alex Hermann

2 4

uac_req: pseudovariables, avps
by Jon Bonilla 23 Aug '11

23 Aug '11

Hi all I'm trying to generate a request using uac_req in a failure route when the call has been canceled. (Kamailio 3.1.3) Here's the code: failure_route[FAILURE_ROUTE_LALA] { if t_is_canceled() { $uac_req(method) = "INVITE"; $uac_req(ouri) = "sip:127.0.0.1:5090"; $uac_req(ruri) = $ru; $uac_req(furi) = $fu; $uac_req(turi) = $tu; $uac_req(hdrs) = "X-PUSH-Type: mytype\r\nX-PUSH-CLI: $avp(s:first_caller_cli)\r\nX-PUSH-DST: $rU\r\n"; uac_req_send(); route(ROUTE_STOP_RTPPROXY); exit; } ... } The problem here is that the headers X-PUSH-CLI and X-PUSH-DST have literal '$avp(s:first_caller_cli)' and '$rU' values. Here's the INVITE I send: U 2011/08/23 17:07:09.467666 127.0.0.1:5062 -> 127.0.0.1:5090 INVITE sip:ngcptest1@domain.com SIP/2.0' Via: SIP/2.0/UDP 127.0.0.1:5062;branch=z9hG4bK925.85ac80a2000000000000000000000000.0' To: sip:ngcptest1@domain.com' From: sip:sipwise1@domain.com;tag=cf3b49e60a035342e3af7df009437068-9f9a' CSeq: 10 INVITE' Call-ID: 25f2b3a34ed8488c-3417(a)127.0.0.1' Content-Length: 0' User-Agent: Sipwise NGCP Proxy 2.X' X-PUSH-Type: mytype' X-PUSH-CLI: $avp(s:first_caller_cli)' X-PUSH-DST: $rU' I also tried to put the headers in a $var and in a $avp and set the value like: $var(cancel_push_request)="X-PUSH-Type: missed_call\r\nX-PUSH-CLI: $avp(s:first_caller_cli)\r\nX-PUSH-DST: $rU\r\n"; $uac_req(hdrs) = $var(cancel_push_request); If I xlog the values like this: xlog("L_INFO", "-------- Comprobando valores '$avp(s:first_caller_cli)' ------ '$rU'); The kamailio log show the correct value: INFO: <script>: -------- Comprobando valores '4312345' ------ 'ngcptest1' How could I use my stored values in the INVITE generation? cheers, Jon

2 2

max branches
by Jon Bonilla 23 Aug '11

23 Aug '11

Hi all Is there any configuration option in kamailio 3.1 to set the max number of branches? I have a system that needs to run over 30 times to failure_route and call append_brach() from there. At 11-12 iteractions I get the following errors: ERROR: tm [t_fwd.c:651]: ERROR: add_uac: maximum number of branches exceeded ERROR: tm [t_fwd.c:1528]: ERROR: t_forward_nonack: failure to add branches ERROR: tm [tm.c:1368]: ERROR: w_t_relay_to: t_relay_to failed ERROR: tm [t_reply.c:965]: ERROR: run_failure_handlers: Error in run_top_route cheers, Jon

4 4

Testing upcoming version 3.2.0
by Daniel-Constantin Mierla 23 Aug '11

23 Aug '11

Hello, for those looking to test v3.2.0 and need some instructions, for Kamailio flavour is a wiki page showing step-by-step installation: * http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-devel-from-git Practically 3.2.0 is the current devel branch. Testing SER flavour the difference is just using FLAVOUR=ser when compiling the application. Developers as well as users are encouraged to add details about the new additions in v3.2.0 to wiki page: * http://sip-router.org/wiki/features/new-in-devel Also, please add instructions for migrating from 3.1.x to 3.2.0 to the wiki page: * http://sip-router.org/wiki/install/3.1.x-to-3.2.x Thanks, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

Development branch frozen for v3.2.0
by Daniel-Constantin Mierla 23 Aug '11

23 Aug '11

Hello, since several hours ago, the development of new features in master branch has been frozen, to allow proper testing for release of version 3.2.0. That means no new features should be committed to GIT master branch. Just before the release (expected in 1.5months) we will create a dedicate branch for 3.2.x series and master branch will be again open for new stuff. Meanwhile, developers can use personal branches to push code in the public space. GIT master branch will be used in this period for: - bug fixing - hammering of the new features in 3.2.0 - documentation improvements - merging of duplicated modules There are two good news for 3.2.0: - the number of the new features is very impressive. Besides the old squad, we gained many new developers lately, coming with interesting contributions, therefore I am sure you'll have lot of fun testing them - the core components (transport layers, asynchronous processing, timers, locking & memory managers, a.s.o.) were barely touched since 3.0, so this is the 3rd release in a row using the new scalable core framework. That shows the maturity and there will be no much to focus on them If you need assistance while trying some new features, feel free to email to sr-dev mailing list. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

RFC-5766 / RFC-6062 compliant TURN clients?
by Rafal Boni 23 Aug '11

23 Aug '11

I know this might not be totally on-topic for the list, but I figured the list participants might have experience in this domain, so I'm lobbing it out there... I've got some situations where I'd like to connect users who are oftentimes sitting on very locked-down networks (ie, corporate networks) to some SIP-based VoIP infrastructure, but the idea of opening random UDP ports on the networks in question is a no-go. As a result, I'm experimenting with TURN over TCP or TLS as a solution to enable connectivity... however, I can't find any clients that implement the latest specs (X-Lite 4 appears to speak an older version of the TURN protocol; QJsip doesn't work for me; etc. etc.). Do folks here have any suggestions for TURN-compatible clients? I care mostly about PC for the time being (ie, just a proof of concept) but a PC/Mac/Linux solution would be great too. Thanks for any insights, --rafal

2 1

E2E ACK is not logged with siptrace modul kamailio
by MÉSZÁROS Mihály 22 Aug '11

22 Aug '11

Hi, I want to see all incoming and outgoing sip messages in sip trace table, but i am missing messages. I am using TLS so i can't see really trace the conversation only this way. I am wondering if it is know limitation, or an issue: I am experiencing that E2E ACK is not logged in siptrace table. I opened a ticket with the details: http://sip-router.org/tracker/index.php?do=details&task_id=144&project=1&pa… Many Thanks, Misi

3 11

New developer - Matthew Williams
by Daniel-Constantin Mierla 22 Aug '11

22 Aug '11

Hello, a new person has joined our development squad - Matthew Williams, currently working for Flowroute, USA. He is already a developer of SEMS (SIP Express Media Server) and he going to add shortly, just in time for 3.2, some Json related extensions, including a JSON-RPC client module. His GIT commit user id is: mgw. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

Re: [SR-Users] decode_mime_type error
by Jijo 22 Aug '11

22 Aug '11

Hi Marius, The existing implementation is done considering the limitation of decode_mime_type but somehow missed the change in parse_content_type_hdr. I agree with you that its better to do the change in decode_mime_type as it is common for lot of other callers.. Regarding the example you pointed, According to my understanding from the code, I believe that existing implementation for parse_accept_body shall work as the mime types are predefined. So in the example you pointed for second invocation 'some value' is not a defined mime type as it not matching with predefined types. The decode_mime_type returns end pointer and parse_accept_body shall process only with the valid mime type text/plain. another example with multiple mime types.., Accept: text/html, multipart/mixed In this case parse_accept_body returns 2 mime types as both are predefined in the list. Thanks Jijo On Thu, Aug 18, 2011 at 5:58 PM, Bucur Marius <bucur_marius_ovidiu(a)yahoo.com > wrote: > Hi Jijo, > > In my opinion, decode_mime_type is broken, and parse_accept_body does not > work as expected. > For example, this is a valid accept header: > > accept: text/plain;param=",some value". > > but the parse_accept_body will return -1 because the first return value of > decode_mime_type is ",some value\"". The second invocation will think "some > value\"" is a mime type, which obviously isn't. > > I must say I never saw such an accept header, but the standard permits it. > Indeed, a quick fix would be to ignore the return code when comma is found. > This must be done for other functions that call decode_mime_type > like has_body(mime) in textops. > > This is just my opinion, but I think this solution is a bit hackish since > the convention, the "promise" that decode_mime_type should respect is to > return a non-NULL, non-end pointer ONLY when there are more mime types in > the body. And it doesn't. > A lot of functions that call it rely on this. I think it is easier/more > elegant to fix the bug in the decode_mime_type then to change the function > convention in all callers. > Also, this might work for most callers, but parse_accept_body will still be > buggy, as I explained in the example above. > > Cheers, > Marius > > ________________________________ > From: Jijo <realjijo(a)gmail.com> > To: Bucur Marius <bucur_marius_ovidiu(a)yahoo.com> > Sent: Thursday, August 18, 2011 10:48 AM > Subject: Re: [SR-Users] decode_mime_type error > > > Hi Marius, > > Right, the same function is used for Accept and Content-Type. > > I don't think there is any change required for parsing Accept. In case of > Accept the function decode_mime_type is called in a loop until it find all > the media types. On the returning from decode_mime_type(), the main function > parse_accept_xxx() checks the pointer is 'comma' or not. if its comma, then > the function decode_mime_type called again and find the remaining media > types. So i think the existing implementation for Parsing Accept is fine. > > The only change required here is not return error for comma while parsing > the Content-Type. > > Thanks > Jijo > > > On Thu, Aug 18, 2011 at 12:32 PM, Bucur Marius < > bucur_marius_ovidiu(a)yahoo.com> wrote: > > Hello Jijo, > > > >You are right, multiple mime-types are not allowed. But it seems like > decode_mime_type had the purpose of also parsing the Accept header: > > > >Accept: text/html, image/jpeg, multipart > > > >hence the search for commas :). > >The only function that uses this feature and calls decode_mime_type in a > loop is parse_accept_body. The other functions that call decode_mime_type > just fail when returning ret != end, and since multiple mime types in > content-type are not allowed the behavior is fine. > >The bad thing is the Accept header can have mime parameters too (which can > contain comma). > > > >Accept = "Accept" HCOLON [ accept-range *(COMMA accept-range) > ] > >accept-range = media-range *(SEMI accept-param) > >media-range = ( "*/*" / ( m-type SLASH "*" ) / ( m-type SLASH > m-subtype ) ) *( SEMI m-parameter ) > > > >So I think the best thing would be to change decode_mime_type so that it > does a more thorough parsing (e.g. check for parameter) hence a parameter > value can contain commas (only if quoted). > > > > > >Cheers, > > > >Marius > >________________________________ > >From: Jijo <realjijo(a)gmail.com> > >To: Bucur Marius <bucur_marius_ovidiu(a)yahoo.com>; SIP Router - Kamailio > (OpenSER) and SIP Express Router (SER) - Users Mailing List < > sr-users(a)lists.sip-router.org> > >Sent: Thursday, August 18, 2011 8:04 AM > >Subject: Re: [SR-Users] decode_mime_type error > > > > > > > >Hi Marius, > > > >Thanks for the response.. I checked the other functions, but they don't > have the check for ret !=end, but they check the pointer and if it is comma > then loop through again until it find all the media types. > > > >As per the RFC3261 multiple media-types are not supoorted in the > Content-Type. So I'm not sure why the author used comma to determine > multiple media types . Probably just followed like Route or Record-Route > headers..?? > > > >Content-Type = ( "Content-Type" / "c" ) HCOLON media-type > >media-type = m-type SLASH m-subtype *(SEMI m-parameter) > >m-type = discrete-type / composite-type > >discrete-type = "text" / "image" / "audio" / "video" > >/ "application" / extension-token > >composite-type = "message" / "multipart" / extension-token > > > >Record-Route = "Record-Route" HCOLON rec-route *(COMMA rec-route) > > > >Thanks > >Jijo > > > > > > > >On Thu, Aug 18, 2011 at 2:23 AM, Bucur Marius < > bucur_marius_ovidiu(a)yahoo.com> wrote: > > > >Hello Jijo, > >> > >>It seems like the decode_mime_type is a somehow broken. The comma is very > well allowed in boundary, as you said. The BNF specified in RFC2046 permits > it. > >>But, the decode_mime_type function ignores everything coming after comma. > More than that, it notifies the function caller that this content type has > multiple mime types. > >>I think the author of the function thought of something like: > >> > >>Content-Type: text/html, image/jpeg // very weird, though > >> > >>This is wrong, hence the comma can be in a parameter (like in your case). > >>I think the best fix, as you said is to remove that check, hence a > non-NULL return value doesn't mean anything (that the content type has > multiple mime types). > >>Another fix is to write a "good" decode_mime_type, that checks if the > comma is inside a parameter. I don't know if effort is worth it. > >> > >>One thing we need to do is to check if there are any functions in > Kamailio that call decode_mime_type and also perform this check (ret != > end). > >> > >>Cheers, > >>Marius > >> > >> > >>________________________________ > >>From: Jijo <realjijo(a)gmail.com> > >>To: sr-users(a)lists.sip-router.org; sr-dev(a)lists.sip-router.org > >>Sent: Wednesday, August 17, 2011 10:54 AM > >>Subject: [SR-Users] decode_mime_type error > >> > >> > >> > >>Hi All, > >> > >>The function parse_content_type_hdr() is failing in decode_mime_type() > when Content-Type parameter "boundary" has value comma as below. The error > message is "ERROR:parse_content_type_hdr: CONTENT_TYPE hdr contains " > >> "more then one mime type :-(! > >> > >>Content Type Header is as below, It works fine if the boundary value is > without comma. > >> > >>Content-Type: multipart/mixed;boundary=",AW" > >> > >>The parse_content_type_hdr() is failing in the following code, > >> > >> ret = decode_mime_type(msg->content_type->body.s, end , &mime); > >> if (ret==0) > >> goto error; > >> if (ret!=end) { > >> > >> LOG(L_INFO,"ERROR:parse_content_type_hdr: CONTENT_TYPE hdr > contains " > >> "more then one mime type :-(!\n"); > >> goto error ; > >> } > >> > >>I thought of fixing this issue by commenting the code for condition ret > !=end. I'm not sure why we we need that check, as mime variable has the > information to process the content. > >> > >>Please let me know if you see any impact. > >> > >>Thanks > >>Jijo > >> > >> > >> > >> > >> > >>_______________________________________________ > >>SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > >>sr-users(a)lists.sip-router.org > >>http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > >> > >>_______________________________________________ > >>SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > >>sr-users(a)lists.sip-router.org > >>http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > >> > > >

2 2

min expires question
by Fabian Borot 22 Aug '11

22 Aug '11

Hello This is how my registrar module is configured, hoping that min-expires set to 30 wont allow registration attempts more often than 30 seconds, but if I configure a UAC to register every 15 seconds I see that kamailio allows it. Shouldn't it reply with "422 Session Interval Too Small" or "423 Interval Too Brief"? # ----- registrar params ----- modparam("registrar", "default_expires", 120) modparam("registrar", "min_expires", 30) modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) 131.380336 192.168.1.6 -> 192.168.1.194 SIP Request: REGISTER sip:192.168.1.194 131.381174 192.168.1.194 -> 192.168.1.6 SIP Status: 200 OK (1 bindings) 144.386014 192.168.1.6 -> 192.168.1.194 SIP Request: REGISTER sip:192.168.1.194 144.387110 192.168.1.194 -> 192.168.1.6 SIP Status: 200 OK (1 bindings) 157.395358 192.168.1.6 -> 192.168.1.194 SIP Request: REGISTER sip:192.168.1.194 157.396340 192.168.1.194 -> 192.168.1.6 SIP Status: 200 OK (1 bindings) 170.404905 192.168.1.6 -> 192.168.1.194 SIP Request: REGISTER sip:192.168.1.194 170.405771 192.168.1.194 -> 192.168.1.6 SIP Status: 200 OK (1 bindings) 183.414375 192.168.1.6 -> 192.168.1.194 SIP Request: REGISTER sip:192.168.1.194 183.415217 192.168.1.194 -> 192.168.1.6 SIP Status: 200 OK (1 bindings) txs fborot

3 4

Sip Router Issues (Topoh's module)
by Richard DEMONGEOT 22 Aug '11

22 Aug '11

Dear all, I've encountered two issues with Sip Router. Build : Kamailio 3.1.4 for Squeeze (Kamailio's debian Packages). When i'm using Topoh with Call-ID encoding, instruction dlg_bye("ALL") generate me an error on logfiles : Aug 4 17:46:26 kamtest /usr/sbin/kamailio[25267]: ERROR: topoh [th_mask.c:166]: invalid input string"dd04db6ccc7074deb9f8a1160946d331@0:0:0:0:0:0:0:0" Aug 4 17:46:26 kamtest /usr/sbin/kamailio[25267]: ERROR: topoh [th_msg.c:480]: cannot decode callid Aug 4 17:46:26 kamtest /usr/sbin/kamailio[25268]: ERROR: topoh [th_mask.c:166]: invalid input string"dd04db6ccc7074deb9f8a1160946d331@0:0:0:0:0:0:0:0" Aug 4 17:46:26 kamtest /usr/sbin/kamailio[25268]: ERROR: topoh [th_msg.c:480]: cannot decode callid Only "INCOMING LEG" is ended, and "OUTGOING LEG", with call-ID encoded continue. Have you any idea about this problem? Can we use dlg_bye("ALL") with call-id encoding? Best regards,

2 5

10 Years SER Conference, Berlin - full booked
by Daniel-Constantin Mierla 22 Aug '11

22 Aug '11

Hello, the 10 years SER conference in Berlin is full booked, meaning that new registrations will be accepted only if someone that registered previously will cancel his/her participation. Since we just discovered such a situation, in the case you did register and you haven't received a confirmation email (not the one automatically generated at registration time, but a second one), please reply to me or write to registration(a)lists.sip-router.org telling the date when you did the registration and we will try to solve it. Looking forward to meeting many of you soon in Berlin! Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

Freezing development for v3.2.0 today
by Daniel-Constantin Mierla 22 Aug '11

22 Aug '11

Hello, just a short reminder that we will close the development of new features for version 3.2.0 at the end of today, GMT. Then the focus should be on testing (e.g., bug fixing) and enhancements to documentation. If you have something to commit, do it today, or if it is impossible for you to make it in the GIT repo today, let us know and we will see what can be done. But don't forget, 3.3.0 will come as well in the no-so-far future. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

masking carriers
by Anders 20 Aug '11

20 Aug '11

Hi, Background: Kamailio 1.5.x + rtpproxy I am not too crazy about telling our customers where we send our calls. But if they take a look at the sip dialog as it is right now, they can see our suppliers/carriers IP addresses. What can we do to mask this, so that the last piece of information stops at our IPs? We are running rtpproxy for the media - if that makes any difference. Someone told me I cannot mask anything without moving to Kamailio 3.x - is that the case? Thanks!! //Anders

2 2

Question about encryption with Kamailio
by margot.basa 19 Aug '11

19 Aug '11

Hello all, I would like to use Kamailio to encrypt contents of SIP messages (using SIP TLS) between 2 endpoints, i.e.: - To use 5061 port instead of 5060 port, - To use sips uri instead of sip uri... For example, T1 and T2 communicates with "Server A" like that: 1) T1 and T2 send REGISTER to "Server A" 2) T1 and T2 received 200 OK from "Server A" ... 3) "Server A" sends an INVITE message to T1 and T2 ... 4) RTP flow between T1 and T2 (this should not be encrypted) ... 5) "Server A" sends a BYE request to T1 and T2 ... All those exchanges are made on Transport layer TCP or UDP on port 5060. T1 and T2 are not able to support TLS but "Server A" needs to receive/send messages in SIP TLS. I would like to insert Kamailio between T1 and "Server A", T2 and "Server B" in order to encrypt contents of SIP messages. I have some questions about that: - I think Kamailio can do that but I am not sure, can you confirm that to me please? - Can I use Kamailio as it is to do that? - Do I have to add a "Route" header in requests in order that requests between T1 and "Server A" go through Kamailio or - Does Kamailio is able to intercept SIP packets automatically (with a certain configuration)? - Do you know difference between Freeswitch and Kamailio? (because I have seen that Freeswitch can do what I need: see Figure4: http://wiki.freeswitch.org/wiki/SIP_TLS) Thank you very much for your input. Regards Une messagerie gratuite, garantie à vie et des services en plus, ça vous tente ? Je crée ma boîte mail www.laposte.net

2 2

New developer: Alexandr Dubovikov
by Daniel-Constantin Mierla 19 Aug '11

19 Aug '11

Hello, a new person has now developer access to GIT repository - Alexandr Dubovikov, contributor in the past with patches, is going to work now on extensions related to Homer project - turning the SIP server in a SIP capturing server. His developer id is adubovikov. You can read more about Homer project at: * http://code.google.com/p/homer/ * http://www.kamailio.org/events/2011-Cluecon/AD-Homerv2.pdf Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

3 2

new functionality: CDR-based accounting
by Timo Reimann 19 Aug '11

19 Aug '11

Hi all, as announced quite a while ago, I finally checked in code that allows to produce CDRs (Call Data Records) directly from SIP-Router and generate logs accordingly. The main code portion resides in modules_k/acc and provides a switch to enable basic CDR generation including start time, end time, and duration. Analogous to the existing logging approach, you may define an extra parameter covering to-be-included dialog pseudo-variables that must be assigned in the configuration script. The new code will take care of transforming the basic and custom CDR fields into CDR logs at the end of a dialog. Speaking of dialogs: The implementation relies heavily on the dialog module. It takes advantage of dialog variables introduces by Carsten Bock and adds a few more features. Most notably, we had to change the dialog callback signature to provide both request and response messages. Having only one of them proved to be insufficient in certain cases; for instance, a locally generated 408 returned a FAKED_REPLY, thus rendering it impossible to access dialog variables through the PV framework. Other modules using dialog callbacks have been updated along the commit, third-party modules outside the repository will need to do so too (and think about whether using the request or response is the Right Thing to do). Due to the changes brought to the dialog module, I pushed the new acc and dialog code into a separate branch called treimann/acc-cdr. Feel free to give it a try by consulting the updated documentation and suggesting (or, if it's good enough, implementing :) ) improvements. A Kamailio 1.5 backport of the code has been in usage for quite some time with us, so generally there shouldn't be any major logical flaws. SIP-Router certainly needs more testing, however, so I'd be glad for any feedback. My plan is to merge the code into master branch prior to the 3.2 feature freeze, unless significant objections arise. Finally, big-time credits go to my co-worker Sven Knoblich who is the main contributor of the code. He's been working on this stuff for the past few months and dreams in CDRs by now. Cheers, --Timo

5 7

LCR Modules Comparison
by Spencer Thomason 19 Aug '11

19 Aug '11

Hello all, Can anyone shed some light into the differences between the available LCR modules? We have migrated our config from OpenSIPS where we used the drouting module to Kamailio where we are using the lcr module. Our ruleset is not crazy huge, about 100k entries. Are there compelling reasons to choose one module over the other? Is there one that has the most active development behind it and is a "standard" for the platform or is there one that will be deprecated in the foreseeable future? Thanks, Spencer

4 3

Re: [SR-Users] decode_mime_type error
by Bucur Marius 19 Aug '11

19 Aug '11

Hi Jijo, In my opinion, decode_mime_type is broken, and parse_accept_body does not work as expected. For example, this is a valid accept header: accept: text/plain;param=",some value". but the parse_accept_body will return -1 because the first return value of decode_mime_type is ",some value\"". The second invocation will think "some value\"" is a mime type, which obviously isn't. I must say I never saw such an accept header, but the standard permits it. Indeed, a quick fix would be to ignore the return code when comma is found. This must be done for other functions that call decode_mime_type like has_body(mime) in textops. This is just my opinion, but I think this solution is a bit hackish since the convention, the "promise" that decode_mime_type should respect is to return a non-NULL, non-end pointer ONLY when there are more mime types in the body. And it doesn't. A lot of functions that call it rely on this. I think it is easier/more elegant to fix the bug in the decode_mime_type then to change the function convention in all callers. Also, this might work for most callers, but parse_accept_body will still be buggy, as I explained in the example above. Cheers, Marius ________________________________ From: Jijo <realjijo(a)gmail.com> To: Bucur Marius <bucur_marius_ovidiu(a)yahoo.com> Sent: Thursday, August 18, 2011 10:48 AM Subject: Re: [SR-Users] decode_mime_type error Hi Marius, Right, the same function is used for Accept and Content-Type. I don't think there is any change required for parsing Accept. In case of Accept the function decode_mime_type is called in a loop until it find all the media types. On the returning from decode_mime_type(), the main function parse_accept_xxx() checks the pointer is 'comma' or not. if its comma, then the function decode_mime_type called again and find the remaining media types. So i think the existing implementation for Parsing Accept is fine. The only change required here is not return error for comma while parsing the Content-Type. Thanks Jijo On Thu, Aug 18, 2011 at 12:32 PM, Bucur Marius <bucur_marius_ovidiu(a)yahoo.com> wrote: Hello Jijo, > >You are right, multiple mime-types are not allowed. But it seems like decode_mime_type had the purpose of also parsing the Accept header: > >Accept: text/html, image/jpeg, multipart > >hence the search for commas :). >The only function that uses this feature and calls decode_mime_type in a loop is parse_accept_body. The other functions that call decode_mime_type just fail when returning ret != end, and since multiple mime types in content-type are not allowed the behavior is fine. >The bad thing is the Accept header can have mime parameters too (which can contain comma). > >Accept = "Accept" HCOLON [ accept-range *(COMMA accept-range) ] >accept-range = media-range *(SEMI accept-param) >media-range = ( "*/*" / ( m-type SLASH "*" ) / ( m-type SLASH m-subtype ) ) *( SEMI m-parameter ) > >So I think the best thing would be to change decode_mime_type so that it does a more thorough parsing (e.g. check for parameter) hence a parameter value can contain commas (only if quoted). > > >Cheers, > >Marius >________________________________ >From: Jijo <realjijo(a)gmail.com> >To: Bucur Marius <bucur_marius_ovidiu(a)yahoo.com>; SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List <sr-users(a)lists.sip-router.org> >Sent: Thursday, August 18, 2011 8:04 AM >Subject: Re: [SR-Users] decode_mime_type error > > > >Hi Marius, > >Thanks for the response.. I checked the other functions, but they don't have the check for ret !=end, but they check the pointer and if it is comma then loop through again until it find all the media types. > >As per the RFC3261 multiple media-types are not supoorted in the Content-Type. So I'm not sure why the author used comma to determine multiple media types . Probably just followed like Route or Record-Route headers..?? > >Content-Type = ( "Content-Type" / "c" ) HCOLON media-type >media-type = m-type SLASH m-subtype *(SEMI m-parameter) >m-type = discrete-type / composite-type >discrete-type = "text" / "image" / "audio" / "video" >/ "application" / extension-token >composite-type = "message" / "multipart" / extension-token > >Record-Route = "Record-Route" HCOLON rec-route *(COMMA rec-route) > >Thanks >Jijo > > > >On Thu, Aug 18, 2011 at 2:23 AM, Bucur Marius <bucur_marius_ovidiu(a)yahoo.com> wrote: > >Hello Jijo, >> >>It seems like the decode_mime_type is a somehow broken. The comma is very well allowed in boundary, as you said. The BNF specified in RFC2046 permits it. >>But, the decode_mime_type function ignores everything coming after comma. More than that, it notifies the function caller that this content type has multiple mime types. >>I think the author of the function thought of something like: >> >>Content-Type: text/html, image/jpeg // very weird, though >> >>This is wrong, hence the comma can be in a parameter (like in your case). >>I think the best fix, as you said is to remove that check, hence a non-NULL return value doesn't mean anything (that the content type has multiple mime types). >>Another fix is to write a "good" decode_mime_type, that checks if the comma is inside a parameter. I don't know if effort is worth it. >> >>One thing we need to do is to check if there are any functions in Kamailio that call decode_mime_type and also perform this check (ret != end). >> >>Cheers, >>Marius >> >> >>________________________________ >>From: Jijo <realjijo(a)gmail.com> >>To: sr-users(a)lists.sip-router.org; sr-dev(a)lists.sip-router.org >>Sent: Wednesday, August 17, 2011 10:54 AM >>Subject: [SR-Users] decode_mime_type error >> >> >> >>Hi All, >> >>The function parse_content_type_hdr() is failing in decode_mime_type() when Content-Type parameter "boundary" has value comma as below. The error message is "ERROR:parse_content_type_hdr: CONTENT_TYPE hdr contains " >> "more then one mime type :-(! >> >>Content Type Header is as below, It works fine if the boundary value is without comma. >> >>Content-Type: multipart/mixed;boundary=",AW" >> >>The parse_content_type_hdr() is failing in the following code, >> >> ret = decode_mime_type(msg->content_type->body.s, end , &mime); >> if (ret==0) >> goto error; >> if (ret!=end) { >> >> LOG(L_INFO,"ERROR:parse_content_type_hdr: CONTENT_TYPE hdr contains " >> "more then one mime type :-(!\n"); >> goto error ; >> } >> >>I thought of fixing this issue by commenting the code for condition ret !=end. I'm not sure why we we need that check, as mime variable has the information to process the content. >> >>Please let me know if you see any impact. >> >>Thanks >>Jijo >> >> >> >> >> >>_______________________________________________ >>SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>sr-users(a)lists.sip-router.org >>http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >>_______________________________________________ >>SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>sr-users(a)lists.sip-router.org >>http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >

2 1

The not so simple Kamailio Load Balancing of Asterisk
by Nick Khamis 19 Aug '11

19 Aug '11

Hello Everyone, I saw the simple load balancing of asterisk using x-SER documentation here "http://www.kamailio.org/dokuwiki/doku.php/asterisk:load-balancing-and-ha". Are there any up-tp-date not so simple documentation for setting up HA and LB asterisk instances using x-SER. Thanks in Advance, Nick Khamis Toronto Hydro Telecom

5 7

decode_mime_type error
by Jijo 18 Aug '11

18 Aug '11

Hi All, The function parse_content_type_hdr() is failing in decode_mime_type() when Content-Type parameter "boundary" has value comma as below. The error message is "ERROR:parse_content_type_hdr: CONTENT_TYPE hdr contains " "more then one mime type :-(! Content Type Header is as below, It works fine if the boundary value is without comma. Content-Type: multipart/mixed;boundary="*,*AW" The parse_content_type_hdr() is failing in the following code, ret = decode_mime_type(msg->content_type->body.s, end , &mime); if (ret==0) goto error; if (ret!=end) { LOG(L_INFO,"ERROR:parse_content_type_hdr: CONTENT_TYPE hdr contains " "more then one mime type :-(!\n"); goto error ; } I thought of fixing this issue by commenting the code for condition ret !=end. I'm not sure why we we need that check, as mime variable has the information to process the content. Please let me know if you see any impact. Thanks Jijo

2 2

Re: [SR-Users] mtree question
by Javier Gallart 18 Aug '11

18 Aug '11

Hi Juha based on my experience with the mtree module: -mt_match() will match the longest prefix in the tree, in your case 00358 -mt_ignore_duplicates makes sense when loading the data in the DB to the memory. There could be duplicate entries in the db and with that directive you tell kamailio how to handle that situation -I don't know about mt_tree_type...(never needed to test it) Hope it helps Regards Javi > ------------------------------ > > Message: 2 > Date: Fri, 12 Aug 2011 10:10:29 +0300 > From: Juha Heinanen <jh(a)tutpro.com> > Subject: [SR-Users] mtree question > To: sr-users(a)lists.sip-router.org > Message-ID: <20036.53733.20652.518606(a)sip.test.fi> > Content-Type: text/plain; charset=us-ascii > > after reading mtree readme, it is not clear to me if mt_match() matches > to longest or any matching prefix in the three. for example, if i have > tree with prefixes 00 and 00358 and string to be matched is > 0035892345670, will mt_match() set pv to value associated with 00 or > 00358? also, if mt_ignore_duplicates is set, will 00358 be considered > as duplicate of 00 or just prefixes that are exactly same. finally, > what are possible mt_tree_type values and what types they stand for? > > -- juha > > > > ------------------------------ > > _______________________________________________ >

3 5

lookup for specified user
by Efelin Novak 17 Aug '11

17 Aug '11

Hi folks, I discussed this issue in previous email ( [SR-Users Modifying $rU after lookup() ) but the question is a bit different now. I would like to change uri like lookup() function does it, but for the specific user. Problem is I want to have one registration (X) for several numbers. When this number is called I have to perform lookup() for this one common registration (X) and send this call there. To do this I changed $rU just before lookup() for X, and after lookup() I changed it back for the original number (both changes were done using variables $rU). However when there are two registrations (X1, X2), only X1 can be changed for the original number and the second one (X2) is sent to user without modification. I also found function reg_fetch_contacts(domain, uri, profile) but in documentation there is $ulc variable mentioned. This variable is not documented. Is there any way how to perform lookup for specific user? Thanks for the answer. Regards, Efelin

3 5

siptrace loosing an ACK
by Timo Klecker 17 Aug '11

17 Aug '11

Hello We are using the siptrace module to save the sip-messages to our database. All the messages are logged except one message. After INVITE, 100 trying and 200 OK we loose the outbound ACK which is forwarded by Kamailio statelessly in t_funcs.c. Is there any way to get a siptrace of this message, too? I've made a picture of the sip-messages and the lost ACK. Kind regards Timo Klecker

3 3

Error message
by Jon Bonilla 16 Aug '11

16 Aug '11

I have an error message in kamailio's log repeated many times. I tried to google it but no results (I think it's forst time I see no results in a google search) /usr/sbin/kamailio[3769]: ERROR: ctl [binrpc_run.c:942]: ERROR: binrpc: rpc_printf: binrpc_addstr failed: buffer too small (overflow) (-2) What does it mean? Any ideas? thanks, Jon

2 1

openSIPS vs Kamailio vs SIP-Router
by Nick Khamis 15 Aug '11

15 Aug '11

Hello Everyone, I can only imagine how many times this question has come up since post 2008. Please forgive my reoccurring of the issue. We are looking to provide carrier grade sip services to our clients world wide. What we need is a lightweight, robust and scalable solution that will allow us to terminate sip calls to our different carriers. Performance, and high throughput are factors very important to my employer. Features such as caller authentication, database back-end, load balancing, and interoperability with asterisk are things we are interested in, as was offered using OpenSER. With three+ open source proxy servers available on the net puts us in a situation where we have more solutions to choose from, at the same time wish the features from one were available in the other, and vice versa. With this in mind, we will have to fall back to other factors such as the most reliable, proven and active projects. As mentioned, we would choose functional stability over endless features that we will never use and that add to the projects fingerprint... I understand that all three projects are forks from OpenSER, people would naturally like to know what differentiates one from the other. Thanks in Advance, Nick Khamis Toronto Hydro Telecom

4 5

Kamailio presentations at Cluecon 2011
by Daniel-Constantin Mierla 15 Aug '11

15 Aug '11

Hello, I uploaded my slides presented at Cluecon 2011, last week in Chicago. In the first part the focus was on what has been happening in the last year around the project, on the second part showing some example of using it to secure VoIP networks. The PDF is available at: * http://asipto.com/u/42 Alexandr Dubovikov presented the recently announced Homer project which turns Kamailio in a scalable capturing system for SIP traffic, the slides for this talk are available at: * http://www.kamailio.org/events/2011-Cluecon/AD-Homerv2.pdf Stefan Wintermeyer presented Gemeinschaft 4 project, which is targeting to offer a very secure out-of-the-box IP PBX system, using Kamailio as one of the major components. The slides will be available in the near future (I will update the lists) -- he will present the same topic at 10 years SER conference in Berlin: * http://sip-router.org/10-years-ser/ Cheers, Daniel

3 2

10 Years SER Conference - update
by Daniel-Constantin Mierla 15 Aug '11

15 Aug '11

Hello, just wanted to let you know that only few seats are still available for the 10 Years SER Conference, on Sep 2, 2011, in Berlin, Germany -- more details at: * http://sip-router.org/10-years-ser/ If you plan to attend, do not wait any longer and register as soon as possible (form at: http://sip-router.org/registration-form/). The host FhG Fokus Institute has a medium size conference room, with no option to expand -- I expect the registration to be closed very soon. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

freezing for major release 3.2.0
by Daniel-Constantin Mierla 15 Aug '11

15 Aug '11

Hello, based on the last devel IRC meeting, the next major release should happen sometime by end of September, therefore we should freeze the development and go into testing phase about one month before. I propose Monday, August 22, the day to freeze development for version 3.2.0. By then, I hope every developer will be able to push to master branch what he/she wants to have in 3.2.0. Write here to mailing lists if you thing about alternatives. The very good side of 3.2.0 is that we didn't need to touch much the core and main modules, meaning we have a solid tested foundation, further testing should be focused on the new modules and the other enhancements. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 1

Wrong formatting of Request URI
by Mino Haluz 15 Aug '11

15 Aug '11

Hi, my kamailio is sometimes receiving ruri from remote clients which I assume are not right formatted. For ex.: sip:user@kamailio=3Buser=3Dphone Is there any mechanism which could transform this into something readable for kamailio or do I have to simply cut off everything after the host name of the RURI ? Mino

2 1

pua_xmpp questions
by laura testi 13 Aug '11

13 Aug '11

Hi all, we are trying to setup a kamailio presence server in SIP/SIMPLE domain to interwork with XMPP domains including the GTalk (see the config below). The architecture is like this: - SIP/SIMPLE server: kamailio3.14 with full configuration with XCAP integration - XMPP Gateway: another kamailio server + ejabberd server - component mode - xmpp moduleL: outbound proxy to the Kamailio SIP/SIMPLE server - pua_xmpp: no outbound proxy? - XMPP server: ejabberd The users from XMPP domain can add the SIP account and see the presence of the SIP users, but not vice versa. The chat from both directions works fine. in the Presence Server, we have configured the kamailio with XCAP integration. We have the following doubts: - does pua_xmpp/pua modules need the xcap integration for the presence integration with xmpp domains? because to use xcap for presence authorization rules, it needs the sip clients support the xcap - does pua_xmpp/pua support xcap? otherwise how to works? - in case of multiple SIP/SIMPLE presence server, how we can configure the server_address of pua_xmpp and presence parameter in xmpp gw? Can you help us to clarify the doubts please? Many thanks in advanced! Best Regards, Laura PS: following are the main configuration of the xmpp GW: --------------------------------------------------------------------------------------------- ... modparam("xmpp", "domain_separator", "*") modparam("xmpp", "backend", "component") modparam("xmpp", "gateway_domain", "<mygwdomain>") modparam("xmpp", "xmpp_domain", "<mygwdomain>") modparam("xmpp", "xmpp_host", "127.0.0.1") modparam("xmpp", "xmpp_password", "secret") modparam("xmpp", "outbound_proxy", "<my oubound proxy uri>") modparam("pua", "outbound_proxy", "<my outbound proxy uri>") modparam("pua", "update_period", 60) modparam("pua", "default_expires", 1200) modparam("presence", "server_address", "<my presence server uri>") modparam("pua_xmpp", "server_address", "<my presence server uri>") route{ route(REQINIT); t_check_trans(); if (uri =~ "sip:.+@.*<myxmppdomain>") { route(PRESENCE); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } route(CHAT); } xlog("L_INFO", "*** xmpp: unhandled message type\n"); t_reply("503", "Service unavailable"); return; } ... route[CHAT] { if(!is_method("MESSAGE")) return; if (!t_newtran()) { sl_reply_error(); exit; } xlog("L_INFO", "*** xmpp-handled MESSAGE message."); if ($cT=~"^text/plain") { if (xmpp_send_message()) { t_reply("200", "Accepted"); } else { t_reply("404", "Not found"); } } else { xlog("L_INFO", "*** xmpp-handled MESSAGE, ignoring not text messages"); t_reply("200", "Accepted"); } t_release(); exit; } route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE|NOTIFY")) return; # create new transaction to catch retransmissions if (!t_newtran()) { sl_reply_error(); exit; } if( is_method("NOTIFY")) { xlog("L_INFO", "*** xmpp-handled NOTIFY message."); if(pua_xmpp_notify()) t_reply("200", "OK"); t_release(); exit; } else if( is_method("SUBSCRIBE")) { xlog("L_INFO", "*** xmpp-handled SUBSCRIBE message.\n"); handle_subscribe(); if($(hdr(Event))== "presence") { pua_xmpp_req_winfo("$ruri", "$hdr(Expires)"); } t_release(); exit; } else if(is_method("PUBLISH")) { handle_publish(); t_release(); exit; } } ...

2 8

subscriber table reload
by Fabian Borot 12 Aug '11

12 Aug '11

Hello When one creates a user using the cli command like this: "kamctl add 200 abc", it is my understanding that the information goes to the subscriber table and into memory. Is that right? But if one populates the subscriber table manually [adding a record on the table w/o the /kamctlcli command], do we need to "load" those accounts into memory before a client tries to register or does kamailio looks it up when the registration request comes in at that time? What we are trying to do is to bulk load subscribers on the subscribers table instead of using the "kamctl add" command. For example, with LCR parameters we modify the tables manually and then do a "lcr_reload" and all the changes will be in memory after the reload. thank you in advance fborot

4 5

mtree question
by Juha Heinanen 12 Aug '11

12 Aug '11

after reading mtree readme, it is not clear to me if mt_match() matches to longest or any matching prefix in the three. for example, if i have tree with prefixes 00 and 00358 and string to be matched is 0035892345670, will mt_match() set pv to value associated with 00 or 00358? also, if mt_ignore_duplicates is set, will 00358 be considered as duplicate of 00 or just prefixes that are exactly same. finally, what are possible mt_tree_type values and what types they stand for? -- juha

1 0

kamailio - proxy/relay
by Sunny 11 Aug '11

11 Aug '11

Hi list, I'm trying to configure kamailio to work as a OBP (Outbound Proxy). Basically KAMAILIO receive requests from UAs, after that it acts like a client and send those requests to the IP-PBX. On the same way the responses are forwarded from IP-PBX to the UAs. [UA] --> [KAMAILIO] --> [IP-PBX ] [UA] <-- [KAMAILIO] <-- [IP-PBX ] Following possible call flow: [UA ] -> [KAMAILIO] SIP/SDP Request: INVITE sip:12345678909@[KAMAILIO], with session description [KAMAILIO] -> [UA ] SIP Status: 100 Trying [KAMAILIO] -> [IP-PBX ] SIP/SDP Request: INVITE sip:12345678909@[IP-PBX ], with session description [IP-PBX ] -> [KAMAILIO] SIP Status: 100 Trying [IP-PBX ] -> [KAMAILIO] SIP Status: 407 Proxy Authentication Required [KAMAILIO] -> [IP-PBX ] SIP Request: ACK sip:12345678909@[IP-PBX ] [KAMAILIO] -> [UA ] SIP Status: 407 Proxy Authentication Required [UA ] -> [KAMAILIO] SIP Request: ACK sip:12345678909@[KAMAILIO] [UA ] -> [KAMAILIO] SIP/SDP Request: INVITE sip:12345678909@[KAMAILIO], with session description [KAMAILIO] -> [UA ] SIP Status: 100 Trying [KAMAILIO] -> [IP-PBX ] SIP/SDP Request: INVITE sip:12345678909@[IP-PBX ], with session description [IP-PBX ] -> [KAMAILIO] SIP Status: 100 Trying [IP-PBX ] -> [KAMAILIO] SIP/SDP Status: 183 Session Progress, with session description [KAMAILIO] -> [UA ] SIP/SDP Status: 183 Session Progress, with session description [IP-PBX ] -> [KAMAILIO] SIP/SDP Status: 180 Ringing, with session description [KAMAILIO] -> [UA ] SIP/SDP Status: 180 Ringing, with session description (...) [UA ] -> [KAMAILIO] SIP Request: CANCEL sip:12345678909@[KAMAILIO] [KAMAILIO] -> [UA ] SIP Status: 200 OK [KAMAILIO] -> [IP-PBX ] SIP Request: CANCEL sip:12345678909@[IP-PBX ] [IP-PBX ] -> [KAMAILIO] SIP Status: 200 OK [IP-PBX ] -> [KAMAILIO] SIP Status: 487 Request Terminated [KAMAILIO] -> [IP-PBX ] SIP Request: ACK sip:12345678909@[IP-PBX ] [KAMAILIO] -> [UA ] SIP Status: 487 Request Terminated [UA ] -> [KAMAILIO] SIP Request: ACK sip:12345678909@[KAMAILIO] Wondering if anyone could help to make this configuration work? Thanks, Sunny

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