- sr-users - kamailio.org

tm.t_uac_wait asynchronous?
by Alex Balashov 24 Nov '10

24 Nov '10

Does tm.t_uac_wait RPC call block a worker process while waiting for a reply, or use some sort of asynchronous callback? Thanks, -- Alex Balashov - Principal Evariste Systems LLC 1170 Peachtree Street 12th Floor, Suite 1200 Atlanta, GA 30309 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

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Re: [SR-Users] Doubts with Registration
by Daniel-Constantin Mierla 24 Nov '10

24 Nov '10

never write private emails. They will be simply discarded. ngrep gets the traffic before the firewall. Do you have a firewall? start kamailio with log_stderror=yes and debug=5. if you don't see any sip message processing, then sip traffic does not reach the application layer. Cheers, Daniel On 11/18/10 10:39 AM, Danny Dias wrote: > Thanks for the answer Daniel, > > 2010/11/17 Daniel-Constantin Mierla<miconda(a)gmail.com> >> Hello, >> >> On 11/17/10 6:58 PM, Danny Dias wrote: >> >> Hello, >> I'm having problems with my kamailio, my phones are unable to register into the Proxy. I have been checking with ngrep-sip and all i'm receiving is many REGISTER from the phones but never getting some response from the Proxy >> The Proxy has a public IP and the phones are behind NAT. The weir thing is that my logs are not showing any output...and as you can see on the script, i have a section to receive REGISTER requests.... >> if (is_method("REGISTER")) { >> if (!save("location")) { >> sl_reply_error(); >> } >> exit; >> } >> >> If everything is fine, then no log message should be printed if debug parameter is low. > But i should see in the Kamailio logs the answers, right? or i have to > configure this on the scripts? > > My debug is on "3" > >> But if you run in debug mode with a high value and nothing is printed, then the SIP traffic does not reach the>application. A reason for that might be a firewall, do you run one? > The SIP traffic is arriving the machine (at least), i have checked > this with the ngrep on the server running kamailio: > > # > U 2010/11/18 10:26:03.331925 212.230.253.164:43686 -> 212.230.19.191:5060 > REGISTER sip:212.230.19.191 SIP/2.0' > Via: SIP/2.0/UDP 10.134.16.164:43686;branch=z9hG4bK247264420;rport' > From:<sip:101@212.230.19.191>;tag=1251867918' > To:<sip:101@212.230.19.191>' > Call-ID: 2022066534-16466-1(a)BA.BDE.BG.BGE' > CSeq: 4647 REGISTER' > Contact:<sip:101@10.134.16.164:43686>;reg-id=2;+sip.instance="<urn:uuid:00000000-0000-1000-8000-000B821FA066>"' > Max-Forwards: 70' > User-Agent: Grandstream GXV3140 1.0.1.1' > Supported: path' > Expires: 3600' > Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, > REFER, UPDATE, MESSAGE' > Content-Length: 0' > ' > > # > U 2010/11/18 10:26:05.339925 212.230.253.164:43686 -> 212.230.19.191:5060 > REGISTER sip:212.230.19.191 SIP/2.0' > Via: SIP/2.0/UDP 10.134.16.164:43686;branch=z9hG4bK247264420;rport' > From:<sip:101@212.230.19.191>;tag=1251867918' > To:<sip:101@212.230.19.191>' > Call-ID: 2022066534-16466-1(a)BA.BDE.BG.BGE' > CSeq: 4647 REGISTER' > Contact:<sip:101@10.134.16.164:43686>;reg-id=2;+sip.instance="<urn:uuid:00000000-0000-1000-8000-000B821FA066>"' > Max-Forwards: 70' > User-Agent: Grandstream GXV3140 1.0.1.1' > Supported: path' > Expires: 3600' > Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, > REFER, UPDATE, MESSAGE' > Content-Length: 0' > > And i never gets a response from Kamailio, for example, a 200 OK, or a > 407 unauthorized... > > What can i do? > >> Cheers, >> Daniel >> >> -- >> Daniel-Constantin Mierla >> http://www.asipto.com > > -- > Ing. Danny Dias > www.DannTEL.net -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Trainings Nov 22-25, 2010, Berlin, Germany Jan 24-26, 2011, Irvine, CA, USA http://www.asipto.com

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kamctl get number of subscribers
by Jon Bonilla 24 Nov '10

24 Nov '10

Hi all I'm trying to get the number of registered subscribers in a kamailio 3.0.4 setup. I'm trying to use usrloc's exported function "ul_get_num_users" but I get a 500 error: kamctl fifo ul_get_num_users 500 command 'ul_get_num_users' not available Using get_statistics I get: test3:~# kamctl fifo get_statistics usrloc: usrloc:registered_users = 2 usrloc:location-users = 2 usrloc:location-contacts = 2 usrloc:location-expires = 0 Trying to query for a specific stat, seems like I do not use the correct syntax: test3:~# kamctl fifo get_statistics usrloc:registered_users 404 Statistics Not Found Anyone could help me with the correct syntax for the function and fifo? thanks

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SER proxy drops incoming SIP requests
by Igor Potjevlesch 24 Nov '10

24 Nov '10

Hello, After days of debugging, I'll need some help from the SER specialists to troubleshoot this problem. The SER version in service is: 0.9.9+cvs20090925 It's running on a Red Hat EL 5.3 server. I currently have about 10.000 users registered. I encounter several seconds (1 to 5) without replies from the SIP proxy every 30 seconds. Some requests are delayed, but most of them are not replied at all. At the same time, all the 6 SER childs (forking is enabled) get about 100% used CPU (from "top") After some troubleshooting, I found that the rythm of those deny of service was exactly the time configured as the "timer_interval" parameter of "usrloc" module. When changing the parameter, problem rythm changes as well. What we did so far: - Check DNS and MySQL performances - Moved the number of childs from 6 to 9 - Changed the "timer_interval" ("usrloc" module) from 60 to 900 - Stopped the "natping" feature ("nathelper" module) as it impacted the service as well. Please let me know if the problem has already been observed and what would be the best way to go deeper in the troubleshooting. I need to solve that as soon as possible: some of our users higly suffer of random REGISTER not beeing taken into account. Sincerely, Igor

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Unforcing RTPProxy for a Cancel
by Joe Hart 24 Nov '10

24 Nov '10

I'm trying to figure out if there is some way to identify a call for which an rtpproxy session has been set up when the call is cancelled. I can't check the flag I set for the call, nor can I check the route param I set (which is what I'm doing to figure out when to unforce the media stream for a BYE). If this is not possible, is there any danger in calling unforce_rtp_proxy() for all calls, regardless of whether media proxying has been set up for the call or not? -- Joe Hart Voice Systems Integrator Evariste Systems LLC 1170 Peachtree Street 12th Floor, Suite 1200 Atlanta, GA 30309 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web : http://www.evaristesys.com/

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How to relay to ITSP or secure gateway.
by Johny Kadarisman Kwan 24 Nov '10

24 Nov '10

Hi there, Is there a way to relay call to secure gateway or ITSP. ie, invite being challange to provide authorization (username/password) Could someone please give me some advice if this is possible. Thanks, Jkwan

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TCP Children and Connection Handling
by Jijo 24 Nov '10

24 Nov '10

Hi All, I'm facing an issue with TCP connection from Proxy(kamailio) to a SIP server on a load test, where kamailio is not processing the message fast enough and which leads socket buffer getting filled up. I have increased the tcp_rd_buf_size but that really didn't help. According to my understanding the each TCP connection is assigned to a TCP children, and TCP children will read until it there is no data for few seconds and return the connection to tcp main process. In this case the TCP connection is to the SIP server which always have some data due to periodic SIP ping ,registration refresh and calls etc. So i believe this connection will be hold always by one children and since that children is processing the message sequentially, it might not be possible to increase the performance on this connection. My question is, If my understanding is correct How do we increase the performance for a particular TCP connection? Does Kamailio has some mechanism to distribute the message from a single TCP connection to different child, when it is overloaded? What is the best way to solve this issue? Thanks in advance. Jijo

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avp_db_query returning wrong data
by Jijo 23 Nov '10

23 Nov '10

Hi All, While doing 5 REGISTER/second, i observed that that avp_db_query is returning data from a different db entry. This happens only with TCP with more than 1 tcp_children. it works fine with UDP. It also work with TCP with tcp_children as 1. Is there any relation between the TCP connections and Postgres in kamailio. I'm using kamilio 3.1.0. How should i debug this issue?? What are the areas need to look? Thanks Jijo

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SIP Routing Logic in Lua
by Daniel-Constantin Mierla 22 Nov '10

22 Nov '10

Hello, during last days I spent some time to extend the native API exported to Lua. Many more functions exported by core and modules can be called from embedded Lua scripts. Note that you get also access to psedo-variable operations and you can call all functions exported by modules that have no fixup pretty safe, via sr.modf(...), without a need to be exported natively. I made an easy-to-do tutorial where all the SIP routing logic is implemented in a Lua script (including authentication, accounting, registrar, user location). You can read it at: http://asipto.com/u/h Therefore, if the configuration file language is not offering everything you need to route your SIP traffic, look at app_lua module and its API, it is a real option right now. Lua is a small and fast embeddable language, the interpreter is linked to and loaded by Kamailio at statup, therefore the performance penalty is not significant (Lua project site is: http://www.lua.org). It is also popular among other SIP telephony projects. Any kind of feedback is appreciated! Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Trainings Nov 22-25, 2010, Berlin, Germany Jan 24-26, 2011, Irvine, CA, USA http://www.asipto.com

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Nuno Ribeiro wants to stay in touch on LinkedIn
by Nuno Ribeiro via LinkedIn 22 Nov '10

22 Nov '10

LinkedIn ------------Nuno Ribeiro requested to add you as a connection on LinkedIn: ------------------------------------------ Luis, I'd like to add you to my professional network on LinkedIn. - Nuno Ribeiro Accept invitation from Nuno Ribeiro http://www.linkedin.com/e/-y8k3ym-ggtfgz8y-47/XkppUi8mK5TI9VJsZIpnfBovln4Su… View invitation from Nuno Ribeiro http://www.linkedin.com/e/-y8k3ym-ggtfgz8y-47/XkppUi8mK5TI9VJsZIpnfBovln4Su… ------------------------------------------ Why might connecting with Nuno Ribeiro be a good idea? Nuno Ribeiro's connections could be useful to you: After accepting Nuno Ribeiro's invitation, check Nuno Ribeiro's connections to see who else you may know and who you might want an introduction to. Building these connections can create opportunities in the future. -- (c) 2010, LinkedIn Corporation

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Double-quote with t_uac_dlg
by Kévin L'heveder 22 Nov '10

22 Nov '10

Hi, I have some troubles to send via t_uac_dlg a REGISTER method with an authorization field cause the values like digest, realm...are containing between double-quotes and that produce a parse error in kamailio. I didn't find how can i escape those double-quotes to prevent from this error. Here's the error logged: Nov 22 15:11:07 [14258] DBG:mi_datagram:mi_datagram_parse_node: we have a quoted value, "From: <sip:131@192.168.40.150>To: <sip:131@192.168.40.150>Contact: <sip:131@192.168.40.210:53000>Authorization: Digest username="131", realm="Asterisk/CM-YAHSAT", algorithm=MD5, uri="sip:192.168.40.202", nonce="5c87d7ed", response="fc126aadd3ba6b6ffbf38aea6be02122"Call-ID: b1ab82011346ff36(a)192.168.40.101CSeq: 10036 REGISTERExpires: 4294967295"Nov 22 15:11:07 [14258] DBG:mi_datagram:mi_datagram_parse_node: " found p is "131", realm="Asterisk/CM-YAHSAT", algorithm=MD5, uri="sip:192.168.40.202", nonce="5c87d7ed", response="fc126aadd3ba6b6ffbf38aea6be02122"Call-ID: b1ab82011346ff36(a)192.168.40.101CSeq: 10036 REGISTERExpires: 4294967295"Nov 22 15:11:07 [14258] DBG:mi_datagram:mi_datagram_parse_node: we have reached the end of attr value, p is "131", realm="Asterisk/CM-YAHSAT", algorithm=MD5, uri="sip:192.168.40.202", nonce="5c87d7ed", response="fc126aadd3ba6b6ffbf38aea6be02122"Call-ID: b1ab82011346ff36(a)192.168.40.101CSeq: 10036 REGISTERExpires: 4294967295"Nov 22 15:11:07 [14258] DBG:mi_datagram:mi_datagram_parse_node: attr value <From: <sip:131@192.168.40.150>To: <sip:131@192.168.40.150>Contact: <sip:131@192.168.40.210:53000>Authorization: Digest username=> foundNov 22 15:11:07 [14258] ERROR:mi_datagram:mi_datagram_parse_node: didn't find newline case1 Nov 22 15:11:07 [14258] ERROR:mi_datagram:mi_datagram_parse_node: parse error around 1Nov 22 15:11:07 [14258] ERROR:mi_datagram:mi_datagram_parse_tree: parse error!Nov 22 15:11:07 [14258] ERROR:mi_datagram:mi_datagram_server: failed to parse the MI tree regards,Koon

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SCTP multihoming support in Kamailio
by mosbah abdelkader 19 Nov '10

19 Nov '10

Hello, Does Kamailio supports SCTP multihoming?

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Re: [SR-Users] iphone
by oscar ying 19 Nov '10

19 Nov '10

Daniel, I can log into serweb.iptel.org with my username 'netlink83' and my mobile number is saved in my account. I hope that is enough to have my mobile phone registered. Maybe I should register to iptel again. What numbers can I call via iptel? How do I configure my own pstn gateway? Is there instructions on iptel.org? Thanks, Oscar Ying 650 776 9821 oscar.ying(a)yahoo.com ________________________________ From: Daniel-Constantin Mierla <miconda(a)gmail.com> To: oscar ying <oscar.ying(a)yahoo.com> Cc: kamailio <sr-users(a)lists.sip-router.org> Sent: Thu, November 18, 2010 2:02:18 AM Subject: Re: [SR-Users] iphone On 11/18/10 4:13 AM, oscar ying wrote: >I've tried two different numbers and they both fail. I can make regular >mobile calls but not VoIP calls via my SIP softphone (media5fone). What >do you mean if my phone is registered? > A sip phone register to SIP server with a username and password - that is what I mean with registered. Note that iptel.org does not allow you to call to pstn, unless you have your own pstn gateway configured. You can do voip calls for free. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Trainings Nov 22-25, 2010, Berlin, Germany Jan 24-26, 2011, Irvine, CA, USA http://www.asipto.com

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Storing multiple SQL results in htable/mtree
by Robert McGilvray 19 Nov '10

19 Nov '10

Hello, I'm interested in using the htable and/or mtree modules to keep my aliases and did lookups in memory to minimize the amount of SQL lookups that are being performed. It's easy enough to use htable to cache a single value after it is retrieved but what I'm not clear on is how I can do it when SQL returns multiple destinations. Consider the following query and returned results: SELECT username,domain,CAST(q_value AS CHAR) FROM aliases where alias_username = '10001' AND alias_domain = 'us.sip.globeop.com' AND active = '1' ORDER by q_value; +----------+--------------------+-----------------------+ | username | domain | CAST(q_value AS CHAR) | +----------+--------------------+-----------------------+ | 8500 | us.sip.globeop.com | 1.000 | | 3874 | us.sip.globeop.com | 1.000 | | 7050 | us.sip.globeop.com | 1.000 | | 2100 | us.sip.globeop.com | 1.000 | In my existing script I just append the results as branches and all works well but I end up with a lot of unnecessary SQL traffic. I see the htable supports array but I'm not sure how to go about assigning multiple values to a single key to use later. How can I take the above results and store them in memory for use by other transactions? Thanks, Bob -------------------------------------------------------------------------- This email with all information contained herein or attached hereto may contain confidential and/or privileged information intended for the addressee(s) only. If you have received this email in error, please contact the sender and immediately delete this email in its entirety and any attachments thereto.

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SIP Scanning Attacks Experiences
by Daniel-Constantin Mierla 19 Nov '10

19 Nov '10

Hello, during the testing period of Kamailio 3.1.0, while running it at voipuser.org, I had the chance to watch live and analyze a SIP scanning attack. Yesterday I noticed another one by looking at Siremis 2.0 charts, therefore I wrote an article with some hints about what you can use to protect your SIP services within Kamailio configuration file. You can read it at: * http://asipto.com/u/i Hope is going to be useful for many of you! Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Trainings Nov 22-25, 2010, Berlin, Germany Jan 24-26, 2011, Irvine, CA, USA http://www.asipto.com

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iphone
by oscar ying 18 Nov '10

18 Nov '10

Will my iptel SIP work with iPhone 3G and iPhone app Media5fone? Thanks, Oscar Ying 650 776 9821 oscar.ying(a)yahoo.com

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Doubts with Registration
by Danny Dias 17 Nov '10

17 Nov '10

Hello, I'm having problems with my kamailio, my phones are unable to register into the Proxy. I have been checking with ngrep-sip and all i'm receiving is many REGISTER from the phones but never getting some response from the Proxy The Proxy has a public IP and the phones are behind NAT. The weir thing is that my logs are not showing any output...and as you can see on the script, i have a section to receive REGISTER requests.... if (is_method("REGISTER")) { if (!save("location")) { sl_reply_error(); } exit; } Thanks in advance for your help

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Kamailio 1.5.4 - IP addresses WhiteList
by Miguel Baptista 17 Nov '10

17 Nov '10

Hi all, I want to enable pike module based on IP addresses. I mean ... I don't want to check if requests from some specific IP addresses trigged the Pike's threshold. but for all other IP addresses I want to check it. I was thinking about using some kind of IP addresses' whitelist. The userblacklist module has some whitelist but it's based on username/domain not IP addresses. Can someone please give me some advice how to implement it properly. Is there any module that will allow me to do it properly? Best Regards, Miguel Baptista

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Dispatcher is using gateways in Probing mode
by alex pappas 17 Nov '10

17 Nov '10

Dear all, Does anyone can explain how Dispatcher module is sending calls to the gateways in the different states? I mean if I have 3 gateways and the one of them is in Probing mode why Dispatcher is continuing sending there calls? Why this gateway is not changing from Probing to Inactive since is not replaying to OPTIONS messages send by Kamailio? Configuration: modparam("dispatcher", "flags", 2) modparam("dispatcher", "ds_ping_interval", 5) modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_probing_threshhold", 2) modparam("dispatcher", "dst_avp", "$avp(s:dst_avp)") modparam("dispatcher", "grp_avp", "$avp(dsgrp)") modparam("dispatcher", "cnt_avp", "$avp(i:202)") modparam("dispatcher", "ds_append_branch", 0) using db for gw: setid 123,destination sip:xxx.xxxx.xxx, flag 0 ... in ROUTE: if(ds_select_domain("$avp(s:disp_dstgrp)", "4")) in Failure Route if(!ds_next_domain()) I want Dispatcher when it sees a node gateway that is dead , to Not send any other calls. Thank you in advance Alex

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Creating multiple Accounts on kamailio with script.
by Siddhardha Garige 16 Nov '10

16 Nov '10

Hello all, I am conducting some load testing and need to create 50,000 accounts on kamailio. I want to write a shell script to add 50,000 users. Kamctl add [user] [password] prompts for a password and I checked kamctl -help and there is no option for providing password along with the command. Is is possible to disable password and run my script? Thanks Sid "May the light be with you." ______________________________________________ Siddhardha Garige www.luminepixels.com

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Regarding SER Listening on Custom Port
by Saurabh Srivastava 16 Nov '10

16 Nov '10

Hello SER Team, We have made SER listen on two ports via adding this command (listen=eth0:5060 eth0:5062) in ser.cfg file. Now we are able to get the server listening on two ports as 5060 & 5062 and it is sending 200 OK for the user agents which try to communicate on both the ports. SER is able to register two user agents, one on 5060 and one on 5062. But when we are trying to make a communication between the two registered user agents (i.e. one on 5060 and other on 5062) then no communication is possible (i.e. we cannot make calls and chats). Though when the two user agents are registered on same port (i.e. both UAs on 5060 or both UAs on 5062) then communication is possible between the two clients (i.e. we can make calls and chats). Can you people please guide me through that how can I make the 2 UAs registered to SER on different ports (i.e. one on 5060 and other on 5062) communicate with each other (i.e. they can make calls and chats). I have attached a log as well which shows the same problem. This is a wireshark captured output on the machine which has SER installed. Thanks and Best Regards Saurabh Srivastava

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Handle '486 busy here' from upstream carrier
by JR Richardson 15 Nov '10

15 Nov '10

Hi All, Asterisk><kamailio 3.0.2><sip carrier I see normal tansactions until the carrier send back a '486 Busy Here' then kamailio sends an 'INVITE SDP', then I get a '100 trying' from the carrier then a '482 Loop Detected" from the carrier. Shouldn't kamailio relay the 486 to the original invite server and not send SDP? I'm guessing I'm not handling 486 propperly in my config: http://pastebin.com/crfMe81D Here is a pastebin of the call graph: http://pastebin.com/rnQZDyFU I was thinking about including this in my failure route: if (t_check_status("486")) { append_branch(); t_relay(); } Would that do any good? Thanks. JR -- JR Richardson Engineering for the Masses

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Problem with Kamailio not routing ACK to a 200 OK
by Morten Isaksen 15 Nov '10

15 Nov '10

Hi, I have a strange problem with Kamailio 3.0.2. When one of our end users makes a call Kamailio does not route the ACK (in response to the 200 OK). For all other end users it works fine. For me it looks the the has_totag() checks for some reason fails and then t_check_trans() thinks it is a ACK to a local transactions and then terminates the script. Otherwise there should be more lines in the log file. The conf is pretty standard. route{ xlog("L_WARN", "New request - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); xlog("L_WARN", "ua=$ua"); 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; } if (has_totag()) { xlog("L_WARN", "has_totag start - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { xlog("L_WARN", "loose_route - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests #route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { xlog("L_WARN", "ACK t_check_trans - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); # non loose-route, but stateful ACK; must be an ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { xlog("Ignoring ACK - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); # ACK without matching transaction ... ignore and discard.\n"); exit; } } sl_send_reply("404","Not here"); } xlog("L_WARN", "has_totag end - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); exit; } #initial requests # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } setflag(4); t_check_trans(); ... The log files show: Oct 28 10:51:13 sip-core-1 /usr/local/sbin/kamailio[10503]: WARNING: <script>: New request - M=ACK RURI=sip:1105@178.21.248.56:5060 F=sip:87776688@sip.uni-tel.dk T=sip:869 87106(a)sip.uni-tel.dk IP=178.21.248.20 ID=AI231CA9BD0A4A1C53(a)10.0.0.150 Oct 28 10:51:13 sip-core-1 /usr/local/sbin/kamailio[10503]: WARNING: <script>: ua=Aastra Intelligate The trace U 2010/10/28 10:51:02.616337 178.21.248.7:5060 -> 178.21.248.56:5060 INVITE sip:86987106@178.21.248.56 SIP/2.0. Record-Route: <sip:178.21.248.7;lr;ftag=AI8DA85D59B9B6634F;did=f48.5fd1db02>. Record-Route: <sip:178.21.248.20;lr;ftag=AI8DA85D59B9B6634F>. Via: SIP/2.0/UDP 178.21.248.7;branch=z9hG4bK690c.9cfea506.0. Via: SIP/2.0/UDP 178.21.248.20;branch=z9hG4bK690c.97354e4.0. Via: SIP/2.0/UDP 87.104.233.108:5060;rport=5060;branch=z9hG4bK3c70fae55c8b362ff34e6c782cc21592. To: <sip:86987106@sip.uni-tel.dk>. From: <sip:87776688@sip.uni-tel.dk>;tag=AI8DA85D59B9B6634F. Call-ID: AI231CA9BD0A4A1C53(a)10.0.0.150. CSeq: 2 INVITE. Max-Forwards: 68. Contact: <sip:87776688@87.104.233.108:5060;line=AI7EFC34995E724DD7>. Accept: application/sdp. Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER. P-Preferred-Identity: <sip:87776688@sip.uni-tel.dk>. Privacy: none. User-Agent: Aastra Intelligate. Content-Type: application/sdp. Content-Length: 280. X-trunktype: IC. . v=0. o=intelligate 1194032777 1194032777 IN IP4 87.104.233.106. s=call. c=IN IP4 178.21.248.22. t=0 0. m=audio 60984 RTP/AVP 8 0 18 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=ptime:20. # U 2010/10/28 10:51:02.636854 178.21.248.56:5060 -> 178.21.248.7:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 178.21.248.7;branch=z9hG4bK690c.9cfea506.0. Via: SIP/2.0/UDP 178.21.248.20;branch=z9hG4bK690c.97354e4.0. Via: SIP/2.0/UDP 87.104.233.108:5060;rport=5060;branch=z9hG4bK3c70fae55c8b362ff34e6c782cc21592. From: <sip:87776688@sip.uni-tel.dk>;tag=AI8DA85D59B9B6634F. To: <sip:86987106@sip.uni-tel.dk>;tag=1c2073920452. Call-ID: AI231CA9BD0A4A1C53(a)10.0.0.150. CSeq: 2 INVITE. Supported: em,timer,replaces,path,early-session,resource-priority. Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE. Server: Audiocodes-Sip-Gateway-Mediant 2000/v.5.60A.035.002. Content-Length: 0. . ##### U 2010/10/28 10:51:04.134445 178.21.248.56:5060 -> 178.21.248.7:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 178.21.248.7;branch=z9hG4bK690c.9cfea506.0. Via: SIP/2.0/UDP 178.21.248.20;branch=z9hG4bK690c.97354e4.0. Via: SIP/2.0/UDP 87.104.233.108:5060;rport=5060;branch=z9hG4bK3c70fae55c8b362ff34e6c782cc21592. From: <sip:87776688@sip.uni-tel.dk>;tag=AI8DA85D59B9B6634F. To: <sip:86987106@sip.uni-tel.dk>;tag=1c2073920452. Call-ID: AI231CA9BD0A4A1C53(a)10.0.0.150. CSeq: 2 INVITE. Contact: <sip:1105@178.21.248.56:5060>. Record-Route: <sip:178.21.248.7;lr;ftag=AI8DA85D59B9B6634F;did=f48.5fd1db02>,<sip:178.21.248.20;lr;ftag=AI8DA85D59B9B6634F>. Supported: em,timer,replaces,path,early-session,resource-priority. Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE. Server: Audiocodes-Sip-Gateway-Mediant 2000/v.5.60A.035.002. Content-Type: application/sdp. Content-Length: 259. . v=0. o=AudiocodesGW 2073965634 2073965290 IN IP4 178.21.248.56. s=Phone-Call. c=IN IP4 178.21.248.56. t=0 0. m=audio 6050 RTP/AVP 8 101. c=IN IP4 178.21.248.56. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. # U 2010/10/28 10:51:04.134748 178.21.248.7:5060 -> 178.21.248.20:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 178.21.248.20;branch=z9hG4bK690c.97354e4.0. Via: SIP/2.0/UDP 87.104.233.108:5060;rport=5060;branch=z9hG4bK3c70fae55c8b362ff34e6c782cc21592. From: <sip:87776688@sip.uni-tel.dk>;tag=AI8DA85D59B9B6634F. To: <sip:86987106@sip.uni-tel.dk>;tag=1c2073920452. Call-ID: AI231CA9BD0A4A1C53(a)10.0.0.150. CSeq: 2 INVITE. Contact: <sip:1105@178.21.248.56:5060>. Record-Route: <sip:178.21.248.7;lr;ftag=AI8DA85D59B9B6634F;did=f48.5fd1db02>,<sip:178.21.248.20;lr;ftag=AI8DA85D59B9B6634F>. Supported: em,timer,replaces,path,early-session,resource-priority. Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE. Server: Audiocodes-Sip-Gateway-Mediant 2000/v.5.60A.035.002. Content-Type: application/sdp. Content-Length: 259. . v=0. o=AudiocodesGW 2073965634 2073965290 IN IP4 178.21.248.56. s=Phone-Call. c=IN IP4 178.21.248.56. t=0 0. m=audio 6050 RTP/AVP 8 101. c=IN IP4 178.21.248.56. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. # U 2010/10/28 10:51:04.136586 178.21.248.56:5060 -> 178.21.248.7:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 178.21.248.7;branch=z9hG4bK690c.9cfea506.0. Via: SIP/2.0/UDP 178.21.248.20;branch=z9hG4bK690c.97354e4.0. Via: SIP/2.0/UDP 87.104.233.108:5060;rport=5060;branch=z9hG4bK3c70fae55c8b362ff34e6c782cc21592. From: <sip:87776688@sip.uni-tel.dk>;tag=AI8DA85D59B9B6634F. To: <sip:86987106@sip.uni-tel.dk>;tag=1c2073920452. Call-ID: AI231CA9BD0A4A1C53(a)10.0.0.150. CSeq: 2 INVITE. Contact: <sip:1105@178.21.248.56:5060>. Record-Route: <sip:178.21.248.7;lr;ftag=AI8DA85D59B9B6634F;did=f48.5fd1db02>,<sip:178.21.248.20;lr;ftag=AI8DA85D59B9B6634F>. Supported: em,timer,replaces,path,early-session,resource-priority. Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE. Server: Audiocodes-Sip-Gateway-Mediant 2000/v.5.60A.035.002. Content-Type: application/sdp. Content-Length: 259. . v=0. o=AudiocodesGW 2073965634 2073965290 IN IP4 178.21.248.56. s=Phone-Call. c=IN IP4 178.21.248.56. t=0 0. m=audio 6050 RTP/AVP 8 101. c=IN IP4 178.21.248.56. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. # U 2010/10/28 10:51:04.136837 178.21.248.7:5060 -> 178.21.248.20:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 178.21.248.20;branch=z9hG4bK690c.97354e4.0. Via: SIP/2.0/UDP 87.104.233.108:5060;rport=5060;branch=z9hG4bK3c70fae55c8b362ff34e6c782cc21592. From: <sip:87776688@sip.uni-tel.dk>;tag=AI8DA85D59B9B6634F. To: <sip:86987106@sip.uni-tel.dk>;tag=1c2073920452. Call-ID: AI231CA9BD0A4A1C53(a)10.0.0.150. CSeq: 2 INVITE. Contact: <sip:1105@178.21.248.56:5060>. Record-Route: <sip:178.21.248.7;lr;ftag=AI8DA85D59B9B6634F;did=f48.5fd1db02>,<sip:178.21.248.20;lr;ftag=AI8DA85D59B9B6634F>. Supported: em,timer,replaces,path,early-session,resource-priority. Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE. Server: Audiocodes-Sip-Gateway-Mediant 2000/v.5.60A.035.002. Content-Type: application/sdp. Content-Length: 259. . v=0. o=AudiocodesGW 2073965634 2073965290 IN IP4 178.21.248.56. s=Phone-Call. c=IN IP4 178.21.248.56. t=0 0. m=audio 6050 RTP/AVP 8 101. c=IN IP4 178.21.248.56. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. ############################## U 2010/10/28 10:51:12.881179 178.21.248.56:5060 -> 178.21.248.7:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 178.21.248.7;branch=z9hG4bK690c.9cfea506.0. Via: SIP/2.0/UDP 178.21.248.20;branch=z9hG4bK690c.97354e4.0. Via: SIP/2.0/UDP 87.104.233.108:5060;rport=5060;branch=z9hG4bK3c70fae55c8b362ff34e6c782cc21592. From: <sip:87776688@sip.uni-tel.dk>;tag=AI8DA85D59B9B6634F. To: <sip:86987106@sip.uni-tel.dk>;tag=1c2073920452. Call-ID: AI231CA9BD0A4A1C53(a)10.0.0.150. CSeq: 2 INVITE. Contact: <sip:1105@178.21.248.56:5060>. Record-Route: <sip:178.21.248.7;lr;ftag=AI8DA85D59B9B6634F;did=f48.5fd1db02>,<sip:178.21.248.20;lr;ftag=AI8DA85D59B9B6634F>. Supported: em,timer,replaces,path,early-session,resource-priority. Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE. Server: Audiocodes-Sip-Gateway-Mediant 2000/v.5.60A.035.002. Content-Type: application/sdp. Content-Length: 259. . v=0. o=AudiocodesGW 2073965634 2073965290 IN IP4 178.21.248.56. s=Phone-Call. c=IN IP4 178.21.248.56. t=0 0. m=audio 6050 RTP/AVP 8 101. c=IN IP4 178.21.248.56. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. # U 2010/10/28 10:51:12.882388 178.21.248.7:5060 -> 178.21.248.20:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 178.21.248.20;branch=z9hG4bK690c.97354e4.0. Via: SIP/2.0/UDP 87.104.233.108:5060;rport=5060;branch=z9hG4bK3c70fae55c8b362ff34e6c782cc21592. From: <sip:87776688@sip.uni-tel.dk>;tag=AI8DA85D59B9B6634F. To: <sip:86987106@sip.uni-tel.dk>;tag=1c2073920452. Call-ID: AI231CA9BD0A4A1C53(a)10.0.0.150. CSeq: 2 INVITE. Contact: <sip:1105@178.21.248.56:5060>. Record-Route: <sip:178.21.248.7;lr;ftag=AI8DA85D59B9B6634F;did=f48.5fd1db02>,<sip:178.21.248.20;lr;ftag=AI8DA85D59B9B6634F>. Supported: em,timer,replaces,path,early-session,resource-priority. Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE. Server: Audiocodes-Sip-Gateway-Mediant 2000/v.5.60A.035.002. Content-Type: application/sdp. Content-Length: 259. . v=0. o=AudiocodesGW 2073965634 2073965290 IN IP4 178.21.248.56. s=Phone-Call. c=IN IP4 178.21.248.56. t=0 0. m=audio 6050 RTP/AVP 8 101. c=IN IP4 178.21.248.56. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. ##### <<<<<<<<<<<<<<<<<<<<<< This is the problem packet >>>>>>>>>>>>>>>>>>>>> U 2010/10/28 10:51:13.267863 178.21.248.20:5060 -> 178.21.248.7:5060 ACK sip:1105@178.21.248.56:5060 SIP/2.0. Record-Route: <sip:178.21.248.20;lr;ftag=AI8DA85D59B9B6634F>. Via: SIP/2.0/UDP 178.21.248.20;branch=z9hG4bK690c.97354e4.2. Via: SIP/2.0/UDP 87.104.233.108:5060;rport=5060;branch=z9hG4bK07103fe69c8af048b9b8216eb2f7233f. To: <sip:86987106@sip.uni-tel.dk>;tag=1c2073920452. From: <sip:87776688@sip.uni-tel.dk>;tag=AI8DA85D59B9B6634F. Call-ID: AI231CA9BD0A4A1C53(a)10.0.0.150. CSeq: 2 ACK. Max-Forwards: 69. Route: <sip:178.21.248.7;lr=on;ftag=AI8DA85D59B9B6634F;did=f48.5fd1db02>. User-Agent: Aastra Intelligate. Content-Length: 0. -- Morten Isaksen

2 8

append_branch
by Rizwan Qureshi 15 Nov '10

15 Nov '10

Hi All, I just want to know how I can use append_branch() function for register requests. What I want to do is register all users on all asterisk servers by forwarding the register requests (to all asterisk servers) and then if the sip proxy throws call to any of the server the asterisk server, the call should be successful. If its possible then please let me know. Cheers -- Best Regards Rizwan Qureshi

2 1

Re: [SR-Users] advertised_address and RR
by Ovidiu Sas 15 Nov '10

15 Nov '10

On Mon, Nov 15, 2010 at 8:04 AM, Torbjörn Abrahamsson <torbjorn.abrahamsson(a)gmail.com> wrote: > Hello. > > > We are trying to use Kamailio 3.0.3 behind a NAT. After reading the cookbook > we added the advertised_address parameter set to the public IP in our cfg. > This works for the via-header but the record-route-header still uses the > internal IP. Shouldn’t this be changed as well? The documentation says that > advertised_address should apply to RR. advertised_address parameter works for Via headers. > We have also tried to use record_route_preset, but still ends up with the > internal IP in the record-route-header. record_route_preset and record_route are exclusive - use one or the other but not both for the same message Regards, Ovidiu Sas

2 1

advertised_address and RR
by Torbjörn Abrahamsson 15 Nov '10

15 Nov '10

Hello. We are trying to use Kamailio 3.0.3 behind a NAT. After reading the cookbook we added the advertised_address parameter set to the public IP in our cfg. This works for the via-header but the record-route-header still uses the internal IP. Shouldnt this be changed as well? The documentation says that advertised_address should apply to RR. We have also tried to use record_route_preset, but still ends up with the internal IP in the record-route-header. We found some discussions about this when googleing, dating back to 2005. We also found an open-ser-bugreport regarding this. Neither of these resources provided an insight in what we have missed. BR, Torbjörn Abrahamsson

1 0

siremis 2.0.0 CDRS link issues !
by X- x -X 15 Nov '10

15 Nov '10

Hi all, I was wondering if Im the only one getting this error (in siremis 2.0.0 with kamailio 3.1). i.e. I have activated the accounting modules & all but still when I click the "CDR list" in "Accounting Services" it errors on me saying that CDRS table is missing, which ofcourse is missing from the "openser" mysql DB. Why is CDRS table missing ? Is it okay (advisable) to create a CDRS table mentioned in siremis 1.0 docs (http://siremis.asipto.com/install-accounting/ ) Cheers --- X [2010-11-12 14:01:08 (GMT)] An exception occurred while executing this script: Error message: #0, Error in query:: SELECT T0.cdr_id, T0.src_username, T0.src_domain, T0.dst_username, T0.dst_domain, T0.call_start_time, T0.duration, T0.sip_call_id, T0.sip_from_tag, T0.sip_to_tag, T0.src_ip, T0.cost, T0.created FROM `cdrs` T0 LIMIT 10. SQLSTATE[42S02]: Base table or view not found: 1146 Table 'openser.cdrs' doesn't exist Script name and line number of error: /var/www/siremis-2.0.0/openbiz/bin/data/BizDataObj_Lite.php:447 function: _run_search ( Array(2) ) @ /var/www/siremis-2.0.0/openbiz/bin/data/BizDataObj_Lite.php 243 function: fetch ( ) @ /var/www/siremis-2.0.0/openbiz/bin/easy/EasyForm.php 690 function: fetchDataSet ( ) @ /var/www/siremis-2.0.0/openbiz/bin/easy/FormRenderer.php 69 function: renderSmarty ( Object(EasyForm), "/var/www/siremis-2.0.0/siremis/modules/ser/template/grid.tpl" ) @ /var/www/siremis-2.0.0/openbiz/bin/easy/FormRenderer.php 45 function: render ( Object(EasyForm) ) @ /var/www/siremis-2.0.0/openbiz/bin/easy/EasyForm.php 1648 function: renderHTML ( ) @ /var/www/siremis-2.0.0/openbiz/bin/easy/EasyForm.php 1538 function: render ( ) @ /var/www/siremis-2.0.0/openbiz/bin/easy/ViewRenderer.php 83 function: renderSmarty ( Object(EasyView), "/var/www/siremis-2.0.0/siremis/modules/ser/template/view.tpl" ) @ /var/www/siremis-2.0.0/openbiz/bin/easy/ViewRenderer.php 40 function: render ( Object(EasyView) ) @ /var/www/siremis-2.0.0/openbiz/bin/easy/EasyView.php 274 function: _render ( ) @ /var/www/siremis-2.0.0/openbiz/bin/easy/EasyView.php 237 function: render ( ) @ /var/www/siremis-2.0.0/openbiz/bin/BizController.php 221 function: renderView ( "ser.view.CdrsListView", "", "", Null, "" ) @ /var/www/siremis-2.0.0/openbiz/bin/BizController.php 107 function: dispatchRequest ( ) @ /var/www/siremis-2.0.0/openbiz/bin/BizController.php 32 function: include_once ( "/var/www/siremis-2.0.0/openbiz/bin/BizController.php" ) @ /var/www/siremis-2.0.0/siremis/bin/controller.php 6 function: include ( "/var/www/siremis-2.0.0/siremis/bin/controller.php" ) @ /var/www/siremis-2.0.0/siremis/bin/_forward.php 90 function: include ( "/var/www/siremis-2.0.0/siremis/bin/_forward.php" ) @ /var/www/siremis-2.0.0/siremis/index.php 3 Please ask system administrator for help...

2 1

Siremis 2.0: error in LCR table editing??
by Klaus Feichtinger 15 Nov '10

15 Nov '10

Hello list, has anybody else tried using SIREMIS 2.0 for administrating LCR tables in kamailio 3.1.0? I am not happy with the form / input boxes that are presented in the web browser, because some fields are not as expected..... Following points are - from my point of view - incorrect: 1) table "LCR_Gateway_List": the fields 'URI-scheme', 'Transport' and 'Strip' are represented by checkboxes. However, in reality these fields are defined with smallint(3) types and e.g. transport is representing values from NULL...4 - not boolean! 2) table "LCR_Target_List": the field priority is represented by a checkbox, too. However, also this field is not boolean. Priorities are represented by numbers, starting with '0' = highest priority a.s.o. Can anybody check if this "problem" is a real one, or only present on my PC? I have tested with IE and Firefox - the result is still the same. regards, Klaus

5 6

Call from web
by Sergey Babichev 12 Nov '10

12 Nov '10

2 2

Re: [SR-Users] Question about the Q-value.
by Kosilov Fedor 12 Nov '10

12 Nov '10

OK, and how do you usually deal with this? I was thinking of putting the Q-value into the "Display Name" field, so when the device registers, I could get it using $fn (reference to display name of 'From' header) and put it directly to the location table in the database...

4 6

help in kamailio and tunnel server
by hala alramli 12 Nov '10

12 Nov '10

i want to instal tunnel server in kamailio . i want to ask if there is any opensource voip tunnel server . and if there is an tutorial to configure kamailio with tunnel server thanks for help hala

2 1

Re: [SR-Users] noack and t_relay erros errors
by JR Richardson 11 Nov '10

11 Nov '10

> Hi All, > > I'm still getting these errors and I'm struggling to resolve the > problem. I think I'm missing an append_branch or something simple in > my config, a little guiedance will be appriciated. > > The error: > > Nov 11 10:23:26 sip-router1 /usr/local/sbin/kamailio[23739]: ERROR: tm > [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding > Nov 11 10:23:26 sip-router1 /usr/local/sbin/kamailio[23739]: ERROR: tm > [tm.c:1300]: ERROR: w_t_relay_to: t_relay_to failed > > The sip graph causing the error: > > http://pastebin.com/VHhQ1sJY > > My config: > > http://pastebin.com/BYn4g5ur > > Do I need to have a mechanism to deal with a 486 form the carrier? > > Thanks. > > JR > -- > JR Richardson > Engineering for the Masses > > > > ------------------------------ > > Message: 4 > Date: Thu, 11 Nov 2010 11:52:21 -0500 > From: Alex Balashov <abalashov(a)evaristesys.com> > Subject: Re: [SR-Users] noack and t_relay erros errors > To: sr-users(a)lists.sip-router.org > Message-ID: <4CDC1F45.1000704(a)evaristesys.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > 1. One problem may be that you are calling route[RELAY] from your > failure route and using sl_send_reply(). The route[RELAY] is still > being invoked in a failure route execution context, even if you have > managed to contextually invoke another route, or subroutine if you will. > Stateless replies (sl_reply_error()) cannot be sent from within a > failure route. > > 2. Yes. throw an append_branch() into your failure route. > > -- > Alex Balashov - Principal > Evariste Systems LLC > 1170 Peachtree Street NE > 12th Floor, Suite 1200 > Atlanta, GA 30309 > Tel: +1-678-954-0670 > Fax: +1-404-961-1892 > Web: http://www.evaristesys.com/ > Well, that seemed to help greatly, so far no more errors after changing my failure route to: failure_route[FAIL_ONE] { if (t_is_canceled()) { exit; } ds_mark_dst("P"); ds_next_dst(); append_branch(); t_relay(); } Thanks Alex, right on target again. JR -- JR Richardson Engineering for the Masses

2 2

Re: [SR-Users] Variable substitution with Dispatcher
by Lee Archer 11 Nov '10

11 Nov '10

Hi, can anyone answer this? Regards Lee From: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of Lee Archer Sent: 10 November 2010 10:25 To: sr-users(a)lists.sip-router.org Subject: [SR-Users] Variable substitution with Dispatcher Hi, I am trying to compare 3.0.4 and 3.1 and the following works in 3.1 but not 3.0.4. Should this work in 3.0 or was it only fixed for 3.1? ds_select_dst($tU,"4"); Regards Lee

2 1

noack and t_relay erros errors
by JR Richardson 11 Nov '10

11 Nov '10

Hi All, I'm still getting these errors and I'm struggling to resolve the problem. I think I'm missing an append_branch or something simple in my config, a little guiedance will be appriciated. The error: Nov 11 10:23:26 sip-router1 /usr/local/sbin/kamailio[23739]: ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding Nov 11 10:23:26 sip-router1 /usr/local/sbin/kamailio[23739]: ERROR: tm [tm.c:1300]: ERROR: w_t_relay_to: t_relay_to failed The sip graph causing the error: http://pastebin.com/VHhQ1sJY My config: http://pastebin.com/BYn4g5ur Do I need to have a mechanism to deal with a 486 form the carrier? Thanks. JR -- JR Richardson Engineering for the Masses

2 2

Extension state
by Lucas Alvarez 10 Nov '10

10 Nov '10

Hi, is there a way "database driven" to know who is using the phone? Thanks in advance. Lucas

2 1

multidomain: running multiple kamailio on one host
by MÉSZÁROS Mihály 10 Nov '10

10 Nov '10

Hello all! Is there any known problem to run multiple kamailio/sip-router instance in one host. I need this to handle multidomain situation. The multidomain what is built in is not satisfactory, because the UA-s what are from varying vendors. And some has strange or a very limited SIP implementation. Some can only register ip address, some can use RFC3263, but some can use only DNS A record. :( Any answer, suggestion appreciated very well! Thanks, Misi

7 8

Asterisk 1.6 and Kamailio 3.1 realtime integration tutorial
by Daniel-Constantin Mierla 10 Nov '10

10 Nov '10

Hello, I got the time to upgrade my tutorial about Asterisk and Kamailio realtime integration to latest stable release of Kamailio, v3.1.0: You can find the document at: * http://asipto.com/u/c Hope it is useful for some people within this community. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Trainings Nov 22-25, 2010, Berlin, Germany Jan 24-26, 2011, Irvine, CA, USA http://www.asipto.com

1 0

Re: [SR-Users] "append_rpid_hf()" and "append_hf()" issues
by Dominguez Jover, Ricardo 10 Nov '10

10 Nov '10

Hi Alex, I think I'm doing what you say, look at the configuration my config file: append_rpid_hf("$fU", ";party=calling;id-type=subscriber;privacy=off;screen=yes"); consume_credentials(); (the following lines are commented) ####### I also tested the following # append_hf("Remote-Party-ID:<sip:$avp(rpid)@$avp(domain);user=phone>;privacy=$avp(privacy);party=calling\r\n"); # append_hf("P-Asserted-Identity:<sip:$avp(rpid)@$avp(domain)>\r\n", "Call-ID"); # consume_credentials(); Is that right or you mean something like this: append_rpid_hf("$avp(rpid)", ";party=calling;id-type=subscriber;privacy=off;screen=yes"); consume_credentials(); ? Anyway I've just tested this last configuration and I can't see "Remote Party ID" headers in SIP message. Kind regards, Ricardo -----Mensaje original----- De: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] En nombre de Alex Balashov Enviado el: miércoles, 10 de noviembre de 2010 11:24 Para: sr-users(a)lists.sip-router.org Asunto: Re: [SR-Users] "append_rpid_hf()" and "append_hf()" issues Ricardo, The 'rpid' value is loaded from the 'subscriber' table as part of the authentication functions; if you consume_credentials() before it, it will be gone. You should append_rpid_hf() the RPID header first, and consume_credentials() afterward. Cheers, -- Alex

1 0

Variable substitution with Dispatcher
by Lee Archer 10 Nov '10

10 Nov '10

Hi, I am trying to compare 3.0.4 and 3.1 and the following works in 3.1 but not 3.0.4. Should this work in 3.0 or was it only fixed for 3.1? ds_select_dst($tU,"4"); Regards Lee

1 0

"append_rpid_hf()" and "append_hf()" issues
by Dominguez Jover, Ricardo 10 Nov '10

10 Nov '10

Hi everybody, I've been testing and setting up kamalio 3.0.3 with MySQL since a few weeks ago. All the basic functionalities are working properly, but now I have an issue which I'm not been able to solve by myself. I've read documentation and posts, but no idea about what is happening. This is the issue. I'm trying to append Remote Party ID field to the SIP message header. I've been trying with both "append_rpid_hf()" and "append_hf()" and I can't see any Remote Party Header at the SIP messages sniffed with wireshark. Here is the configuration I've tried: #para cargar los Remote party ID de la tabla suscriber modparam("auth_db", "load_credentials", "$avp(s:rpid)=rpid") modparam("auth", "rpid_avp", "$avp(s:rpid)") modparam("auth", "rpid_prefix", "<sip:") modparam("auth", "rpid_suffix", ";user=phone>;party=calling;screen=yes;id-type=subscriber;privacy=off") #Modificacion para no pedir autenticacion si viene desde un trusted if (!allow_trusted("$si", "$(proto{s.toupper})")) { if (!proxy_authorize("umh.es", "subscriber")) { proxy_challenge("", "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; } } append_rpid_hf("$fU", ";party=calling;id-type=subscriber;privacy=off;screen=yes"); consume_credentials(); ####### I also tested the following # append_hf("Remote-Party-ID: <sip:$avp(rpid)@$avp(domain);user=phone>;privacy=$avp(privacy);party=cal ling\r\n"); # append_hf("P-Asserted-Identity: <sip:$avp(rpid)@$avp(domain)>\r\n", "Call-ID"); # consume_credentials(); } } In a similar post I've read that the user had a problem because of the domain table. I have both the domain name and the IP of the host defined, and I also get response with "$avp(s:rpid) , when RPID field is defined to a subscriber. As I said I can't find the mistake, so any help would be appreciated. Thanks in advance and best regards, Ricardo Dominguez

2 1

Re: [SR-Users] LCR database issues
by Lee Archer 10 Nov '10

10 Nov '10

Anyone any idea why these are reporting command disabled? Regards Lee From: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of Lee Archer Sent: 08 November 2010 17:27 To: sr-users(a)lists.sip-router.org Subject: [SR-Users] LCR database issues Hi, I have kamailio configured to get it's LCR info from a mysql database but when I try to get this info from the command line I am told the command is disabled. Why is this? [root@kam-3-1-s1 kamailio]# kamctl lcr show_gws ERROR: command disabled [root@kam-3-1-s1 kamailio]# kamctl lcr dump_gws ERROR: command disabled [root@kam-3-1-s1 kamailio]# kamctl lcr show_routes ERROR: command disabled [root@kam-3-1-s1 kamailio]# kamctl lcr dump_routes ERROR: command disabled [root@kam-3-1-s1 kamailio]# kamctl lcr show_gws ERROR: command disabled lcr show_gws....... show database gateways lcr show_routes.... show database routes lcr dump_gws....... show in memory gateways lcr dump_routes.... show in memory routes lcr reload ........ reload lcr gateways and routes lcr eval_weights .. evaluates probability for given GW's weights [root@kam-3-1-s1 kamailio]# kamctl lcr reload ERROR: command disabled Regards Lee

2 4

kamctl duplication of output
by Lee Archer 10 Nov '10

10 Nov '10

Is there a problem with kamctl and duplication of the existing commands? [root@kam-3-1-s1 ~]# kamctl /usr/local/sbin/kamctl 3.1.0 Existing commands: -- command 'cisco_restart' - restart CISCO phone (NOTIFY) cisco_restart <uri> ................ restart phone configured for <uri> -- command 'online' - dump online users from memory online ............................. display online users -- command 'monitor' - show internal status monitor ............................ show server's internal status -- command 'ping' - ping a SIP URI (OPTIONS) ping <uri> ......................... ping <uri> with SIP OPTIONS -- command 'ul|alias' - manage user location or aliases ul show [<username>]................ show in-RAM online users ul show --brief..................... show in-RAM online users in short format ul rm <username> [<contact URI>].... delete user's usrloc entries ul add <username> <uri> ............ introduce a permanent usrloc entry ul add <username> <uri> <expires> .. introduce a temporary usrloc entry -- command 'cisco_restart' - restart CISCO phone (NOTIFY) cisco_restart <uri> ................ restart phone configured for <uri> -- command 'online' - dump online users from memory online ............................. display online users -- command 'monitor' - show internal status monitor ............................ show server's internal status -- command 'ping' - ping a SIP URI (OPTIONS) ping <uri> ......................... ping <uri> with SIP OPTIONS -- command 'ul|alias' - manage user location or aliases ul show [<username>]................ show in-RAM online users ul show --brief..................... show in-RAM online users in short format ul rm <username> [<contact URI>].... delete user's usrloc entries ul add <username> <uri> ............ introduce a permanent usrloc entry ul add <username> <uri> <expires> .. introduce a temporary usrloc entry Regards Lee

2 1

parsing issue 3.1.0
by Jeremya 10 Nov '10

10 Nov '10

These text lines parse if ($sht(ipban=>$si)!=$null) $sht(ipban=>$si)=1; These don't parse if($sht (ipban=>$si)!=$null) $sht (ipban=>$si)=1; In both cases the gap between $sht and the opening '(' is the problem. Is this expected behaviour? Jeremy

2 1

Dispatcher reload in Kamailio 3.1
by Lee Archer 09 Nov '10

09 Nov '10

Hi, after adding another dispatcher destination I tried to reload my dispatcher but got the following: [root@kam-3-1-s1 kamailio]# kamctl dispatcher reload 500 No reload support for call load dispatching Other kamctl dispatcher commands work so am I missing something? Regards Lee

2 3

LCR database issues
by Lee Archer 09 Nov '10

09 Nov '10

Hi, I have kamailio configured to get it's LCR info from a mysql database but when I try to get this info from the command line I am told the command is disabled. Why is this? [root@kam-3-1-s1 kamailio]# kamctl lcr show_gws ERROR: command disabled [root@kam-3-1-s1 kamailio]# kamctl lcr dump_gws ERROR: command disabled [root@kam-3-1-s1 kamailio]# kamctl lcr show_routes ERROR: command disabled [root@kam-3-1-s1 kamailio]# kamctl lcr dump_routes ERROR: command disabled [root@kam-3-1-s1 kamailio]# kamctl lcr show_gws ERROR: command disabled lcr show_gws....... show database gateways lcr show_routes.... show database routes lcr dump_gws....... show in memory gateways lcr dump_routes.... show in memory routes lcr reload ........ reload lcr gateways and routes lcr eval_weights .. evaluates probability for given GW's weights [root@kam-3-1-s1 kamailio]# kamctl lcr reload ERROR: command disabled Regards Lee

2 1

no audio
by Rizwan Qureshi 08 Nov '10

08 Nov '10

Hi List, I am using dispatcher module to route everything from registerations to calls toa asterisk pbx. thats all what openser is doing at the moment. outgoing calls are fine but I get CHANUNAVAILABLE status for incoming calls on Asterisk. I guess the reverse sip dialogue are not working properly through openser. Following is my open ser config route{ 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; } #record_route(); ds_select_dst("1","2"); forward(); } I have tried with and without record_route() but nothing working for incoming calls. Any ideas?

1 0

query dispatcher status from config
by Klaus Darilion 08 Nov '10

08 Nov '10

Hi! Is it somehow possible to query the status (active/inactive/probing) of a dispatcher target from within kamailio.cfg? Thanks Klaus

1 0

Kamailio 3.1 + LCR + CARRIERROUTE howto ?
by Sébastien Cramatte 08 Nov '10

08 Nov '10

Hello, I'm new in kamailio/sr. I've installed 3.1 packages on my debian squeeze. I would like to configure LCR and CARRIERROUTE modules but I don't find suficient doc to do it properly. Does someone can give me resources where I can found : - required modparam for LCR and CARRIERROUTE - SQL examples datas for both modules - example of route logic using both modules Thank you

3 3

checking if calls are still active
by Vic Jolin 08 Nov '10

08 Nov '10

Hi, Just want to ask what is the best way to check if calls are still active especially when we are not handling media? Thanks

4 6

avp_load_radius unknown command
by Vitaly Pougatchev 07 Nov '10

07 Nov '10

Please, help me.. kamailio-3.1.0# ser -c loading modules under /usr/local/lib/ser/modules:/usr/local/lib/ser/modules_k:/usr/local/lib/ser/modules_s 0(12801) : <core> [cfg.y:3412]: parse error in config file /usr/local/etc/ser/ser.cfg, line 242, column 36: unknown command, missing loadmodule? 0(12801) : <core> [cfg.y:3412]: parse error in config file /usr/local/etc/ser/ser.cfg, line 248, column 42: unknown command, missing loadmodule? ---line 242 --- kamailio-3.1.0#cat /usr/local/etc/ser/ser.cfg | head -242 | tail -1 if(avp_load_radius("callee")) ---line 248 --- kamailio-3.1.0#cat /usr/local/etc/ser/ser.cfg | head -248 | tail -1 avp_printf("i:100", "$Tf");

1 0

Error on creating user or edit user
by Joseph Ong 07 Nov '10

07 Nov '10

siremis 2.0 new setup not sure if I made a mistake in setup of siremis. add new user in Application Menu->Users->User Management Add [Sun Nov 07 02:30:30 2010] [error] [client 1.2.3.4] PHP Parse error: syntax error, unexpected ',' in /var/www/siremis/openbiz/bin/Expression.php(330) : eval()'d code on line 1, referer: http://localhost/siremis/system/user_new ----- *function:* eval_syntax ( "$ret = joseph, 3, 10;" ) @ /var/www/siremis/openbiz/bin/Expression.php 327 *function:* evaluateExpression ( "{[fld_username], 3, 10}", Object(UserForm) ) @ /var/www/siremis/openbiz/bin/Expression.php 190 *function:* replaceVarExpr ( "@validate:betweenLength([fld_username], 3, 10)", Object(UserForm) ) @ /var/www/siremis/openbiz/bin/Expression.php 311 *function:* evaluateExpression ( "{@validate:betweenLength([fld_username], 3, 10)}", Object(UserForm) ) @ /var/www/siremis/openbiz/bin/easy/element/Element.php 472 *function:* validate ( ) @ /var/www/siremis/openbiz/bin/easy/EasyForm.php 1370 *function:* validateForm ( ) @ /var/www/siremis/siremis/modules/system/form/UserForm.php 146 *function:* validateForm ( ) @ /var/www/siremis/siremis/modules/system/form/UserForm.php 51 *function:* CreateUser ( ) @ /var/www/siremis/openbiz/bin/BizController.php 310 *function:* invoke ( ) @ /var/www/siremis/openbiz/bin/BizController.php 110 *function:* dispatchRequest ( ) @ /var/www/siremis/openbiz/bin/BizController.php 32 *function:* include_once ( "/var/www/siremis/openbiz/bin/BizController.php" ) @ /var/www/siremis/siremis/bin/controller.php 6

1 0

Re: [SR-Users] Terminated call processing Status: 487
by JR Richardson 05 Nov '10

05 Nov '10

>> I'm seeing random errors in my Kamailio 3.0.2 log: >> >> ERROR: tm [tm.c:1300]: ERROR: w_t_relay_to: t_relay_to failed >> ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding > > Are you sure those are the only errors in the log? > > I've gotten that before when getting bogus or missing 'Via' headers > from bandwidth.com in the 487s. > Here is a pastebin of the conversation, what is interesting is line 29 through 32. http://pastebin.com/QcaAeqjN For some reason Kamailio is sending 2 additional 487's without a response. I believe this is why I am getting the log errors. I played around with "modparam("tm", "cancel_b_method", 2)" but have the same results. Here is a pastebin of my kamailio.cfg: http://pastebin.com/Tn3humr0 Any suggestions on where to put some logging statements and what to look for would be helpful. Thanks. JR -- JR Richardson Engineering for the Masses

2 2

Re: [SR-Users] installing dispatcher module
by Daniel-Constantin Mierla 05 Nov '10

05 Nov '10

do not send private emails. Keep the mailing list cc-ed. Thanks, Daniel On 11/5/10 1:41 PM, Rizwan Qureshi wrote: > Thanks. Problem solved anyway. > > On Thu, Nov 4, 2010 at 10:04 PM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > Hello, > > run kamailio with -c parameter to get hints about the line where > the error happens. > > Daniel > > > On 11/4/10 4:17 PM, Rizwan Qureshi wrote: >> Hi List, >> This is my first email to this list and I am a kamailio newbie. I >> have worked on asterisk a lot but this is the first time im going >> to use kamailio. >> >> I have successfully installed kamailio on my centos system. I can >> see all the modules located in the >> /usr/local/lib/kamailio/modules_k/ AND >> /usr/local/lib/kamailio/modules/ directories. But the problem is >> when i try to load the dispatcher module from kamailio.cfg it >> gives me the following error. >> >> loading modules under >> /usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/ >> ERROR: bad config file (1 errors) >> >> Has anyone any clue why this is happening? I cant find any >> details about thsi error otherwise i would have posted it. >> >> Cheers >> >> Best Regards >> Rizwan Qureshi >> >> >> _______________________________________________ >> 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 > > -- Daniel-Constantin Mierla http://www.asipto.com

1 0

Terminated call processing Status: 487
by JR Richardson 04 Nov '10

04 Nov '10

Hi All, I'm seeing random errors in my Kamailio 3.0.2 log: ERROR: tm [tm.c:1300]: ERROR: w_t_relay_to: t_relay_to failed ERROR: tm [t_fwd.c:1379]: ERROR: t_forward_nonack: no branches for forwarding After going blind staring at wireshark, I can consistently reproduce this error by initiating a call, then after about a second ro two, quickly hanging up. So here is my setup: SIP Phone><Asterisk><Kamailio><SIP Service Provider I see all transaction messages flowing properly through all nodes and I don't get any hung channels, all call legs release properly. I'm guessing I need a special [route] in Kamailio to handle 487 request so they look normal and not issue an error? Kamailio is forwarding the 487 messages ok, so why do I get a t_relay error? What info would be helpful to diagnose the errors? Any insight would be appreciated. Thanks. JR -- JR Richardson Engineering for the Masses

3 2

installing dispatcher module
by Rizwan Qureshi 04 Nov '10

04 Nov '10

Hi List, This is my first email to this list and I am a kamailio newbie. I have worked on asterisk a lot but this is the first time im going to use kamailio. I have successfully installed kamailio on my centos system. I can see all the modules located in the /usr/local/lib/kamailio/modules_k/ AND /usr/local/lib/kamailio/modules/ directories. But the problem is when i try to load the dispatcher module from kamailio.cfg it gives me the following error. loading modules under /usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/ ERROR: bad config file (1 errors) Has anyone any clue why this is happening? I cant find any details about thsi error otherwise i would have posted it. Cheers Best Regards Rizwan Qureshi

2 1

Try to build the SER 2.0 RC1 on Solaris 10 but failed.
by Ivan Yiu 04 Nov '10

04 Nov '10

I try to build the SER using gmake, after resolving some dependency issue, the script can run but finally got a error as follows. yacc -d -b cfg cfg.y conflicts: 1 shift/reduce gmake: *** [cfg.tab.c] Broken Pipe What is the problem?

2 1

How can I remove all RPID headers from SIP message
by Mino Haluz 04 Nov '10

04 Nov '10

Hi, I tried to add RPID headers at the beggining of SIP options on the client side, and I was able to change the caller number simply this way. How can I avoid this ? I'm searching for some function in Kamailio which would delete all RPID headers so that Kamailio could add only one (and correct) RPID header (I'm using append_rpid_hf) Thanks.

2 1

udp_send "Invalid argument"
by Juha Heinanen 04 Nov '10

04 Nov '10

in some of my tests i get the following udp send related errors, when request is t_relayed: Oct 24 11:20:20 test /usr/sbin/sip-proxy[10032]: INFO: Routing initial INVITE <sip:+35892223333@test.fi> to <sip:+35892223333@193.166.3.2;transport=udp> Oct 24 11:20:20 test /usr/sbin/sip-proxy[10032]: ERROR: <core> [udp_server.c:566]: ERROR: udp_send: sendto(sock,0xb2d1f1e8,1190,0,193.166.3.2:5060,16): Invalid argument(22) Oct 24 11:20:20 test /usr/sbin/sip-proxy[10032]: : <core> [udp_server.c:571]: CRITICAL: invalid sendtoparameters one possible reason is the server is bound to localhost and attempts to send to the net Oct 24 11:20:20 test /usr/sbin/sip-proxy[10032]: ERROR: tm [../../forward.h:148]: msg_send: ERROR: udp_send failed Oct 24 11:20:20 test /usr/sbin/sip-proxy[10032]: ERROR: tm [t_fwd.c:1244]: ERROR: t_send_branch: sending request on branch 0 failed $du is set to sip:+35892223333@193.166.3.2;transport=udp and the proxy is listening on many sockets, both 127.0.0.1 and external ip address. any idea what might cause the errors? -- juha

5 8

LCR Question
by Trevor Francis 03 Nov '10

03 Nov '10

I have managed to work my way through the LCR module documentation and added the loadmodule and module parameters. However, I dont know where to add the routing logic that will allow me to add LCR rules and gateways in Siremis and to have Kamailio recognize them. Below is my Kamailio config: Advice? root@u15396663:/var/log# cat /usr/local/etc/kamailio//kamailio.cfg #!KAMAILIO # #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_ACCDB #!define WITH_NAT # # Kamailio (OpenSER) SIP Server v3.1 - default configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: <sr-users(a)lists.sip-router.org> # # Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!define DBURL "mysql://root:f1n4lh0p@localhost/openser" #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ #auto_aliases=no /* add local domain aliases */ #alias="sip.mydomain.com" /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:10.0.0.10:5060 /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 #!ifdef WITH_TLS enable_tls=yes #!endif ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" #!endif ####### Modules Section ######## # set paths to location of modules #!ifdef LOCAL_TEST_RUN mpath="modules_k:modules" #!else mpath="/usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "mi_rpc.so" loadmodule "acc.so" loadmodule "mi_datagram.so" loadmodule "rtimer.so" loadmodule "sqlops.so" loadmodule "cfgutils.so" # ----- LCR ----- loadmodule "lcr.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- 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) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* 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_ouser=$tU;dst_user=$rU;dst_domain=$rd;src_ip=$si") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # use caching modparam("domain", "db_mode", 1) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@finalhop.net<sip%3Apinger(a)finalhop.net> ") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif # ----- Mi_Datagram ----- modparam("mi_datagram", "socket_name", "udp:127.0.0.1:8033") # ----- Charts ----- modparam("rtimer", "timer", "name=tst;interval=300;mode=1;") modparam("rtimer", "exec", "timer=tst;route=STATS") modparam("sqlops","sqlcon", "ca=>mysql://root:f1n4lh0p@localhost/openser") # ----- LCR ----- modparam("lcr","db_url", "mysql://root:f1n4lh0p@localhost/openser") modparam("lcr","lcr_gw_table","lcr_gw") modparam("lcr","id_column", "id") modparam("lcr","lcr_id_column", "lcr_id") modparam("lcr","tag_column","tag") modparam("lcr","weight_column","gw_weight") modparam("lcr","flags_column","flags") modparam("lcr","gw_name_column","gw_name") modparam("lcr","ip_addr_column","ip_addr") modparam("lcr","port_column","port") modparam("lcr","params_column", "params") modparam("lcr","hostname_column","hostname") modparam("lcr","uri_scheme_column","uri_scheme") modparam("lcr","strip_column","strip") modparam("lcr","transport_column","transport") modparam("lcr","tag_column", "tag") modparam("lcr","defunct_column", "defunct") modparam("lcr","lcr_rule_table","lcr_rule") modparam("lcr","prefix_column","prefix") modparam("lcr","from_uri_column","from_uri") modparam("lcr","priority_column","priority") modparam("lcr","stopper_column", "stopper") modparam("lcr","enabled_column", "enabled") modparam("lcr","lcr_rule_target_table", "lcr_rule_target") modparam("lcr", "rule_id_column", "rule_id") modparam("lcr", "gw_id_column", "gw_id") modparam("lcr", "priority_column", "priority") modparam("lcr","weight_column", "weight") modparam("lcr", "lcr_count", 1) modparam("lcr","gw_uri_avp", "$avp(i:709)") modparam("lcr","ruri_user_avp", "$avp(i:500)") modparam("lcr", "flags_avp", "$avp(i:712)") modparam("lcr", "defunct_capability", 0) modparam("lcr", "lcr_id_avp", "$avp(s:lcr_id_avp)") modparam("lcr", "defunct_gw_avp", "$avp(s:defunct_gw_avp)") modparam("lcr", "lcr_rule_hash_size", 128) modparam("lcr", "lcr_gw_count", 128) modparam("lcr", "fetch_rows", 1024) ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route route { # 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")) { 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; } } # 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 $ru\n"); } # Sample onreply route onreply_route[REPLY_ONE] { xdbg("incoming reply\n"); #!ifdef WITH_NAT if ((isflagset(FLT_NATS) || isbflagset(FLB_NATB)) && status=~"(183)|(2[0-9][0-9])") { force_rtp_proxy(); } if (isbflagset("6")) { fix_nated_contact(); } #!endif } # Siremis Charts route[STATS] { sql_query("ca", "insert into statistics (time_stamp,shm_used_size,shm_real_used_size," "shm_max_used_size,shm_free_used_size,ul_users,ul_contacts) values ($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(); ##} } root@u15396663:/var/log#

1 0

Connection tests
by Olle E. Johansson 03 Nov '10

03 Nov '10

Hello friends, Is there any way from the outside (like from MI) I can check the existing TCP/TLS connections? Can I drop them somehow? /O

3 2

buffer overflow
by Maw Zin Tun - Real Telekom 03 Nov '10

03 Nov '10

Hi List, Any idea how to correct this error ? Nov 3 15:01:18 s15391037 /usr/local/kamailio-3.0/sbin/kamailio[27812]: ERROR: <core> [pvapi.c:967]: no more space for text [84] Nov 3 15:01:18 s15391037 /usr/local/kamailio-3.0/sbin/kamailio[27812]: ERROR: <core> [pvapi.c:995]: buffer overflow -- increase the buffer size... Nov 3 15:01:18 s15391037 /usr/local/kamailio-3.0/sbin/kamailio[27812]: ERROR: sqlops [sql_api.c:204]: cannot print the sql query Thanks, Maw Tun Real Telekom

2 1

DoS on Kamailio
by Bernhard Suttner 03 Nov '10

03 Nov '10

Hi, is there a Kamailio module to prevent DoS Attacks like SIP Register Scanners? I know, that there is a way to implement that with Fail2Ban and xlog but it would be nice to have a "pure" Kamailio solution for that? Any idea is appreciated. Best regards, Bernhard Suttner

3 2

Siremis V2.0.0 Released
by Elena-Ramona Modroiu 03 Nov '10

03 Nov '10

Hi, Siremis v2.0.0 is out – the web management interface for Kamailio (Openser) v3.1.0 and SIP Express Router (SER). This is a major release, with countless improvements and new features since v1.x series, among them: * major re-factoring of web interface * better accessibility * simplified menu structure * completely new look * dashboard with the map of all available tools * developed on top of Cubi and PHPOpenBiz v2.4 frameworks * web installation wizard * added new modules: xcap, dialog, new lcr * usage of separate database for siremis itself * management of users that can login to siremis * management of menu can be done from web interface * building Apache conf and htaccess file can be done by Makefie * charts to monitor location transport layers Step by step installation tutorial, screenshots and demo are available on the web at: * Installation Tutorial - http://asipto.com/u/6 * Screenshots - http://asipto.com/u/7 * Web Demo - http://asipto.com/u/8 Siremis is used during Kamailio Advanced Training classes for management of SIP server, a good opportunity to learn about Siremis itself, next locations are: * Kamailio Advanced Training for Carrier Services, November 22-25, 2010, Berlin, Germany (http://asipto.com/u/kat) * Kamailio Advanced Training for Carrier Services, January 24-26, 2010, Irvine, CA, USA (http://asipto.com/u/katu) More details about Siremis project are available at: * http://siremis.asipto.com Regards, Ramona

3 6

Latest Siremis build problem
by Trevor Francis 03 Nov '10

03 Nov '10

Not Found The requested URL /siremis/system/general_default was not found on this server. Apache/2.2.8 (Ubuntu) PHP/5.2.4-2ubuntu5.12 with Suhosin-Patch Server at 74.208.174.183 Port 80 I have tested this on both Firefox and Safari.

2 1

using tls from modules_s instead of core
by Jijo 02 Nov '10

02 Nov '10

Hi all. What changes i have to make in the build to use tls from modules_s instead from core modules. Thanks Jijo

2 1

loop through variables
by Bernhard Suttner 02 Nov '10

02 Nov '10

Hi, I want something like that: # define some router to use later in the script (global section) Mod_param(router(1), "10.10.10.1") mod_param(router(2), "10.10.10.2") mod_param(router(3), "10.10.10.3") Mod_param(router_count, "3") within the different routes I want to loop through the routers like that: $var(i) while($(var(i) < $var(router_count)) { if ($var(router($var(i)) == src_ip") { .. do something } } Is that somehow possible? Any hint or idea is appreciated! Best regards, Bernhard Suttner

5 11

Question about the Q-value.
by Kosilov Fedor 02 Nov '10

02 Nov '10

Hello, everybody! As I understand, the Q-value must be set by a client-side device. We use mainly Linksys SPA2102, I have searched through the web interface of this device but have not found where this value should be set. Is this parameter not supported by simple SOHO devices? Or have I misunderstood something?

3 2

siremis 2.0 and Captcha issues !
by X- x -X 02 Nov '10

02 Nov '10

Hi all, Is anyone else is also having problems in seeing captcha while logging in http://myhost.mydomain/siremis/index.php/user/login<http://192.168.90.125/siremis/index.php/user/login> (during installation, I haven't been able to login since Im not able to see any image. just the image reference thing) I've tried firefox, chome & opera (all with flash plugins installed & of-course simple images also show fine) Is there a way I can disable them. Seem like these 2 files are driving the whole captch show /var/www/siremis-2.0.0/siremis/bin/antispam_image.php /var/www/siremis-2.0.0/siremis/modules/user/form/Antispam.php Are there any distro version specific libraries required. Im trying the whole thing on debian lenny. Cheers -- X

2 2

SMS module not found parameter.
by peter_green lion 02 Nov '10

02 Nov '10

hi all, i am a new bie in kamailio. i use SMS module to send message to GSM net work. my kamailio.cfg is : modparam("sms", "modems", "wavecom[d=/dev/ttyS;c=+84900000022;b=57600; m=new;l=20]") modparam("sms", "networks", "D1[m=10]; d2[m=10]") modparam("sms", "links", "wavecom[D1;d2]") modparam("sms", "default_net", "localhost") modparam("sms", "domain_str", "D1.d2") modparam("sms", "max_sms_parts", 10) modparam("sms", "use_contact", 1) modparam("sms", "sms_report_type", 1) if(method == "MESSAGE"){ if(has_body("text/plain")){ sms_send_msg_to_net("D1"); } } but when i start kamailio the log show : Oct 25 06:28:26 localhost kamailio: ERROR: <core> [modparam.c:143]: set_mod_param_regex: parameter <domain_str> not found in module <sms> Oct 25 06:28:26 localhost kamailio: : <core> [cfg.y:3333]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 173, column 42: Can't set module parameter please help to configure it correct. i use wavecom modem and some infomation: baudrate=57600 c=+84900000022 ip server:192.168.1.100 modem connect with server via com1 (/dev/ttyS0) and kamailio version 3.0.2 thanks for help me. Peter Green

2 3

Dead Charts in Siremis 2.0
by Trevor Francis 02 Nov '10

02 Nov '10

I managed to get the SHM chart working by manually adding the field "random" inside the table "statistics." However, the Charts "Load Charts" and "TM Charts" gave me "Internal System Errors" unless I add the fields tm_active, rcv_req_diff, fwd_req_diff, and 2xx_trans_diff to the statistics table in mysql. I am not sure if these statistics are being updated, because my charts dont appear to be displaying information. Can anyone speak to what is going on here? TGF

2 1

Kamalio 3.1, Siremis 2.0 Problem
by Trevor Francis 02 Nov '10

02 Nov '10

I am running into the following errors: /usr/local/sbin/kamailio[6774]: ERROR: db_mysql [km_dbase.c:120]: driver error on query: Unknown column 'random' in 'field list' /usr/local/sbin/kamailio[6774]: ERROR: <core> [db_query.c:130]: error while submitting query /usr/local/sbin/kamailio[6774]: ERROR: sqlops [sql_api.c:217]: cannot do the query I believe it is preventing my LCR from working properly.

2 1

New Gateway Configuration
by Trevor Francis 01 Nov '10

01 Nov '10

My final question/comment for the day, I promise. I am used to using Siremis 1.x or configuring gateways. We are testing a new box on Kam 3.1 and Siremis 2.0. Previously to add an external LCR gateway, I would plop in the gateway name, group id, and tag to the LCR Gateway section and create an LCR rule with the LCR ID, Group ID, Prefix, and Priority that points to the LCR gateway that I created and I could fly straight. Since there are new attributes in 3.1 and in Siremis, how does the new configuration look? I have successfully called from softphone subscriber to softphone subscriber with no issues, but cannot get the LCR to allow me to dial beyond my internal network. Ideas? TGF

1 0

Kamailio 3.1, Siremis 2.0
by Trevor Francis 01 Nov '10

01 Nov '10

I am running into the following errors: /usr/local/sbin/kamailio[6774]: ERROR: db_mysql [km_dbase.c:120]: driver error on query: Unknown column 'random' in 'field list' /usr/local/sbin/kamailio[6774]: ERROR: <core> [db_query.c:130]: error while submitting query /usr/local/sbin/kamailio[6774]: ERROR: sqlops [sql_api.c:217]: cannot do the query I believe it is preventing my LCR from working properly. -- Trevor G. Francis Managing Member trevor.francis(a)tgrahamcapital.com Ph. +1 888.616.0662 Fx. +1 214.722.1318 P.O Box 54771 Oklahoma City, OK 73154 MSN: trevor.francis(a)fiberhaus.com Personal emails should be addressed to: tfrancis(a)fas.harvard.edu --

1 0

Setting up TLS between proxy and authentication server
by Couprie Geoffroy 01 Nov '10

01 Nov '10

Hello, I am testing TLS communication with Kamailio 3.0.2, and I encounter a strange problem. My setup is like this: Client <-UDP-> Proxy server <- TLS with client certificate authentication -> Authentication server 192.168.24.1 192.168.24.128 192.168.24.129 The two servers are instance of Kamailio 3.0.2 When the client sends a REGISTER, the proxy retransmits the message to the authentication server, which sends back a 401 Unauthorized. But it seems the proxy closes the TLS connexion right after forwarding the REGISTER, and doesn't receive the 401 message. The TLS handshake is OK, and the client certificate is required (I didn't add the verification part yet). The REGISTER message goes through TLS, and is received by the authentication server. Then, the proxy sends a TLS alert (Close-notify), and after that message, the authentication server sends back the 401, and the proxy ignores that message. Could it be caused by a timeout? Is there a way to keep the TLS connection opened? Here are the relevant files (I don't like to send mails of more than 100 lines): authentication server configuration: http://pastebin.com/QBmnNc4e authentication server log: http://pastebin.com/uYdHDG5G proxy server configuration: http://pastebin.com/8WPPJBtM proxy server log: http://pastebin.com/JTwJSKtk I am just testing TLS, so I have tried to remove most of the irrelevant parts. Best regards, Geoffroy ________________________________ Ce message et les pi?ces jointes sont confidentiels et r?serv?s ? l'usage exclusif de ses destinataires. Il peut ?galement ?tre prot?g? par le secret professionnel. Si vous recevez ce message par erreur, merci d'en avertir imm?diatement l'exp?diteur et de le d?truire. L'int?grit? du message ne pouvant ?tre assur?e sur Internet, la responsabilit? du groupe Atos Origin ne pourra ?tre recherch?e quant au contenu de ce message. Bien que les meilleurs efforts soient faits pour maintenir cette transmission exempte de tout virus, l'exp?diteur ne donne aucune garantie ? cet ?gard et sa responsabilit? ne saurait ?tre recherch?e pour tout dommage r?sultant d'un virus transmis. This e-mail and the documents attached are confidential and intended solely for the addressee; it may also be privileged. If you receive this e-mail in error, please notify the sender immediately and destroy it. As its integrity cannot be secured on the Internet, the Atos Origin group liability cannot be triggered for the message content. Although the sender endeavours to maintain a computer virus-free network, the sender does not warrant that this transmission is virus-free and will not be liable for any damages resulting from any virus transmitted.

3 5

drop in branch route
by Juha Heinanen 31 Oct '10

31 Oct '10

based on core cookbokk, it is not clear to me what happens when drop is executed in branch route: drop Stop the execution of the configuration script and alter the implicit action which is done afterwards. If the function is called in a 'branch_route' then the branch is discarded (implicit action for 'branch_route' is to forward the request). i understand that request is not forwarded to that branch, but if that branch is the only branch, does t_relay() then return -1 or is script execution terminated? -- juha

1 0

setdebug() and 3.X
by Sergey Okhapkin 30 Oct '10

30 Oct '10

setdebug() function is no longer provided by core. How to change debug level dynamically in the script to trace execution of a few statements only?

3 11

How to log local SIP replies?
by Sergey Okhapkin 30 Oct '10

30 Oct '10

Is there a way to log local replies sent by kamailio? For example, I want to log a reply sent by kamailio in response to REGISTER request.

2 1

Check out Elias's photos on Facebook
by Elias Gitonga 30 Oct '10

30 Oct '10

Hi Serusers, I set up a Facebook profile where I can post my pictures, videos and events and I want to add you as a friend so you can see it. First, you need to join Facebook! Once you join, you can also create your own profile. Thanks, Elias To sign up for Facebook, follow the link below: http://www.facebook.com/p.php?i=100001050585836&k=Z6E3Y6W5U2VBWABJPB63QUTTV… Already have an account? Add this email address to your account: http://www.facebook.com/n/?merge_accounts.php&e=serusers%40iptel.org&c=2814… ======================================= serusers(a)iptel.org was invited to join Facebook by Elias Gitonga. If you do not wish to receive this type of email from Facebook in the future, please follow the link below to unsubscribe. http://www.facebook.com/o.php?k=679489&u=100001575534555&mid=335e9d2G5af36e… Facebook, Inc. P.O. Box 10005, Palo Alto, CA 94303

1 0

Re: [SR-Users] setdebug() and 3.X
by Daniel-Constantin Mierla 30 Oct '10

30 Oct '10

On 10/29/10 5:40 PM, Sergey Okhapkin wrote: > I can't test now - high volume traffic is running. But I looked at the diff, it > seems to me OK, I see no reason why it could not work:-) Because of the law, the Murphy law ;-) Probably I will test sometime next week if you don't do it before and then backport. Cheers, Daniel > On Friday 29 October 2010, you wrote: >> On 10/29/10 4:33 PM, Daniel-Constantin Mierla wrote: >>> On 10/29/10 4:09 PM, Sergey Okhapkin wrote: >>>> I just realized that 3.1 executes main routing block when broken SIP >>>> message >>>> is received! But sanity_check() function returns 0 on failed checks >>>> and script >>>> execution stops, my favorite xlog() is not executed. >>> OK, that can be fixed very easy. Iirc, in some cases you don't get the >>> sip message in the config if it is really broken, but I am not sure if >>> error_route was executed for that case anyhow. >> I committed on git master, id: >> >> 39a8108c62a2beafba18417613b9660a8abef86a >> >> >> Can you take the patch and test it on 3.1? If all is fine, then I can >> backport - even the default cfg expects non-drop behavior since it >> attempts to print a log message there. >> >> Cheers, >> Daniel >> >>> Cheers, >>> Daniel >>> >>>> Oct 29 09:48:54 west /usr/local/sbin/kamailio[12225]: ERROR:<core> >>>> [parser/msg_parser.c:177]: ERROR: get_hdr_field: bad to header >>>> Oct 29 09:48:54 west /usr/local/sbin/kamailio[12225]: INFO:<core> >>>> [parser/msg_parser.c:351]: ERROR: bad header field [To: Chi Nhung<sip: >>>> ] >>>> Oct 29 09:48:54 west /usr/local/sbin/kamailio[12225]: ERROR:<core> >>>> [parser/msg_parser.c:177]: ERROR: get_hdr_field: bad to header >>>> Oct 29 09:48:54 west /usr/local/sbin/kamailio[12225]: INFO:<core> >>>> [parser/msg_parser.c:351]: ERROR: bad header field [To: Chi Nhung<sip: >>>> ] >>>> Oct 29 09:48:54 west /usr/local/sbin/kamailio[12225]: ERROR:<core> >>>> [parser/msg_parser.c:177]: ERROR: get_hdr_field: bad to header >>>> Oct 29 09:48:54 west /usr/local/sbin/kamailio[12225]: INFO:<core> >>>> [parser/msg_parser.c:351]: ERROR: bad header field [To: Chi Nhung<sip: >>>> ] >>>> Oct 29 09:48:54 west /usr/local/sbin/kamailio[12225]: ERROR:<core> >>>> [msg_translator.c:1946]: ERROR: build_res_buf_from_sip_req: alas, parse >>>> _headers failed >>>> Oct 29 09:48:54 west /usr/local/sbin/kamailio[12225]: WARNING: sanity >>>> [sanity.c:253]: sanity_check(): check_required_headers(): failed to s >>>> end 400 via sl reply >>>> >>>> On Friday 29 October 2010, Daniel-Constantin Mierla wrote: >>>>> On 10/29/10 11:53 AM, Sergey Okhapkin wrote: >>>>>> Daniel, I think this could be implemented without re-introduction of >>>>>> error_route, >>>>> it will not be re-introduced for sure, we have now a different core >>>>> framework which care reuse event_route mechanism. >>>>> >>>>> In the past the error route was executed when the error was discovered, >>>>> so it several places of the code, and I am not sure that the core can >>>>> continue up to config execution of something is really broken. >>>>> >>>>> I will have it in mind when I get some spare time. >>>>> >>>>> Cheers, >>>>> Daniel >>>>> >>>>>> what about core parameter route_on_error=yes/no (default no)? If >>>>>> route on error is enabled, then sanity module will do the trick: >>>>>> >>>>>> route{ >>>>>> if(!sanity_check("1159")) { >>>>>> xlog("L_INFO","Bad message from >>>>>> $proto:$si:$sp\n$mb\n"); >>>>>> break; >>>>>> } >>>>>> .... >>>>>> >>>>>> It will be up to script writer what to do in main routing block if >>>>>> message parsing fails. >>>>>> >>>>>> On Friday 29 October 2010, Sergey Okhapkin wrote: >>>>>>> It would be nice to get error_route back. The route was trivial in my >>>>>>> cfg file: >>>>>>> >>>>>>> error_route { >>>>>>> xlog("L_INFO","Bad message from $proto:$si:$sp\n$mb\n"); >>>>>>> drop(); >>>>>>> } >>>>>>> >>>>>>> On Friday 29 October 2010, Daniel-Constantin Mierla wrote: >>>>>>>> On 10/28/10 1:41 PM, Sergey Okhapkin wrote: >>>>>>>>> I've found one more feature missed in 3.1 - error_route is >>>>>>>>> eliminated >>>>>>>>> and sanity module added. But if received SIP message is >>>>>>>>> malformed and >>>>>>>>> can't be parsed, routing script is not executed and I have no >>>>>>>>> way to >>>>>>>>> log the message to kamailio log. I did it in error_route before. >>>>>>>>> >>>>>>>>> This logging is important to me to support customers. SIP ALG in >>>>>>>>> routers often breaks Via line :-( >>>>>>>> Since I developed error_route, I remember that it handled just >>>>>>>> sporadic >>>>>>>> cases, but I can add an event_route to pass the buffer and some >>>>>>>> hints >>>>>>>> to configuration file in case the initial parsing fails and the >>>>>>>> message >>>>>>>> is dropped before getting to config file. >>>>>>>> >>>>>>>> It is good to know that someone use such case, introduction of >>>>>>>> sanity >>>>>>>> was clearly mentioned and so far there was no complain. >>>>>>>> >>>>>>>> Cheers, >>>>>>>> Daniel >>>>>>>> >>>>>>>>> On Thursday 28 October 2010, Daniel-Constantin Mierla wrote: >>>>>>>>>> On 10/26/10 2:22 PM, Sergey Okhapkin wrote: >>>>>>>>>>> setdebug() function is no longer provided by core. How to change >>>>>>>>>>> debug level dynamically in the script to trace execution of a few >>>>>>>>>>> statements only? >>>>>>>>>> seems was lost in transition to 3.0, I will check to see what >>>>>>>>>> can be >>>>>>>>>> done with the new architecture. >>>>>>>>>> >>>>>>>>>> Thanks, >>>>>>>>>> Daniel >>>>>>>>> _______________________________________________ >>>>>>>>> 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 >>>>>> _______________________________________________ >>>>>> 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 -- Daniel-Constantin Mierla http://www.asipto.com

2 1

NAT ping replies
by Sergey Okhapkin 29 Oct '10

29 Oct '10

Kamailio 3.1 log is filled with lines like Oct 29 03:32:46 west /usr/local/sbin/kamailio[632]: INFO: <script>: incoming reply from udp:188.62.4.186:65333 SIP/2.0 404 Not Found Via: SIP/2.0/UDP 204.74.213.5:5060;branch=0 From: <sip:pinger@sip.callwithus.com>;tag=3993f0c To: <sip:188.62.4.186:65333> Call-ID: aff32da5-89ef0136-71dc8c5(a)204.74.213.5 CSeq: 1 OPTIONS Content-Length: 0 Oct 29 03:32:46 west /usr/local/sbin/kamailio[632]: INFO: <core> [forward.c:774]: broken reply to forward - no 2nd via Shouldn't replies to NAT ping messages be silently dropped like in 1.4?

2 2

Billing party and radius accounting
by Efelin Novak 29 Oct '10

29 Oct '10

Hi folks, I have a problem with setting billing party in my radius messages. My accounting is done by CDRTool via radius messages to radius server. The thing I don`t know is how to set billing party variable (by default $avp(s:billing_party) with reliable value. I can try adding Diversion header and fill the variable with it, but can I trust the Diversion header? When UAC sets redirect(302) and Invite comes, he replies 302 and Kamailio passes this message to origin. However is it possible to change Diversion header by UAC? This could be way how to make free calls and bill it to someone else. I was thinking of uac_redirect module. If I get it right this module saves original number in memory and does not use message itself to store it. In documentation there is written that accounting can by done by setting modparam("uac_redirect", "acc_function", "acc_rad_request"). Even when I use exported function set_accept_filter() the accounting itself isn't done. I`d like to get original number from module uac_redirect and set it to $avp(s:billing_party). This would solve the problem of billing forwarded calls. Am I getting it right? Is there anything else what can be passed to billing party variable to make correct bililng? I'm using kamailio 3.1.x and CDRTool v 8.0.6 Thanks, Efelin

4 12

tm.t_uac_start examples from CLI
by Alex Balashov 29 Oct '10

29 Oct '10

Can someone post an example of usage of tm.t_uac_start from the command line using sercmd over a UNIX domain socket (e.g. /tmp/ser_ctl)? I am quite unclear on how to pass minimally necessary header values to it in this form and cannot seem to get the syntax right. Thanks! -- Alex Balashov - Principal Evariste Systems LLC 1170 Peachtree Street 12th Floor, Suite 1200 Atlanta, GA 30309 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

1 0

problem getting siriemis to login
by Gerry Kernan 29 Oct '10

29 Oct '10

Hi I am trying to get siremis working with kamailio, butwhen i open the login page and enter the username and password nothing happens. I have followed the instruction on the siremis home page. Created mysql db called openser and inserted username admin& password admin. mysql> use openser; Reading table information for completion of table and columnnames You can turn off this feature to get a quicker startup with-A Database changed mysql> select * from ob_users; +-------+--------+----------+ | SYSID | USERID | PASSWORD | +-------+--------+----------+ | 1 | admin |admin | +-------+--------+----------+ 1 row in set (0.00 sec) mysql> edited /var/www/html/siremis-1.0.1/siremis/metadata/Config.xml <?xml version="1.0"standalone="no"?> <Config> <DataSource> <DatabaseName="Default" Driver="Pdo_Mysql"Server="localhost" DBName="openser" User="admim"Password="admin" /> <DatabaseName="Seremis" Driver="Pdo_Mysql"Server="localhost" DBName="openser" User="admin"Password="admin" /> </DataSource> </Config> Ifsomeone could point out to me where i have gone wrong i would really appreciateit. Regards, Gerry Kernan InfinityIT Suite 17 The Mall, Beacon court, Sandyford, Dublin 18. p:+353-1-2930090 f:+353-1-2930137

1 0

Kamailio Business Directory
by Daniel-Constantin Mierla 28 Oct '10

28 Oct '10

Hello, I am please to announce the launch of Kamailio Business Directory. This is a web page hosted by the project site that will list companies and individuals offering products, services or solutions based on Kamailio or SER. It is an open directory that tries to enable more visibility to the business market around the project and help finding quickly the best service for specific market demands. We invite everyone doing such business to apply to be listed, please follow the instructions at: http://www.kamailio.org/w/business-directory-rules/ We have already several companies listed, couple of them to still in review process, therefore check it again soon, the directory is available at: http://www.kamailio.org/w/business-directory/ Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

About "reply_code" in 'carrierroute' module
by Iñaki Baz Castillo 28 Oct '10

28 Oct '10

Hi, how is the syntax of the "reply_code" column in 'carrierroute' module? In the doc I some examples: 408 404 50. Obviously "50." means any code starting by "50", but is it a typical regular expression? For example, does it allow something like "500|503" or "50[03]"? Thanks. -- Iñaki Baz Castillo <ibc(a)aliax.net>

2 4

Asterisk, Kamailio, Openfire and Social Media Integration
by Daniel-Constantin Mierla 28 Oct '10

28 Oct '10

Hello, courtesy of Kelvin Chua, I received the slides he presented yesterday at Astricon 2010 to the session "Asterisk, Kamailio, Openfire and Social Media Integration". More details at: http://www.kamailio.org/w/2010/10/asterisk-kamailio-openfire-and-social-med… Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

PSTN call
by michel freiha 28 Oct '10

28 Oct '10

Dear All, Can someone help me to connect my kamilio in order to make a PSTN call by rewriting host tp PSTN gateway? I changed my config in a manner to do that...The line is ringing but as soon as I open the line on other side the call will hangup...Please find the piece of code # RTPProxy control route[RTPPROXY] { #!ifdef WITH_NAT if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ rewritehost("XX.XX.XX.XX"); force_rtp_proxy(); } if (!has_totag()) add_rr_param(";nat=yes"); #!endif return; } Regards

2 1

SIP application type
by Mino Haluz 28 Oct '10

28 Oct '10

Hi, I'm using radius server to do an accounting, but it does not store an application type (audo/video). Audio/video calls has to be rated with cdrtool, that is why I need this information. It is not even sent through radius protocol, so the radius server cannot get it. My extra-params are: modparam("acc", "radius_extra", "User-Name=$Au; \ Calling-Station-Id=$from; \ Called-Station-Id=$to; \ Sip-Translated-Request-URI=$avp(s:translated_uri); \ Sip-RPid=$avp(s:rpid); \ Source-IP=$si; \ Source-Port=$sp; \ Canonical-URI=$avp(s:can_uri); \ Billing-Party=$avp(s:billing_party); \ Divert-Reason=$avp(s:divert_reason); \ X-RTP-Stat=$hdr(X-RTP-Stat); \ Contact=$hdr(contact); \ Event=$hdr(event); \ SIP-Proxy-IP=$avp(s:sip_proxy_ip); \ ENUM-TLD=$avp(s:enum_tld)") Suggestions? Thanks.

4 4

HELP
by michel freiha 28 Oct '10

28 Oct '10

Hello All, I have my kamailio server running and working properly as much as it's listening to an IP address..>As soon as i add an alias as a domain mae, I'm not be able to make call anymore and i'm getting Proxy Authentication required on my X-lite registered on the server...Please find below SIP packets: U +0.045706 Kamailio_IP:5060 -> X-Lite_Public_IP:49088 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 192.168.0.164:49088 ;branch=z9hG4bK-d8754z-a83b4359a1218b33-1---d8754z-;rport=49088;received=X-Lite_Public_IP. To: "00XXXXXX"<sip:00XXXXXX@Aliasdomain.com >;tag=7adebec44d02c48c117ad920bb5281f8.c246. From: <sip:michofr@Aliasdomain.com>;tag=00727922. Call-ID: MDc5YzVkNjdjYTZkMTIwMTdlYzE2YTRlZmExMDllNWM.. CSeq: 4 INVITE. Proxy-Authenticate: Digest realm="Aliasdomain.com", nonce="TMgYGEzIFuyzAuVUGSbPwq3ZJaBE8k4a". Server: kamailio (3.1.0 (i386/linux)). Content-Length: 0. . U +0.253584 X-Lite_Public_IP:49088 -> Kamailio_IP:5060 INVITE sip:00XXXXXX@Aliasdomain.com SIP/2.0. Via: SIP/2.0/UDP 192.168.0.164:49088 ;branch=z9hG4bK-d8754z-a83b4359a1218b33-1---d8754z-;rport. Max-Forwards: 70. Contact: <sip:michofr@X-Lite_Public_IP:49088>. To: "00XXXXXX"<sip:00XXXXXX@Aliasdomain.com>. From: <sip:michofr@Aliasdomain.com>;tag=00727922. Call-ID: MDc5YzVkNjdjYTZkMTIwMTdlYzE2YTRlZmExMDllNWM.. CSeq: 4 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Proxy-Authorization: Digest username="michofr",realm="asterisk",nonce="2fe0dd55",uri="sip:00XXXXXX@Aliasdomain.com ",response="a2f6f58360e4a3059a82964f9e8ddbe2",algorithm=MD5. Proxy-Authorization: Digest username="michofr",realm="Aliasdomain.com",nonce="TMgYF0zIFutYZomwP0F6LJLcvg4y/gwt",uri="sip:00XXXXXX@Aliasdomain.com ",response="143021428e9b74db816f76a97b522a60",algorithm=MD5. User-Agent: eyeBeam release 1102q stamp 51814. Content-Length: 283. . v=0. o=- 6 2 IN IP4 192.168.0.164. s=CounterPath eyeBeam 1.5. c=IN IP4 192.168.0.164. t=0 0. m=audio 17956 RTP/AVP 18 101. a=alt:1 1 : 0V1r/5Dg TGVW9xcA 192.168.0.164 17956. a=fmtp:18 annexb=yes. a=fmtp:101 0-15. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000. a=sendrecv. U +0.057365 Kamailio_IP:5060 -> X-Lite_Public_IP:49088 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 192.168.0.164:49088 ;branch=z9hG4bK-d8754z-a83b4359a1218b33-1---d8754z-;rport=49088;received=X-Lite_Public_IP. To: "00XXXXXX"<sip:00XXXXXX@Aliasdomain.com >;tag=7adebec44d02c48c117ad920bb5281f8.c246. From: <sip:michofr@Aliasdomain.com>;tag=00727922. Call-ID: MDc5YzVkNjdjYTZkMTIwMTdlYzE2YTRlZmExMDllNWM.. CSeq: 4 INVITE. Proxy-Authenticate: Digest realm="Aliasdomain.com", nonce="TMgYGEzIFuyzAuVUGSbPwq3ZJaBE8k4a". Server: kamailio (3.1.0 (i386/linux)). Content-Length: 0. . U +0.255609 X-Lite_Public_IP:49088 -> Kamailio_IP:5060 ACK sip:00XXXXXX@Aliasdomain.com SIP/2.0. Via: SIP/2.0/UDP 192.168.0.164:49088 ;branch=z9hG4bK-d8754z-4312233712260f32-1---d8754z-;rport. To: "00XXXXXX"<sip:00XXXXXX@Aliasdomain.com >;tag=7adebec44d02c48c117ad920bb5281f8.c9fd. From: <sip:michofr@Aliasdomain.com>;tag=00727922. Call-ID: MDc5YzVkNjdjYTZkMTIwMTdlYzE2YTRlZmExMDllNWM.. CSeq: 3 ACK. Content-Length: 0. . U +0.015090 X-Lite_Public_IP:49088 -> Kamailio_IP:5060 ACK sip:00XXXXXX@Aliasdomain.com SIP/2.0. Via: SIP/2.0/UDP 192.168.0.164:49088 ;branch=z9hG4bK-d8754z-a83b4359a1218b33-1---d8754z-;rport. To: "00XXXXXX"<sip:00XXXXXX@Aliasdomain.com >;tag=7adebec44d02c48c117ad920bb5281f8.c246. From: <sip:michofr@Aliasdomain.com>;tag=00727922. Call-ID: MDc5YzVkNjdjYTZkMTIwMTdlYzE2YTRlZmExMDllNWM.. CSeq: 4 ACK. Content-Length: 0. . U +0.136372 X-Lite_Public_IP:49088 -> Kamailio_IP:5060 ACK sip:00XXXXXX@Aliasdomain.com SIP/2.0. Via: SIP/2.0/UDP 192.168.0.164:49088 ;branch=z9hG4bK-d8754z-a83b4359a1218b33-1---d8754z-;rport. To: "00XXXXXX"<sip:00XXXXXX@Aliasdomain.com >;tag=7adebec44d02c48c117ad920bb5281f8.c246. From: <sip:michofr@Aliasdomain.com>;tag=00727922. Call-ID: MDc5YzVkNjdjYTZkMTIwMTdlYzE2YTRlZmExMDllNWM.. CSeq: 4 ACK. Content-Length: 0.

2 1

Re: [SR-Users] libosip2 parse body error
by Daniel-Constantin Mierla 27 Oct '10

27 Oct '10

Never ever send private emails - they will be discarded, keep the mailing list cc-ed. Iptel org is running SIP Express Router 3.1. Take a ngrep of the traffic and send it here, the hexa codes you send are not for humans to read, like we are here on mailing lists. Otherwise make a tar.gz of pcap file. SER changes content length only if it is an update to sdp (for nat traversal), otherwise is the one set by the other phone. Daniel On 10/27/10 12:21 PM, t wrote: > Thanks Daniel, > > fyi, the Content-Length doesn't match the actual length in the invite > ok returned from iptel.org <http://iptel.org> (213.192.59.75), and > the same situation also occurs from ekiga.net <http://ekiga.net>. > > libosip2 interprets the rfc as saying that this 'should' be correct. > (and I buy this!). > > it looks like an openser (opensips?) quirk. > > anyway, I have put in code locally to fixup the Content-Length. > > Thanks, > Tony. > > 73 75 62 0d 0a 43 6f 6e 74 65 6e 74 2d 4c 65 6e sub..Content-Len > 67 74 68 3a 20 20 20 32 30 31 0d 0a 0d 0a 76 3d gth: 201....v= > 30 0d 0a 6f 3d 74 77 69 6e 6b 6c 65 20 31 33 33 0..o=twinkle 133 > 33 37 30 37 32 38 39 20 38 30 39 37 35 37 36 37 3707289 80975767 > 33 20 49 4e 20 49 50 34 20 31 39 32 2e 31 36 38 3 IN IP4 192.168 > 2e 32 35 34 2e 31 0d 0a 73 3d 2d 0d 0a 63 3d 49 .254.1..s=-..c=I > 4e 20 49 50 34 20 31 39 32 2e 31 36 38 2e 32 35 N IP4 192.168.25 > 34 2e 31 0d 0a 74 3d 30 20 30 0d 0a 6d 3d 61 75 4.1..t=0 0..m=au > 64 69 6f 20 35 30 32 32 20 52 54 50 2f 41 56 50 dio 5022 RTP/AVP > 20 39 38 20 31 30 31 0d 0a 61 3d 72 74 70 6d 61 98 101..a=rtpma > 70 3a 39 38 20 73 70 65 65 78 2f 31 36 30 30 30 p:98 speex/16000 > 0d 0a 61 3d 72 74 70 6d 61 70 3a 31 30 31 20 74 ..a=rtpmap:101 t > 65 6c 65 70 68 6f 6e 65 2d 65 76 65 6e 74 2f 38 elephone-event/8 > 30 30 30 0d 0a 61 3d 66 6d 74 70 3a 31 30 31 20 000..a=fmtp:101 > 30 2d 31 35 0d 0a 0-15.. > > On Wed, Oct 27, 2010 at 5:40 AM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > Hello, > > you claim that contact header is malformed or libosip2 has issues > in parsing a good content header? > > In case of first option, can you tell what it believes is wrong. > Your message is hard to follow since is not clear text, next time > send only the text part. > > For the second option, you have to approach libosip2 support forums. > > Cheers, > Daniel > > > On 10/25/10 10:44 PM, t wrote: >> Hi, >> >> Not sure how to provide this info (or even if anybody cares), but >> libosip2 parser objects to incorrect (short) Content-Length from >> iptel.org <http://iptel.org>. >> >> Regards, >> Tony. >> >> >> _______________________________________________ >> 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 > > -- Daniel-Constantin Mierla http://www.asipto.com

5 7

libosip2 parse body error
by t 27 Oct '10

27 Oct '10

Hi, Not sure how to provide this info (or even if anybody cares), but libosip2 parser objects to incorrect (short) Content-Length from iptel.org. Regards, Tony.

2 1

SMS module big error ????????????
by peter_green lion 27 Oct '10

27 Oct '10

hi all, i have configure SMS module and have some error. please help to fix it. my configure in kamailio : modparam("sms", "modems", "nokia[d=/dev/ttyUSB0;p=0000;m=old;s=1;l=10;r=20]") modparam("sms", "networks", "MOBIFONE[m=10]") modparam("sms", "links", "nokia[MOBIFONE]") if(method == "MESSAGE"){ if(has_body("text/plain")){ sms_send_msg_to_net("MOBIFONE"); } } but i start kamailio, log show that: Oct 26 05:32:35 localhost /usr/local/sbin/kamailio[13594]: ERROR: sms [libsms_modem.c:282]: Modem is not registered to the network Oct 26 05:32:35 localhost /usr/local/sbin/kamailio[13594]: WARNING: sms [libsms_getsms.c:256]: something happend with the modem -> was reinit -> let's retry Oct 26 05:32:36 localhost /usr/local/sbin/kamailio[13594]: WARNING: sms [libsms_modem.c:356]: modem wants the PIN again! Oct 26 05:32:36 localhost /usr/local/sbin/kamailio[13594]: WARNING: sms [libsms_modem.c:370]: re -init the modem!! Oct 26 05:32:36 localhost /usr/local/sbin/kamailio[13594]: INFO: sms [libsms_modem.c:221]: init modem nokia on /dev/ttyUSB0. Oct 26 05:32:37 localhost /usr/local/sbin/kamailio[13594]: INFO: sms [libsms_modem.c:250]: Checking if Modem is registered to the network Oct 26 05:32:38 localhost /usr/local/sbin/kamailio[13594]: NOTICE: sms [libsms_modem.c:275]: Waiting 2 sec. before retrying Oct 26 05:33:09 localhost last message repeated 10 times Oct 26 05:33:38 localhost last message repeated 9 times Oct 26 05:33:40 localhost /usr/local/sbin/kamailio[13594]: ERROR: sms [libsms_modem.c:282]: Modem is not registered to the network Oct 26 05:33:40 localhost /usr/local/sbin/kamailio[13594]: WARNING: sms [libsms_getsms.c:256]: something happend with the modem -> was reinit -> let's retry Oct 26 05:33:41 localhost /usr/local/sbin/kamailio[13594]: WARNING: sms [libsms_modem.c:356]: modem wants the PIN again! Oct 26 05:33:41 localhost /usr/local/sbin/kamailio[13594]: WARNING: sms [libsms_modem.c:370]: re -init the modem!! Oct 26 05:33:41 localhost /usr/local/sbin/kamailio[13594]: INFO: sms [libsms_modem.c:221]: init modem nokia on /dev/ttyUSB0. Oct 26 05:33:41 localhost /usr/local/sbin/kamailio[13594]: INFO: sms [libsms_modem.c:250]: Checking if Modem is registered to the network Oct 26 05:33:42 localhost /usr/local/sbin/kamailio[13594]: NOTICE: sms [libsms_modem.c:275]: Waiting 2 sec. before retrying so i cannot send ok receive sms. with contact :+84...@sipdomain my nokia phone :6020 i can send sms via nokia by gnokii. please help or suggest to get it wok. thanks alot.

2 1

Dialplan module sample
by Uriel Rozenbaum 27 Oct '10

27 Oct '10

Hey guys, Does someone have an example for rules used with this module? I understand how to use the module, but not how to fill the database. I intend to use some prefix matching and regex (if available). Thanks in advance! Uriel

2 1

Working with Variables.
by dotnetdub 27 Oct '10

27 Oct '10

Hi List, Where is the best place to lookup documentation about working with variables. I come from the if len(x$)=y then xxx xxx xxx End If school! Looking for the best place to lookup documentation for checking characters, lengths of strings and removing or prepending stuff based on the results. Thanks Brian

2 2

Sanity module documentation error?
by Sergey Okhapkin 26 Oct '10

26 Oct '10

The routing script has if(sanity_check("1159", "6")) { xlog("L_INFO","Bad message from $proto:$si:$sp\n$mb\n"); break; } Seems like sanity_check function returns true is sanity check passes, but not failed. Am I right? 0(14085) DEBUG: <core> [receive.c:145]: After parse_msg... 0(14085) DEBUG: <core> [receive.c:186]: preparing to run routing scripts... 0(14085) DEBUG: <core> [parser/parse_to.c:803]: end of header reached, state=10 0(14085) DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [35]; uri=[sip:167147511@204.74.213.5:5090] 0(14085) DEBUG: <core> [parser/msg_parser.c:187]: DEBUG: to body [<sip:167147511@204.74.213.5:5090> ] 0(14085) DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <73> <REGISTER> 0(14085) DEBUG: <core> [parser/msg_parser.c:199]: DEBUG: get_hdr_body : content_length=0 0(14085) DEBUG: <core> [parser/msg_parser.c:101]: found end of header 0(14085) DEBUG: <core> [parser/parse_to.c:174]: DEBUG: add_param: tag=jzbyj 0(14085) DEBUG: <core> [parser/parse_to.c:803]: end of header reached, state=29 0(14085) DEBUG: sanity [mod_sanity.c:217]: all sanity checks passed 0(14085) INFO: <script>: Bad message from udp:71.59.87.156:5066

3 7

Psuedo Variable.
by David J. 26 Oct '10

26 Oct '10

Hello List; I was following along the Kamailio Asterisk Realtime Integration; I modified the script to get it working on Kamailio v 3.1. One area I had a question about was here... We have the following block; # Forward REGISTER to Asterisk route[REGFWD] { if(!is_method("REGISTER")) { return; } $var(rip) = $sel(cfg_get.asterisk.bindip); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.asterisk.bindport); $uac_req(furi)="sip:" + $au + "@" + $var(rip); $uac_req(turi)="sip:" + $au + "@" + $var(rip); $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $ad + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n"; if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n"; uac_req_send(); } If I change the line to read : $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $ad from : $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $sel(cfg_get.kamailio.bindip) I get the following error; loading modules under /usr/local/kamailio-3.1-proxy/lib/kamailio/modules_k/:/usr/local/kamailio-3.1-proxy/lib/kamailio/modules/ $ad 0(13715) ERROR: <core> [pvapi.c:445]: bad parameters 0(13715) : <core> [cfg.y:3409]: parse error in config file /usr/local/kamailio-3.1-proxy/etc/kamailio/kamailio.cfg, line 717, column 48-47: unknown script pseudo variable ERROR: bad config file (1 errors) I would imagine that if the "$au" pseudo variable is available the "$ad" variable would be available as well. Please help point out my mistake. Thanks. David.

6 10

Re: [SR-Users] Drouting usage
by Javier Gallart 26 Oct '10

26 Oct '10

Thanks Marius, that was it. Regards Javier On Mon, Oct 25, 2010 at 10:08 PM, <sr-users-request(a)lists.sip-router.org>wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > > You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. Drouting usage (Javier Gallart) > 2. PSTN call (michel freiha) > 3. Re: Drouting usage (marius zbihlei) > 4. bflag under failure_route (kamailio 1.5) (I?aki Baz Castillo) > 5. Re: bflag under failure_route (kamailio 1.5) > (Daniel-Constantin Mierla) > 6. Re: bflag under failure_route (kamailio 1.5) (I?aki Baz Castillo) > 7. Re: bflag under failure_route (kamailio 1.5) > (Daniel-Constantin Mierla) > 8. Re: bflag under failure_route (kamailio 1.5) (I?aki Baz Castillo) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Mon, 25 Oct 2010 15:16:56 +0200 > From: Javier Gallart <jgallartm(a)gmail.com> > Subject: [SR-Users] Drouting usage > To: sr-users(a)lists.sip-router.org > Message-ID: > <AANLkTik8pWtu9uT1=mPb_zSnztej5kHzPrEVPubgzeY7(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Hello > > sorry if this sounds too "newbie". I'm interested in the drouting module. > The first INVITE is correctly forwarded to the gw foudn in the drouting > table: > if (is_method("INVITE") && !has_totag()) { > do_routing("0"); > record_route(); > } > However for subsequent messages (ACK, BYE) in the dialog I don't know how > to > keep track of the ip resolved by the drouting logic so the messages are > correctly forwarded to the same gw. I've made some tests using the dialog > module with no success. Any hint? > > Thanks in advance > > Javier >

1 0

ERROR
by michel freiha 26 Oct '10

26 Oct '10

Guys I'm getting the below error...Can anybody help please? Oct 22 12:27:17 localhost /usr/local/sbin/kamailio[28566]: ERROR: <script>: time_dbg:method=INVITE;time=1287750437;callid=Y2FjNDAyZDY4YTk5ZWQ1Yjc4MDk4ZGZiZjk5YjIyYmE. - in NAT out route Regards

2 1

Outbound load balancing
by Lee Archer 26 Oct '10

26 Oct '10

Hi, does Kamailio support any sort of outbound load balancing where you can set the resources available at a gateway and the system keeps a list of what's in use? Opensips has something similar but does kamailio? Thanks Lee

4 4

Re: [SR-Users] Redirecting 302 problem
by Iñaki Baz Castillo 25 Oct '10

25 Oct '10

(I forward and reply this mail to the maillist). 2010/10/26 Mino Haluz <mino.haluz(a)gmail.com>: > Maybe I have not explained it very clearly, but the issue was hidden > in using fix_nated_contact(). I commented it out and everything works > perfect :) Ok, that's a different problem. If you call fix_nated_contact() for a 302 then obviously Kamailio will rewrite the URI host part of the Contact header with the response public source address. -- Iñaki Baz Castillo <ibc(a)aliax.net>

1 0

Redirecting 302 problem
by Mino Haluz 25 Oct '10

25 Oct '10

Hi, my kamailio receives 302 from ip 1.1.1.1 with a correct Contact-header format 123@kamailiohost , but when it forwards to my cisco as, it suddenly changes the host part to 1.1.1.1 .. It is very strange because, this IP does not have anything common with the callee number. Do you know where could be a problem? Mino

3 2

bflag under failure_route (kamailio 1.5)
by Iñaki Baz Castillo 25 Oct '10

25 Oct '10

Hi, I know that handling bflags outside of branch_route is not a good idea. Anyhow I have a doubt: 1) In a normal route I set "setbflag(i:1)", so the bflag will be set for all the branches that could be generated for this incoming transaction. This also involves creating new branches manually under failure_route, am I right? Note: I do know that it would be much better just to use flag instead of bflags. 2) The outgoing transaction(s) fail so I enter into failure_route. If I inspect here bflag(i:1) will it be set or not? I remember that it will be set. 3) Then I call append_branch in failure_route, or perhaps call loockup("location") and I expect that all the generated branches will have the bflag set, am I right? Thanks a lot. -- Iñaki Baz Castillo <ibc(a)aliax.net>

2 8

Drouting usage
by Javier Gallart 25 Oct '10

25 Oct '10

Hello sorry if this sounds too "newbie". I'm interested in the drouting module. The first INVITE is correctly forwarded to the gw foudn in the drouting table: if (is_method("INVITE") && !has_totag()) { do_routing("0"); record_route(); } However for subsequent messages (ACK, BYE) in the dialog I don't know how to keep track of the ip resolved by the drouting logic so the messages are correctly forwarded to the same gw. I've made some tests using the dialog module with no success. Any hint? Thanks in advance Javier

2 1

Re: [SR-Users] Debug xmpp module with kamailio 3.0.4
by Huy Nguyen 25 Oct '10

25 Oct '10

Hi Robert and list, @Robert: thank you so much. After change some things as the guide of Robert, i can see more LM_DBG from other modules as: tm, core (msg_parser.c, t_lookup.c...... )...but what i need is the log from xmpp module (e.g xmpp.c, xmpp_component.c......). As the guide on xmpp page, i just can debug with xmpp_send_message command, and i can't check whether kamailio was registered successful as the component of local jabberd server. How can check this ? Thanks, Huy Nguyen www.htk-inc.com Date: Mon, 25 Oct 2010 13:25:14 +0200 From: robert(a)exa-omicron.nl To: huy_quocnguyen(a)live.com CC: sr-users(a)lists.sip-router.org Subject: Re: [SR-Users] Debug xmpp module with kamailio 3.0.4 On 10/25/2010 12:31 PM, Huy Nguyen wrote: log_facility=LOG_LOCAL0 How can I check all LM_DBG logs in some files as xmpp.c, xmpp_component.c .... to view the registration processing of kamailio to local jabberd s erver ? Huy, In the default syslog.conf of centos you have a line: *.info;mail.none;authpriv.none;cron.none /var/log/messages Which only sends log level info and lower to /var/log/messages. If you add a line: local0.* -/var/log/kamailio.log You will find all any debugging output of kamailio in /var/log/kamailio.log Don't forget to restart syslog Regards, Robert Verspuy -- Exa-Omicron Patroonsweg 10 3892 DB Zeewolde Tel.: 088-OMICRON (66 427 66) http://www.exa-omicron.nl

1 0

Debug xmpp module with kamailio 3.0.4
by Huy Nguyen 25 Oct '10

25 Oct '10

Hi all, About xmpp module on kamailio 3.0.4 (Centos server). I enabled DEBUG mode with level 7, but when check on /var/log/message I don't see any log from xmpp module (LM_DBG): #!ifdef WITH_DEBUG debug=7 log_stderror=no #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 If I change LM_DBG to LM_ERR, I just see 3 line as below: Oct 25 17:03:48 civilpol /usr/local/sbin/kamailio[31029]: ERROR: xmpp [xmpp.c:345]: cmd_send_message Oct 25 17:03:48 civilpol /usr/local/sbin/kamailio[31029]: ERROR: xmpp [xmpp.c:379]: message from <sip:101@sip.htk.com> Oct 25 17:03:48 civilpol /usr/local/sbin/kamailio[31029]: ERROR: xmpp [xmpp.c:388]: using R-URI as destination How can I check all LM_DBG logs in some files as xmpp.c, xmpp_component.c .... to view the registration processing of kamailio to local jabberd server ? Thanks, Huy Nguyen www.htk-inc.com

2 1

Re: [SR-Users] dictionary attacks
by Daniel-Constantin Mierla 24 Oct '10

24 Oct '10

probably omitted by mistake, but please keep the mailing list cc-ed. On 10/24/10 3:38 PM, Sergey Okhapkin wrote: > Note that I check return code of www_authorize to be -1 (invalid user) and > block IP in this case only. Other error codes should not block the IP address. This one remembered me that in 3.1 we merged the auth modules and we used the one coming from ser because it has better nonce protection and other enhancements than kamailio version. That means the return codes have changed, the new ones are listed now at: http://kamailio.org/docs/modules/stable/modules_k/auth_db.html#id2753068 Added also note in migration wiki page: http://www.kamailio.org/dokuwiki/doku.php/install:3.0.x-to-3.1.0#modules_k_… Cheers, Daniel > On Sunday 24 October 2010, you wrote: >> I watched live an attack on voipuser.org while running 3.1 before >> release. It lasted 18 hours. I didn't want to ban it because was useful >> for testing and see if it reveals any weak. In most of the cases it hit >> pike module. I got some data and plan to make an article about it soon. >> >> Anyhow, as a result of that, default config for kamailio has a section >> for detecting and banning such "bad" IPs, using pike to detect floods >> and htable to keep it blocked. Search WITH_ANTIFLOOD directive. It can >> be enhanced like you pointed here, so if the authorize fails, add the IP >> in the banned list stored in htable. >> >> Using fail2ban together with IP tables has the advantage of dropping the >> packets before getting to application and eating cpu, although in the >> case of voipuser.org the cpu was not affected much - the rate was >> 170-200 requests per second. >> >> Cheers, >> Daniel >> >> On 10/24/10 3:06 PM, Sergey Okhapkin wrote: >>> I'm second for fail2ban. I block IP addresses with failed registration >>> attempts for 1 hour. Here is my setup: >>> >>> kamailio.cfg: >>> >>> if (is_method("REGISTER")) { >>> if(www_authorize("", "subscriber")< 0) { >>> if($rc == -1) { >>> xlog("L_INFO","Invalid username from >>> $proto:$si:$sp\n"); sl_send_reply("200","OK"); >>> } else >>> www_challenge("", "0"); >>> exit; >>> } >>> .... >>> >>> /etc/fail2ban/filter.d/openser.conf: >>> >>> [Definition] >>> #_daemon = kamailio >>> failregex = Invalid username from ...:<HOST>: >>> >>> /etc/fail2ban/jail.conf: >>> >>> findtime = 600 >>> >>> [openser-iptables] >>> enabled = true >>> filter = openser >>> action = iptables-allports[name=OPENSER, protocol=all] >>> logpath = /var/log/openser/openser # Replace with your sr log location >>> maxretry = 10 >>> bantime = 3600 >>> >>> On Sunday 24 October 2010, Uriel Rozenbaum wrote: >>>> Juha, >>>> >>>> I think we should be specially careful about black-lists. We receive >>>> many of these attacks in a per-day basis and a lot of them are from >>>> residential addresses or university, so I'm guessing some kind of worm >>>> or trojan performing the attack from various IPs. >>>> >>>> If you have the time, try fail2ban deamon. It can relate some >>>> brute-force events and act accordingly blocking an IP on iptables, >>>> executing a script. You send to "jail" those addresses for a period of >>>> time, then you can get them out again; and of course you can manually >>>> revert. >>>> >>>> Last, as a description of the attacks I saw, first it runs an NMAP >>>> like scan checking which IPs answer from 5060, then it starts sending >>>> registers (usually asterisk answers 404 if the user does not exist), >>>> then when the proxy challenges, it interprets the user is found and >>>> starts making dictionary attacks on the password (1234, admin, and so >>>> on). Keep safe complicated passwords, make kamailio challenge >>>> everything and you'll be safe. and again, fail2ban is a pretty good >>>> solution for brute force. >>>> >>>> This might help you finding a solution for your attacks. >>>> >>>> Cheers, >>>> Uriel >>>> >>>> On Sun, Oct 24, 2010 at 8:54 AM, Juha Heinanen<jh(a)tutpro.com> wrote: >>>>> while doing some tests, i noticed that one of my proxies started to >>>>> receive lots of register requests with different user names starting >>>>> from a letter. there was also invite attempts in the logs. they came >>>>> from ip 202.82.16.99 which according to traceroute is somewhere in >>>>> china. >>>>> >>>>> should we start publishing a black list of these attack ip addresses? >>>>> >>>>> -- juha >>>>> >>>>> _______________________________________________ >>>>> 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 >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>> sr-users(a)lists.sip-router.org >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://www.asipto.com

3 7

dictionary attacks
by Juha Heinanen 24 Oct '10

24 Oct '10

while doing some tests, i noticed that one of my proxies started to receive lots of register requests with different user names starting from a letter. there was also invite attempts in the logs. they came from ip 202.82.16.99 which according to traceroute is somewhere in china. should we start publishing a black list of these attack ip addresses? -- juha

6 10

Reverse Filterlists idea
by "Nicolas Rüger" 23 Oct '10

23 Oct '10

Hello, Marius Zbihlei has patched the userblacklist module, so that it can handle characters as well (NOT in main release 3.1 yet). Thanks Marius. Therefore it might now be possible to filter general SIP-URIs!?? My idea is simple and described here. Please give some Feedback!!! As the "prefix"-functionality up to now was referring to SIP-URIs consisting of digits (=real telephon numbers) like 004930123456(a)1and1.de the general URI might be something like user_123(a)server.domain.de So far the "prefix"-functionality works because usernames that are real telephone numbers, that always start with country-code, followed by town- or regional-code and end with a unique number of the user (in this example 0049=Germany, (0)30=Berlin, 123456=user's unique number). To use "prefix"-functionality with general (including non-digits) URIs it must be evaluated reverse (from back to front) as the domain ends with the country- or top-level-domain and becomes more detailed the reverse way. So the IDEA is: Insert the domain as prefix in userblacklist-table in reverse, to use the functionality. I would use a perl-script to reverse the SIP-URI of the calling party in routing logic and then check it against the already reverted domain in the userblacklist-table. Do you think this is a good/well-working idea??? Any concerns or suggestions are appreciated... Regards, Nicolas -- GMX DSL Doppel-Flat ab 19,99 €/mtl.! Jetzt auch mit gratis Notebook-Flat! http://portal.gmx.net/de/go/dsl

4 8

Re: [SR-Users] Kamailio unable to forward Register request to asterisk
by Ovidiu Sas 22 Oct '10

22 Oct '10

You need to have no 'secret' field in the asterisk sipusers view. Please keep the list in cc. Private e-mail will go unanswered. Regards, Ovidiu Sas On Fri, Oct 22, 2010 at 1:53 PM, Sundeep Kumar Mishra <sundeep123kumariiit(a)gmail.com> wrote: > Hi, > Thanks for you quick reply. > We understand where we had been wrong. Since it was not quite clear in > the tutorial could u tell us how to disable authentication in asterisk? > Regards, > Sundeep Kumar Mishra > > > On Fri, Oct 22, 2010 at 7:29 PM, Ovidiu Sas <osas(a)voipembedded.com> wrote: >> >> If you use that particular config, you need to disable authentication >> on the asterisk side. >> Just read careful the tutorial. >> >> Regards, >> Ovidiu Sas >> >> On Fri, Oct 22, 2010 at 9:15 AM, Sundeep Kumar Mishra >> <sundeep123kumariiit(a)gmail.com> wrote: >> > Hi all, >> > I have installed an asterisk server at 10.3.3.143:5080 and a >> > kamailio at 10.3.3.85:5060. I am using sip communicator as softphone SIP >> > Client. It is able to register at kamailio but the kamailio is not able >> > to >> > forward register request to asterisk. >> > >> > We have gone through the tutorial >> > >> > http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb >> > and used the configuration file after changing the asterisk ip and >> > kamailio >> > ip. >> > >> > We found out that when a normal softphone authenticates directly to >> > asterisk >> > it sends and recieve >> > >> > |--softphone -- REGISTER >> > |--asterisk -- 401 Unauthorised . CHallenge >> > |--softphone -- REGISTER - Response to Challenge >> > |--asterisk -- 200 OK (If response was correct) >> > >> > But I saw that when kamailio forwards register to asterisk, it sends >> > first >> > REGISTER packet but doesn't second the second one. It sends SUBSCRIBE >> > instead of REGISTER with response to the challenge. >> > >> > >> > Regards, >> > Sundeep Kumar Mishra >> > >> > _______________________________________________ >> > 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

Kamailio unable to forward Register request to asterisk
by Sundeep Kumar Mishra 22 Oct '10

22 Oct '10

Hi all, I have installed an asterisk server at 10.3.3.143:5080 and a kamailio at 10.3.3.85:5060. I am using sip communicator as softphone SIP Client. It is able to register at kamailio but the kamailio is not able to forward register request to asterisk. We have gone through the tutorial http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdba… used the configuration file after changing the asterisk ip and kamailio ip. We found out that when a normal softphone authenticates directly to asterisk it sends and recieve |--softphone -- REGISTER |--asterisk -- 401 Unauthorised . CHallenge |--softphone -- REGISTER - Response to Challenge |--asterisk -- 200 OK (If response was correct) But I saw that when kamailio forwards register to asterisk, it sends first REGISTER packet but doesn't second the second one. It sends SUBSCRIBE instead of REGISTER with response to the challenge. Regards, Sundeep Kumar Mishra

3 2

Error in syslog: BUG tm [t_msgbuilder.c:351]: unhandled reason cause -18344
by Mikko Lehto 22 Oct '10

22 Oct '10

Hi I am testing certain UAC behavior on proxy initiated 408 response with SIP Router. Call canceling is working fine - UAC gets 408 and GW gets CANCEL. However this error message is seen in syslog after fr_inv_timeout fires: BUG: tm [t_msgbuilder.c:351]: unhandled reason cause -18344 In script I set t_set_fr before t_relay, otherwise it is default: route[RELAY] { if (is_method("INVITE")) { t_on_reply("REPLY_ONE"); t_on_failure("FAIL_TEST"); } t_set_fr(4000); if (!t_relay()) { sl_reply_error(); } exit; } SIP Router is self compiled from Git: sip-router 3.1.0 (i386/linux) 34d2f6 Here are TM parameters: #modparam("tm", "local_cancel_reason", "0") modparam("tm", "noisy_ctimer", 1) modparam("tm", "auto_inv_100_reason", "tryink") modparam("tm", "failure_reply_mode", 3) modparam("tm", "fr_timer", 30000) modparam("tm", "fr_inv_timer", 120000) The above error message is found in t_msgbuilder.c on #ifdef CANCEL_REASON_SUPPORT -block. Looks like I do not have compiled in support for that new feature as I get error when uncommenting local_cancel_reason tm parameter: ERROR: <core> [modparam.c:150]: set_mod_param_regex: parameter <local_cancel_reason> not found in module <tm> Is this bug as log message suggests? -- Mikko Lehto Setera

2 4

Siremis user guide
by Shrouk Khan 22 Oct '10

22 Oct '10

hi has anyone ever found any user guide for siremis ? it seems to take information directly from database and showing it . there is no tool tip or any clue as to what is what -- Regards Shrouk Khan (Khan) System Administrator / Telecommunication System Developer Office: +354 4400807 (Reykjavik) +44 2031370800 (London) Mobile: +66 875049439 (Bangkok) Web: www.softverk.is Reykjavik, Iceland // London, UK // Bangkok, Thailand

3 4

[OT] Fwd: [dispatch] proposed SIXPAC charter
by Victor Pascual Avila 22 Oct '10

22 Oct '10

---------- Forwarded message ---------- From: Elwell, John <john.elwell(a)siemens-enterprise.com> Date: Wed, Oct 20, 2010 at 9:45 AM Subject: Re: [dispatch] proposed SIXPAC charter To: Peter Saint-Andre <stpeter(a)stpeter.im>, "dispatch(a)ietf.org" <dispatch(a)ietf.org> I find this a worthwhile topic to pursue. I had been wondering whether this activity would turn out to be more of a profiling exercise, and whether the IETF might not be the best choice of venue for such work. >From the current draft charter it looks like there will be at least some protocol extension work, for which I believe the IETF is the correct venue. On the other hand, the Unified Communications Interoperability Forum (UCIF) is seeking to advance the state of play on XMPP interoperability, and if we were just talking about a profile or BCP, that might have been a better venue. Perhaps the IETF should focus on requirements and protocol extensions, and consider whether the BCP work would be better done elsewhere. Or at least, there should be some coordination with other activities relating to XMPP interoperability. John > -----Original Message----- > From: dispatch-bounces(a)ietf.org > [mailto:dispatch-bounces@ietf.org] On Behalf Of Peter Saint-Andre > Sent: 19 October 2010 21:17 > To: dispatch(a)ietf.org > Subject: [dispatch] proposed SIXPAC charter > > Earlier this year, some folks in the RAI area proposed an > initiative to > define a few small extensions to both SIP and XMPP that would make it > easier to develop and deploy dual-stack SIP+XMPP endpoints. Based on > feedback provided on the DISPATCH list and received from the RAI ADs, > I've taken the liberty of rewriting the proposed charter, in the hopes > that fresh text will spur a more conclusive discussion. I'm > mostly just > trying to help the proponents put their best foot forward, so if folks > here have more feedback I'd expect that people like Simo Veikkolainen > and Emil Ivov will be able to engage in further discussion. > > /psa > > ### > > SIXPAC (SIP Integration with XMPP in Presence Aware Clients) > ============================================================== > =========== > > Problem Statement > > Both the Session Initiation Protocol (SIP) and the Extensible > Messaging > and Presence Protocol (XMPP) are widely deployed technologies for > real-time communication over the Internet. In order to offer > a complete > suite of features as well as communication across multiple networks, > several user-oriented software applications support both SIP and XMPP, > and more software developers have expressed interest in building such > "dual-stack" solutions. Unfortunately, it is difficult to provide a > good end-user experience in such applications because SIP and XMPP are > not aware of each other. For example: > > - XMPP presence does not include availability states related to a SIP > voice call or video call (e.g., "on the phone"), thus preventing an > a dual-stack endpoint from showing presence-based > communication hints > > - There is no correlation between an XMPP IM session and a SIP voice > or video session, thus preventing a dual-stack endpoint > from providing > integrated user interfaces and communications history > > - SIP accounts and XMPP accounts are not directly correlated > in contact > lists or vCards (and not all deployed services support > storage of such > information), thus preventing a dual-stack endpoint from > knowing that > a contact has both SIP and XMPP capabilities > > Although some proprietary solutions exist to the foregoing > problems, it > would be preferable to define standardized solutions for the sake of > improved interoperability. > > Objectives > > Because both SIP and XMPP are easily extended through new SIP headers > and XMPP elements, it should be possible to provide tighter > integration > within dual-stack SIP/XMPP user agents to improve the user experience. > > Any such extensions should meet the following criteria: > > - Be completely optional and backwards-compatible for all endpoints > > - Work without changes to deployed infrastructure such as existing > SIP and XMPP servers, B2BUAs, firewalls, etc. > > The SIXPAC WG will define a small number of SIP and XMPP extensions to > solve the following use cases in dual-stack endpoints: > > - Including SIP-based availability states in XMPP presence (limited to > basic presence and availability states only, not the full range of > PIDF extensions) > > - Correlating an XMPP IM session with a SIP voice/video session, and > vice-versa > > - Advertising a SIP account address over XMPP and an XMPP account > address over SIP > > Additional use cases are out of scope, including anything > related to or > requiring server integration, multiparty communication, SIP-based IM > and presence, XMPP-based voice and video, file transfer, generalized > service discovery and capabilities exchange, full protocol translation > in communication gateways, shared credentials across both SIP and XMPP > accounts, rich presence extensions for features such as geolocation, > etc. Although such topics are important and interesting, they are out > of scope for this group. > > However, in addition to the protocol extensions explicitly mentioned > above, the group may also define best practices related to the > implementation and deployment of dual-stack SIP/XMPP endpoints, > including topics such as user agent configuration. > > Deliverables > > - Use cases and protocol requirements > > - XMPP presence extension for SIP-based availability states > > - Media session correlation extensions for SIP and XMPP > > - Contact address advertisement extensions for SIP and XMPP > > - Best practices for implementation and deployment of dual-stack > endpoints > > Milestones > > Feb 2011 Submit use cases and protocol requirements document to > IESG (Informational) > > Oct 2011 Submit XMPP presence extension document to IESG > (Standards Track) > > Oct 2011 Submit media session correlation extensions document to IESG > (Standards Track) > > Oct 2011 Submit contact address advertisement extension document to > IESG (Standards Track) > > Oct 2011 Submit best practices document to IESG (Informational) > > ### > > > _______________________________________________ dispatch mailing list dispatch(a)ietf.org https://www.ietf.org/mailman/listinfo/dispatch -- Victor Pascual Ávila

5 40

Re: [SR-Users] Kamailio 3.0 cann't access the RTPProxy
by Daniel-Constantin Mierla 22 Oct '10

22 Oct '10

Hello, apart of the email with logs, or private data that was required, please continue the discussion on public mailing list. I need some time to check the logs, the first inconvenience is the rar archive, a zip or tgz is much better -- not a fan of installing binaries from unknown companies on mac os x, going to get unrar from macports. Then solaris is not an OS i have at hand nor use frequently, so need to read about. It is also why using mailing list keeps you in touch with others that can help. Cheers, Daniel On 8/18/10 7:04 PM, KevinJin wrote: > Hi Daniel, > > Does the logs have any hint for what's cause of the issue? I didn't > include the user list in the previous email since logs have the actual > IP info. > > Thanks in advance! > > Best Regard > Kevin > > ------------------------------------------------------------------------ > From: kevin.jzh(a)hotmail.com > To: miconda(a)gmail.com > Subject: RE: [SR-Users] Kamailio 3.0 cann't access the RTPProxy > Date: Tue, 17 Aug 2010 18:51:02 +0800 > > > Hi Daniel, > > Attached are the two kamailio logs, > unix_socket_log --- run rtp proxy wth -s unix:/tmp/rtpproxy.sock > udp_rtpproxy_log --- run rtp proxy with -s udp:*:7722 > > Please help to check what's wrong with it. > > Thanks, > Kevin > ------------------------------------------------------------------------ > Date: Tue, 17 Aug 2010 11:29:46 +0200 > From: miconda(a)gmail.com > To: kevin.jzh(a)hotmail.com > CC: sr-users(a)lists.sip-router.org > Subject: Re: [SR-Users] Kamailio 3.0 cann't access the RTPProxy > > Hello, > > please send full log at startup, your snippets include just few lines > per process, being mixed from different processes. > > Cheers, > Daniel > > > On 8/14/10 3:54 AM, KevinJin wrote: > > Hello, > > ------------------------------------------------------------------------ > Date: Thu, 12 Aug 2010 22:16:18 +0200 > From: miconda(a)gmail.com <mailto:miconda@gmail.com> > To: kevin.jzh(a)hotmail.com <mailto:kevin.jzh@hotmail.com> > CC: sr-users(a)lists.sip-router.org > <mailto:sr-users@lists.sip-router.org> > Subject: Re: [SR-Users] Kamailio 3.0 cann't access the RTPProxy > > Hello, > > On 8/12/10 8:34 PM, KevinJin wrote: > > Hello, > > ------------------------------------------------------------------------ > Date: Thu, 12 Aug 2010 18:46:19 +0200 > From: miconda(a)gmail.com <mailto:miconda@gmail.com> > To: kevin.jzh(a)hotmail.com <mailto:kevin.jzh@hotmail.com> > CC: sr-users(a)lists.sip-router.org > <mailto:sr-users@lists.sip-router.org> > Subject: Re: [SR-Users] Kamailio 3.0 cann't access the RTPProxy > > Hello, > > On 8/12/10 4:47 PM, KevinJin wrote: > > Hi Daniel, > > What does the log below means? Does it mean nathelper has > issue to send the request to RTP proxy first or nathelper > doesn't receive a response after sending a request to the > rtp proxy? > 0(27429) ERROR: nathelper [nathelper.c:2457]: can't send > command to a RTP proxy > > this error is printed when write to socket fails. Do you have > any firewall running on the system? Is the user under which > kamailio runs allowed to write to sockets? > > > There's no firewall on the system, and I run the kamailio > as root, > root 26310 1 0 02:24:19 ? 0:00 > /usr/local/kamailio-3.0.2/sbin/kamailio -f > /usr/local/kamailio-3.0.2/etc/kamail > > You can edit module_k/nathelper/nathelper.c and replace > the line 2457 with: > > LM_ERR("can't send command to a RTP proxy (%s/%d)\n", > strerror(errno), errno); > > Recompile and reinstall. Hopefully will get more hints > about what happens. > > > Here is the error message after the change: > 2(26312) ERROR: nathelper [nathelper.c:2457]: can't send > command to a RTP proxy(Invalid argument/22) > 2(26312) ERROR: nathelper [nathelper.c:2492]: proxy > <udp:210.13.124.15:7722> does not respond, disable it > 2(26312) ERROR: nathelper [nathelper.c:3144]: no available proxies > what could be the cause? > > hmm, invalid argurment ... try with this line: > > LM_ERR("can't send command to a RTP proxy (%s/%d) [sock %d (%d), > vcnt %d]\n", > strerror(errno), errno, rtpp_socks[node->idx], node->idx, vcnt); > > maybe will give some hints about which value is invalid. > > Here's the log after the change: > 4(14415) ERROR: nathelper [nathelper.c:2457]: can't send command > to a RTP proxy (Invalid argument/22) [sock 7 (0), vcnt 18] > 4(14415) ERROR: nathelper [nathelper.c:2492]: proxy > <udp:210.13.x.y:7722> does not respond, disable it > 4(14415) ERROR: nathelper [nathelper.c:3144]: no available proxies > > > Can you try as well with an unix file socket: > > modparam("nathelper", "rtpproxy_sock", "unix:/tmp/rtpproxy.sock") > > then start rtpproxy with -s unix:/tmp/rtpproxy.sock > > > 4(17530) INFO: nathelper [nathelper.c:2369]: rtp proxy > <unix:/tmp/rtpproxy.sock> found, support for it re-enabled > 3(17529) ERROR: nathelper [nathelper.c:2429]: can't send command > to a RTP proxy > 3(17529) ERROR: nathelper [nathelper.c:2492]: proxy > <unix:/tmp/rtpproxy.sock> does not respond, disable it > 3(17529) ERROR: nathelper [nathelper.c:3144]: no available proxies > > Thanks, > Kevin > > I have no solaris (sparc) to try myself... > > Cheers, > Daniel > > > > Test env: > UA1 (Behind NAT) --------> Kamailio & RTPproxy (Public IP) > --------->UA2 (Public IP) > > Thanks, > Kevin > Cheers, > Daniel > > > > 0(27429) ERROR: nathelper [nathelper.c:2492]: proxy > <udp:210.13.124.15:7722> does not respond, disable it > > There's no problem for the resource(CPU, mem etc.) on the > server, the load is very low. > > Thanks in advance! > ---------- > 0(27429) DEBUG: nathelper [nhelpr_funcs.c:148]: type > <application/sdp> found valid > 0(27429) ERROR: nathelper [nathelper.c:3144]: no available > proxies > 0(27429) ERROR: nathelper [nathelper.c:2627]: no available > proxies > 0(27429) DEBUG: nathelper [nhelpr_funcs.c:148]: type > <application/sdp> found valid > 0(27429) INFO: nathelper [nathelper.c:2369]: rtp proxy > <udp:210.13.124.15:7722> found, support for it re-enabled > 0(27429) DEBUG: nathelper [nathelper.c:3196]: proxy reply: > 42040 210.13.124.14 > 0(27429) DEBUG: nathelper [nhelpr_funcs.c:148]: type > <application/sdp> found valid > 0(27429) ERROR: nathelper [nathelper.c:2457]: can't send > command to a RTP proxy > 0(27429) ERROR: nathelper [nathelper.c:2492]: proxy > <udp:210.13.124.15:7722> does not respond, disable it > 0(27429) ERROR: nathelper [nathelper.c:3144]: no available > proxies > 0(27429) ERROR: nathelper [nathelper.c:2627]: no available > proxies > > Thanks, > Kevin > ------------------------------------------------------------------------ > > > _______________________________________________ > 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/ > > > -- > Daniel-Constantin Mierla > http://www.asipto.com/ > > > -- > Daniel-Constantin Mierla > http://www.asipto.com/ -- Daniel-Constantin Mierla http://www.asipto.com/

2 5

Integrate xmpp module with kamailio 3.0.1
by Huy Nguyen 22 Oct '10

22 Oct '10

Hi all, Sorry for restore this topic, but i till confused about the guide on xmpp module page. I'm integrating xmpp module to kamailio 3.0.x at my local network, with server mode. The kamailio server have domain: kamailio1.com (192.168.1.26), the xmpp server have domain: xmpp1.com (192.168.1.20). SIP user: 101(a)kamailio1.com XMPP user: 109(a)xmpp1.com Below is the part mention in xmpp doc I confused about, so i'm pasting that: In the destination address we need an extra character as a domain separator, apart from @. The address of the XMPP(jabber) client looks like this: "sip:username<domain_separator>jabber_server@gateway_domain". The address of a SIP client has the following pattern : "sip_username<domain_separator>openser_domain@xmpp_domain"; A common used character in XMPP transports is "%". It mean, from sip client 101(a)kamailio1.com, if I want to chat with xmpp client 109(a)xmpp1.com, I'll add 109 to the contact list of 101 as: sip:109%xmpp1.com@kamailio1.com. It right ? As the similar, i'll add to the buddy list of 109 as: 101%kamailio1.com(a)xmpp1.com ? Can anyone suggest how to test xmpp module ? Thanks, Huy Nguyen www.htk-inc.com

2 6

Register a different URI than the received in the To header
by Iñaki Baz Castillo 21 Oct '10

21 Oct '10

Hi, not sure if this question has already taken place in this maillist: For some exotic reason sometimes the registrar receives a REGISTER with "To: sip:alice@domain.com" but I want to save it in the location table as "sip:bob@domain.com". AFAIK there is no way to do it, am I wrong? -- Iñaki Baz Castillo <ibc(a)aliax.net>

2 3

Needed in Dallas, Texas, Network Voice Engineer and Field Service Install Technician
by JR Richardson 21 Oct '10

21 Oct '10

We have a couple of positions open, please respond to the posting if qualified and interested. http://www.ntegrated.net/resources/job-opportunities/field-service-install-… http://www.ntegrated.net/resources/job-opportunities/network-engineering-vo… These are full time positions in Dallas, no telecommuters please. Thanks. JR -- JR Richardson Engineering for the Masses

1 0

Kamalio 3.1 based turn-key distribution released !
by X- x -X 21 Oct '10

21 Oct '10

Hey all, Just wanted to share this link. A distribution based on debian (lenny) & kamailio 3.1 pre-configured with presence server/RTP thing has been released. size: 245 MB. http://ciitix.ciit.net.pk/index.php/home/6-announcements-/14-ciitix-voip-10… Download link http://ciitix.ciit.net.pk/index.php/downloads Cheers --- X

2 1

siremis 1.0.1 with kamalio 3.1 ?
by X- x -X 21 Oct '10

21 Oct '10

Hi all, Not sure if this should be put to siremis list or dev list or whatever but bear with me if you could. Since siremis v 2.0 is not out. Is siremis 1.0.1 known to work fine with kamalio 3.1 ? (ref: siremis 1.0.1 install http://siremis.asipto.com/install/ ) Is it ready for production deployment etc.. ? Will greatly appreciate any response from anyone since Ramona seems to be on her sabbatical . Thanks. --

2 1

siremis 2.0 ?
by X- x -X 21 Oct '10

21 Oct '10

Hey all, Congratulation on 3.1 release ! Any news on when siremis v 2.0 is hitting the stores ? In the mean time what about alpha/beta version of it ? Its nowhere on the repos/sites (asipto etc..) Is version 1.0.1 know to work smoothly with kamailio 3.1 ? Cheers --

2 1

HTable 'size' parameter
by Iñaki Baz Castillo 21 Oct '10

21 Oct '10

Hi, in the doc of HTable module all the examples show "a=>size=4;". As the doc says: size - number specifying the size of hash table. The number of entries in the table is 2^size This is, the examples just allow 16 concurrent entries, which is not enough in most of the cases. Did I misunderstand something? or do the module examples use a non very pertinent value? Thanks a lot. -- Iñaki Baz Castillo <ibc(a)aliax.net>

5 8

Kamailio v3.0.4 Released
by Daniel-Constantin Mierla 20 Oct '10

20 Oct '10

Hello, Kamailio (OpenSER) v3.0.4 is released. This is a maintenance release of previous stable branch, 3.0.x, that includes fixes since release of v3.0.3. There is no change to database or configuration file required to upgrade to 3.0.4 from any 3.0.x version. For more details about version 3.0.4, visit: http://www.kamailio.org/w/2010/10/kamailio-v3-0-4-released/ Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

Kamailio v1.5.5 Released
by Daniel-Constantin Mierla 20 Oct '10

20 Oct '10

Hello, Kamailio (OpenSER) version 1.5.5 is released. This is a maintenance release of branch 1.5, the last in its series. The project is officially maintaining the latest two stable branches, these being now 3.0.x and 3.1.x. As it happened in the past, fixes can still get into SVN branch 1.5, but there is no plan to package another version 1.5.x. If you still run 1.5.x, it is highly recommended that you start to plan the migration to 3.x versions. For more details about version 1.5.5, visit: http://www.kamailio.org/w/2010/10/kamailio-v1-5-5-released/ Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

Crash on processing dialog BYE
by Alex Balashov 20 Oct '10

20 Oct '10

We had another one of these today, under high call volume: (gdb) where #0 0x0000003a60430265 in raise () from /lib64/libc.so.6 #1 0x0000003a60431d10 in abort () from /lib64/libc.so.6 #2 0x0000000000530a91 in qm_free (qm=0x2b343e1aa000, p=0x2b343eddf6e8, file=0x2b343dd860c3 "dialog: dlg_hash.c", func=0x2b343dd86b42 "destroy_dlg", line=176) at mem/q_malloc.c:447 #3 0x00002b343dd6a2ea in destroy_dlg (dlg=0x2b343fe1d6f8) at dlg_hash.c:176 #4 0x00002b343dd6d33a in unref_dlg (dlg=0x2b343fe1d6f8, cnt=1) at dlg_hash.c:591 #5 0x00002b343dd73a04 in profile_cleanup (msg=<value optimized out>, flags=<value optimized out>, param=0x6) at dlg_profile.c:317 #6 0x00000000004be9a1 in exec_post_script_cb (msg=0xa06e80, type=<value optimized out>) at script_cb.c:195 #7 0x00000000004989f0 in receive_msg ( buf=0x8b6300 "BYE sip:15746317356@yyy.yyy.yyy.yyy:5060 SIP/2.0\r\nRecord-Route: <sip:yyy.yyy.yyy.yyy;lr;ftag=gK0ebaf6d4>\r\nRecord-Route: <sip:67.231.8.89;lr;ftag=gK0ebaf6d4>\r\nVia: SIP/2.0/UDP xxx.xxx.xxx.xxx;branch=z9hG4bKd757."..., len=<value optimized out>, rcv_info=0x7fff4a043230) at receive.c:221 #8 0x000000000052576a in udp_rcv_loop () at udp_server.c:532 #9 0x0000000000468dcd in main_loop () at main.c:1554 #10 0x000000000046be9f in main (argc=<value optimized out>, argv=0x7fff4a043508) at main.c:2398 Is this a regression, or memory corruption from something else? This is with CANCEL_REASON_SUPPORT #undef'd in tm_reply.h, btw. -- Alex Balashov - Principal Evariste Systems LLC 1170 Peachtree Street 12th Floor, Suite 1200 Atlanta, GA 30309 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

4 7

pdb_query dstavp argument
by Juha Heinanen 20 Oct '10

20 Oct '10

why dstavp argument of pdb_query() needs to be an avp? why a var is not accepted? -- juha

2 2

kamailio 1.5.5 and 3.0.4
by Daniel-Constantin Mierla 20 Oct '10

20 Oct '10

Hello, Based on our project policy to support officially the last two stable branches, I was thinking of packaging kamailio 1.5.5 to mark its end. New fixes will go to svn if it is the case, but no new release in 1.5.x series will be done. Also, since testing for 3.1.0 brought some fixes to 3.0.x, we should do as well 3.0.4. My proposal is next week, Wednesday, Oct 20. Don't forget to report issues you are aware for these two branches. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

4 6

check_user_black_list error
by Bruno Machado 20 Oct '10

20 Oct '10

Hi I'm using version 1.5.4 and the module userblacklist is not working how I was expecting. The function check_user_black_list() returns the correct value. After some 5 minutes, it changes its return value. I'm blocking users that are not in the table. However, after this time, the functions says that they exist, but it is not true. Someone know why it happens? Could it be a module's bug? #avp 100 is a part of the username's RURI if(check_user_blacklist("$fU","$rd","$avp(s:100)")) { sl_send_reply("403", "Forbidden"); exit; } Tks, bruno machado

2 2

space in Allow field of location table causes failure in dbtext reload - v 3.0.0
by Jeremya 20 Oct '10

20 Oct '10

Hi, I have an application that uses dbtext. Starting with an empty location table I have a client register with kamailio. When kamailio is shut down the location records are written. When kamailio starts again it fails when reading the location value. The client does not provide an Allow: header. Field 16 in the location table is a parsed version of the Allow: header. When kamailio shuts down it writes a space ' ' for the value of field 16. This then causes the re-read to fail. Removing the space makes it all work again. I'd like to report this as a bug - and/or - even better - to get a quick patch to sort the problem. Is this the right place to report? Or is there an official form? Thanks Jeremy

2 1

set cfg variables via sercmd
by Morten Isaksen 19 Oct '10

19 Oct '10

Kamailio 3.0.3. I am trying to change a variable (in the cfg variable framwork) from sercmd, but I get an error [root@sip-core-1 ~]# sercmd cfg.set_now_int recording enabled 1 error: 500 - command cfg.set_now_int not found I have installed readline and ncurses which seems to be needed by looking at sercmd.c. Any ideas what I am missing? -- Morten Isaksen

2 2

Fwd: Tutorial: SIP SIMPLE Presence Made Simple with Kamailio 3.1
by Rouskol Andrey 19 Oct '10

19 Oct '10

Daniel, What whould you think about this variant of xcap authentication: event_route[xhttp:request] { xdbg("===== xhttp: request [$rv] $rm => $hu\n"); if($hu=~"^/xcap-root/") { # xcap ops $xcapuri(u=>data) = $hu; if($xcapuri(u=>xuid)=~"^sip:.+@.+") $var(uri) = $xcapuri(u=>xuid); else if($xcapuri(u=>xuid)=~".+@.+") $var(uri) = "sip:"+ $xcapuri(u=>xuid); else $var(uri) = "sip:"+ $xcapuri(u=>xuid) + "@" + $Ri; if($xcapuri(u=>auid)=="xcap-caps") { ... skipped ... } #!ifdef WITH_XHTTPAUTH #!ifdef WITH_MULTIDOMAIN if (!www_authorize("$(var(uri){uri.host})", "subscriber")) { www_challenge("$(var(uri){uri.host})", "0"); #!else if (!www_authorize("xcap", "subscriber")) { www_challenge("xcap", "0"); #!endif exit; } set_reply_close(); set_reply_no_connect(); #!ifdef WITH_XHTTPAUTH # be sure auth user access only its documents #!ifdef WITH_MULTIDOMAIN if($aU=~".+@.+") $var(tmp) = "sip:"+$aU; else $var(tmp) = "sip:"+$aU+"@"+$(var(uri){uri.host}); if ($var(uri)!=$var(tmp)) { xdbg("===== xhttp: Forbidden!!! $var(tmp) - $var(uri)\n"); #!else if ($au!=$(var(uri){uri.user})) { xdbg("===== xhttp: Forbidden!!! $au - $(var(uri){uri.user})\n"); #!endif .. Regards, Andrey.

3 6

Get registration information of a specific subscriber in script
by Adieu 19 Oct '10

19 Oct '10

Hello everybody, I just started to use kamailio. For the basic functions, I can get them working properly. But when I started to write my own script, I encountered some problem. The scenario is like this: I have a linksys spa3102 pstn-voip gateway at home which register itself at my server on the internet. I use adsl at home so the ip address change dynamically. When I want to route all my pstn calls to spa3102, I have to figure out the current ip address of my home connection. I tried to use dynamic dns, but my router does not update the ip address sometimes. So I was thinking about solving this problem in the kamailio configuration script. All I need is to get the registration information of spa3102 so that I can route my pstn call to it. I guess there might be a function doing this work. It expect a subscriber name as a variable and returns the registration information of the subscriber. I tried to dig the function out of kamailio documents but failed. And google doesn't seem to provide much help. I don't know if this function exist or not. Could someone confirm my guess or point out where I can find it? I will be much appreciated. //bow Adieu

2 2

#!endif and SUBSCRIBE issues 3.0.x
by Jeremya 19 Oct '10

19 Oct '10

I've discovered - after quite a while - that #!endif doesn't work if there is extra text on the line #!endif works #!endif # someblock doesn't work. Could this be addressed in code or at least noted in documentation? I also have the case of SUBSCRIBE where it is a later (re)SUBSCRIBE after an initial SUBSCRIBE. This has a to-tag. This is not handled by the default script. Could the default script be altered to handle SUBSCRIBE with to-tag please? Thanks Jeremy

3 2

Fwd: perl_exec() using parameter?
by "Nicolas Rüger" 19 Oct '10

19 Oct '10

Hello, any ideas for parameter passing to perl-scripts are appreciated!!! Regards, Nicolas -------- original-------- Hello, is there any chance to pass parameters (like from_uri or to_uri) from the kamailio routing logic to a perl script? I know that the whole sip message is provided as parameter but I wonder if there's a way to pass just the caller's uri (from tag) or the callee's uri (to tag) as a second parameter to the perl script. By using xlog it's clear that in $fu and $ru one can find the uri's. xlog("from: $fu to: $ru \n") OUTPUT: "from: sip:user01@mykamailio.de to:user02@mykamailio.de" BUT using these variables as a second parameter in perl_exec() leads to an error on kamailio start-up cause of bad config file. perl_exec("my_perl_script",$fu); Any suggestions or ideas??? Thank you... Regards, Nicolas -- GRATIS! Movie-FLAT mit über 300 Videos. Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome

7 21

SER+nathelper+RTPproxy
by Medina Curafina 18 Oct '10

18 Oct '10

Dear Sirs, I am writing to you, 'cause I have some issues about SER work. I installed ser-0.9.4, and test it in my local network, and everything works fine with hello_world.cfg. After that I had different situation. I wanted to make a call from one of my local hosts to remote host (which I registered to your SERWeb in format "username"@iptel.org). Of course I changed my ser.cfg file from hello_world.cfg into example.cfg (which one I send you in Attachment, grey parts are changed lines, because I need to show work of nathelper module of SER). I Installed RTP Proxy from CVS, and followed basic instructions. so, in my local network I am host 102(a)192.168.1.103 (x-Lite4) my domain is default one* (localhost)*, my NAT box is Cisco Lynksis Router. When I make a call (host 102-->SER-->NAT translation-->SERWeb(213.192.59.75)--->external IP of my friends PC), I can reach him (He can hear a phone call, but when he answers can't hear me, and I still get ringing tone, and after 7 second it looks like he canceled a call). In Wireshark I captured INVITE, 100, 180,200, and 487 "Request Terminated" messages. In section of VOIP Calls Graph, I see that messages didn't pass further off SERWeb. I am just beginner in SER use, and I really need your help. I really tried to understand what to do, I get through documentation about RTP Proxy and I feel that I make some mistakes in that part. I was passing through this for so long time, and I don't have anybody to ask for help except you. Do I need to configure my Router, fix configuration of host, configuration file, or something about RTP proxy. Hope hearing from you soon, Very Grateful, Medina H.

1 1

SER nathelper+RTPproxy
by Medina Curafina 18 Oct '10

18 Oct '10

I am writing to you, 'cause I have some issues about SER work. I installed ser-0.9.4, and test it in my local network, and everything works fine with hello_world.cfg. After that I had different situation. I wanted to make a call from one of my local hosts to remote host (which I registered to SERWeb in format "username"@iptel.org). Of course I changed my ser.cfg file from hello_world.cfg into example.cfg (which one I send you in Attachment, grey parts are changed lines, because I need to show work of nathelper module of SER). I Installed RTP Proxy from CVS, and followed basic instructions. so, in my local network I am host host(a)192.168.x.x <102(a)192.168.x.x.> (x-Lite4) my domain is default one* (localhost)*, my NAT box is Cisco Lynksis Router. When I make a call (host 102-->SER-->NAT translation-->SERWeb(213.192.59.75)--->external IP of my friends PC), I can reach him (He can hear a ringing tone, but when he answers can't hear me, and I still get ringing tone, and after 7 second it looks like he canceled a call). In Wireshark I captured INVITE, 100, 180,200, and 487 "Request Terminated" messages. In section of VOIP Calls Graph, I see that messages didn't pass further off SERWeb. I am just beginner in SER use, and I really need your help. I really tried to understand what to do, I get through documentation about RTP Proxy and I feel that I make some mistakes in that part. Is there anyone who can give me advice what I should do, and where am I wrong? Hope hearing from you soon, Very Grateful, Medy H.

1 0

Kamailio as IPv6 to IPv4 Gateway
by Joe Uelk 18 Oct '10

18 Oct '10

Hello all, I'm looking to implement the following scenario: Step 1 SIP Server A sends INVITE to port 5060 over IPv6 to Kamailio: 2001::1 --udp/tcp--> 2001::2:5060 Step 2 Kamailio SIP NATs the INVITE and sends it out IPv4 to SIP Server B on port 6000 1.1.1.1 --udp/tcp--> 1.1.1.2:6000 Step 3 All SIP Server B responses within this SIP dialog are sent back to SIP Server A along the same path. There is more to this but I want to accomplish this first and go from there. Can someone point me toward a resource on how to get this created? Do I need to use the WITH_NAT function in kamailio.cfg? Thanks! Joe

4 5

SIP Ping utility with kamailio
by JR Richardson 16 Oct '10

16 Oct '10

Hi All, Can someone point me in the right direction of a command line SIP Ping utility or how to invoke from Kamailio? I see there is a sip_ping.pl script in voip-hacks, does anyone have copy-paste text version of that, all I can find is the PDF? Thanks. JR -- JR Richardson Engineering for the Masses

4 7

call for assistance
by Meftah Tayeb 16 Oct '10

16 Oct '10

hi folk, could you join irc irc.freenode.net #switchfin to start porting SR to BF? not exactly porting but integrating if you get there we will organise a conf call hosted by me i could provide SIP and evean PSTN access thank you -- Meftah Tayeb inum: +883510001288000 mobile: +213660347746

2 2

how can string "0" be equal to int 0?
by Juha Heinanen 15 Oct '10

15 Oct '10

i have stored in htable a key_value=0 with key_type=1, i.e., string. then i have this piece of test code: if (defined($sht(htable=>$var(carrier_key)::id))) { xlog("L_INFO", "$sht(htable=>$var(carrier_key)::id) is defined\n"); } else { xlog("L_INFO", "$sht(htable=>$var(carrier_key)::id) is NOT defined\n"); }; $var(carrier_id) = $sht(htable=>$var(carrier_key)::id); if ($var(carrier_id) == (int)0) { xlog("L_INFO", "$sht(htable=>$var(carrier_key)::id) is not found\n"); } else { xlog("L_INFO", "$sht(htable=>$var(carrier_key)::id) is found\n"); }; and when the piece is executed, this is printed to syslog: Oct 15 12:20:33 squeeze /usr/sbin/trunk-proxy[2915]: INFO: 0 is defined Oct 15 12:20:33 squeeze /usr/sbin/trunk-proxy[2915]: INFO: 0 is not found i understand the first line, because the entry is in the htable, but how is it possible that string "0" is equal to int 0? -- juha

4 25

upgrade from 3.0 to 3.1
by Anders 15 Oct '10

15 Oct '10

Hi, Sorry about the low-level question, but... - how do I upgrade from Kamailio 3.0 to 3.1? I used git for installing 3.0 (this great guide: http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.0.x-from-git) - but do I need to remove 3.0 and then install 3.1? Thanks, Anders

3 7

Re: [SR-Users] rtp proxy - where to invoke?
by Anders 15 Oct '10

15 Oct '10

Sorry, I didn't understand that. Is it good or bad that it is asymmetrical? And if not, what should it be? As far as I understand from my config, it first checks if NAT is set, and if so, it goes to force_rtp_proxy if it's INVITE and unforce_rtp_proxy if it's a BYE. Is that not correct? > On Wed, Oct 13, 2010 at 1:51 PM, Alex Balashov > <abalashov(a)evaristesys.com> wrote: >> This suggests asymmetrical invocation of rtpproxy, for example in 200 OK but >> not corresponding INVITE. >> >> -- >> Alex Balashov - Principal >> Evariste Systems LLC >> 1170 Peachtree Street >> 12th Floor, Suite 1200 >> Atlanta, GA 30309 >> Tel: +1-678-954-0670 >> Fax: +1-404-961-1892 >> Web: http://www.evaristesys.com/ >> >> On Oct 13, 2010, at 1:33 PM, Anders <vaerge(a)gmail.com> wrote: >> >>> Hi, >>> >>> I'm trying to get the RTP proxy to work with Kamailio (1.5), but right >>> now I am getting this error here: >>> >>> Oct 13 16:54:44 vn1031 /usr/local/sbin/kamailio[13622]: >>> ERROR:nathelper:force_rtp_proxy: incorrect port 0 in reply from rtp >>> proxy >>> >>> There is probably something wrong in my cfg - for the use of >>> force_rtp_proxy that part of my config is here below. See anything odd >>> that could lead to the error above? >>> >>> route { >>> ...... >>> route(4); >>> ...... >>> } >>> >>> route[4]{ >>> force_rport(); >>> if (nat_uac_test("19")) { >>> if (method=="REGISTER") { >>> fix_nated_register(); >>> } else { >>> fix_nated_contact(); >>> } >>> setflag(5); >>> } >>> return; >>> } >>> >>> route[5] { >>> if (is_method("BYE")) { >>> unforce_rtp_proxy(); >>> } else if (is_method("INVITE")){ >>> force_rtp_proxy(); >>> } >>> if (!has_totag()) add_rr_param(";nat=yes"); >>> return; >>> } >>> >>> onreply_route[1] { >>> xdbg("incoming reply\n"); >>> >>> if (!(status =~ "183" || status =~ "200")) >>> exit; >>> #fix_nated_contact(); >>> force_rtp_proxy(); >>> } >>> >>> _______________________________________________ >>> 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 4

getting unexpected values from htable
by Javier Gallart 15 Oct '10

15 Oct '10

Hi all I'm testing the htable module at Kamailio 3.1. Currently it's an extremely simple table with a single row loaded from a postgres DB. These are the relevant lines in the configuration file: modparam("htable", "db_url", "postgres://user:password@localhost :5432/database") modparam("htable", "htable", "a=>size=2;autoexpire=0;dbtable=h_table;") modparam("htable", "key_name_column", "key_name") modparam("htable", "key_type_column", "key_type") modparam("htable", "value_type_column", "value_type") modparam("htable", "key_value_column", "key_value") The only line in the table is: key_name | key_value | key_type | value_type ----------+-----------+----------+------------ 5 | 1 | 1 | 0 However, sht_print() shows this: 0(8570) ERROR: htable [ht_api.c:488]: ===== htable[a] hid: 109 exp: 0> 0(8570) ERROR: htable [ht_api.c:492]: htable[0] -- <0> 0(8570) ERROR: htable [ht_api.c:492]: htable[1] -- <0> 0(8570) ERROR: htable [ht_api.c:492]: htable[2] -- <0> 0(8570) ERROR: htable [ht_api.c:492]: htable[3] -- <1> 0(8570) ERROR: htable [ht_api.c:496]: cell: 5 0(8570) ERROR: htable [ht_api.c:498]: hid: 51 msize: 74 flags: 0 expire: 0 0(8570) ERROR: htable [ht_api.c:502]: v-i:9585128 Any idea of what might be happening? Thanks in advance.

3 3

Best of new in Kamailio 3.1
by Daniel-Constantin Mierla 15 Oct '10

15 Oct '10

Hello, some time ago I started a blog post series to reveal in more details some of the cool new features in version 3.1.0. Not finished yet, but has quite some posts so far, you can see the ToC at: http://bit.ly/c5jjzu Hope is useful for some of you, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

2 1

userblacklist error using characters
by "Nicolas Rüger" 15 Oct '10

15 Oct '10

Hello, HOW can I configure "userblacklist" module to accept the matching of characters as well??? I'm trying to use the "userblacklist" module with characters instead of numbers, so that I can even blacklist SIP-URIs like "user01(a)test.com". Unfortunately the "userblacklist" module or more specific "dtrie.c" returns the following error when I pass such a username: ERROR: <core> [dtrie.c:128]: cannot insert non-numerical character I checked in "dtrie.c" and found the following comment: "Its optimized towards the usecase of a matching tree that contains only digits, e.g.for LCR or blacklist modules. Nevertheless it also supports the matching of characters when configured correctly" So my question is: HOW can I configure "userblacklist" module to accept the matching of characters as well??? Thank you... Regards, Nicolas -- GMX DSL Doppel-Flat ab 19,99 €/mtl.! Jetzt auch mit gratis Notebook-Flat! http://portal.gmx.net/de/go/dsl

6 29

Re: [SR-Users] Integrate xmpp module with kamailio 3.0.1
by Huy Nguyen 15 Oct '10

15 Oct '10

Hello Daniel, > If i use the different domain for kamailio and xmpp (e.g kamailio1.com and xmpp1.com, i'm testing in my local network). If from my xmpp client, e.g 109(a)xmpp1.com, i chat to sip client, e.g 101*kamailio1.com(a)xmpp1.com. >> you still have same domain - xmpp1.com - for both. Maybe i don't understand clearly what you say yet ! I think i used 2 domain: kamailio server have domain: kamailio1.com (101(a)kamailio1.com), xmpp server have domain: xmpp1.com (109(a)xmpp1.com). Login to xmpp client 109(a)xmpp1.com, how can i chat to the kamailio client 101(a)kamailio1.com ??? If I add to my buddy list is: 101*kamailio1.com(a)xmpp1.com --> xmpp1 server think it is a local user ! I need your idea about this testing scenario. I'm using the server mode when integrate xmpp to kamailio. Thanks in advanced, Huy Nguyen www.htk-inc.com Date: Fri, 15 Oct 2010 10:55:29 +0200 From: miconda(a)gmail.com To: huy_quocnguyen(a)live.com Subject: Re: [SR-Users] Integrate xmpp module with kamailio 3.0.1 On 10/15/10 10:52 AM, Huy Nguyen wrote: Hi Daniel and list, If i use the different domain for kamailio and xmpp (e.g kamailio1.com and xmpp1.com, i'm testing in my local network). If from my xmpp client, e.g 109(a)xmpp1.com, i chat to sip client, e.g 101*kamailio1.com(a)xmpp1.com. you still have same domain - xmpp1.com - for both. So when your xmpp client sends to server, the server looks at domain part and says it is my domain so i should have it in my location table. If it is another domain, xmpp2.com, then xmpp server will connect to xmpp2.com to send the message there. Cheers, Daniel I think it'll get the previous problem, because the xmpp server saw that the domain of the customer is xmpp1.com --> this is a user on its domain. How can xmpp server recognize that is the sip user from kamailio server and send the message to kamailio ? Have any wrong in my testing scenario ? Or we must change the order of sip buddy name: 101*xmpp1.com(a)kamailio1.com ? Thanks, Huy Nguyen www.htk-inc.com Date: Fri, 15 Oct 2010 10:31:28 +0200 From: miconda(a)gmail.com To: huy_quocnguyen(a)live.com CC: sr-users(a)lists.sip-router.org Subject: Re: [SR-Users] Integrate xmpp module with kamailio 3.0.1 Hello, it is the same domain you use for both xmpp and sip server? You have to use different ones, otherwise each server considers it is a local user and try to look it up in own location database. Cheers, Daniel On 10/15/10 7:27 AM, Huy Nguyen wrote: Hi, Thanks for your quickly support, but after recompile with the lastest git branch 3.0 version, I still meet the same problem. From my xmpp client, i chat to sip client (e.g 101*sip.htk.com(a)xmpp.htk.com), the xmpp server looking for its database and not found user 101*sip.htk.com. So, it show the error 503. I don't know why the xmpp server not send message to my sip server ? At the current time, can we use the server mode of xmpp module ? If yes, how can establish the connection between xmpp module (on kamailio server) and the xmpp server, if the xmpp module not running at port 5269 ? Thanks, Huy Nguyen www.htk-inc.com Date: Thu, 14 Oct 2010 19:57:57 +0200 From: miconda(a)gmail.com To: huy_quocnguyen(a)live.com CC: sr-users(a)lists.sip-router.org Subject: Re: [SR-Users] Integrate xmpp module with kamailio 3.0.1 Hello, take the latest git version from branch kamailio_3.0. There were some fixes to xmpp module after 3.0.1. Cheers, Daniel On 10/14/10 7:41 PM, Huy Nguyen wrote: Hi all, I'm trying to integrate xmpp module to kamailio 3.0.1. But i wondered about the server mode, if not use the local xmpp server, how to establish the s2s connection with the other xmpp server ? I think it must have the running port 5269 on two xmpp servers to setup the connection. Thanks, Huy Nguyen www.htk-inc.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://www.asipto.com -- Daniel-Constantin Mierla http://www.asipto.com

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Re: [SR-Users] Integrate xmpp module with kamailio 3.0.1
by Daniel-Constantin Mierla 15 Oct '10

15 Oct '10

... reply all didn't work first time ... On 10/15/10 10:52 AM, Huy Nguyen wrote: > Hi Daniel and list, > > If i use the different domain for kamailio and xmpp (e.g kamailio1.com > and xmpp1.com, i'm testing in my local network). If from my xmpp > client, e.g 109(a)xmpp1.com, i chat to sip client, e.g > 101*kamailio1.com(a)xmpp1.com. you still have same domain - xmpp1.com - for both. So when your xmpp client sends to server, the server looks at domain part and says it is my domain so i should have it in my location table. If it is another domain, xmpp2.com, then xmpp server will connect to xmpp2.com to send the message there. Cheers, Daniel > I think it'll get the previous problem, because the xmpp server saw > that the domain of the customer is xmpp1.com --> this is a user on its > domain. How can xmpp server recognize that is the sip user from > kamailio server and send the message to kamailio ? > > Have any wrong in my testing scenario ? Or we must change the order of > sip buddy name: 101*xmpp1.com(a)kamailio1.com ? > > Thanks, > Huy Nguyen > > * > > * > ------------------------------------------------------------------------ > ------------------------------------------------------------------------ > www.htk-inc.com <http://www.htk-inc.com/> > > > > > > > ------------------------------------------------------------------------ > Date: Fri, 15 Oct 2010 10:31:28 +0200 > From: miconda(a)gmail.com > To: huy_quocnguyen(a)live.com > CC: sr-users(a)lists.sip-router.org > Subject: Re: [SR-Users] Integrate xmpp module with kamailio 3.0.1 > > Hello, > > it is the same domain you use for both xmpp and sip server? You have > to use different ones, otherwise each server considers it is a local > user and try to look it up in own location database. > > Cheers, > Daniel > > On 10/15/10 7:27 AM, Huy Nguyen wrote: > > Hi, > > Thanks for your quickly support, but after recompile with the > lastest git branch 3.0 version, I still meet the same problem. > > From my xmpp client, i chat to sip client (e.g > 101*sip.htk.com(a)xmpp.htk.com > <mailto:101*sip.htk.com@xmpp.htk.com>), the xmpp server looking > for its database and not found user 101*sip.htk.com. So, it show > the error 503. > > I don't know why the xmpp server not send message to my sip > server ? At the current time, can we use the server mode of xmpp > module ? If yes, how can establish the connection between xmpp > module (on kamailio server) and the xmpp server, if the xmpp > module not running at port 5269 ? > > Thanks, > Huy Nguyen > > * > > * > ------------------------------------------------------------------------ > ------------------------------------------------------------------------ > www.htk-inc.com <http://www.htk-inc.com/> > > > > > > > > ------------------------------------------------------------------------ > Date: Thu, 14 Oct 2010 19:57:57 +0200 > From: miconda(a)gmail.com <mailto:miconda@gmail.com> > To: huy_quocnguyen(a)live.com <mailto:huy_quocnguyen@live.com> > CC: sr-users(a)lists.sip-router.org > <mailto:sr-users@lists.sip-router.org> > Subject: Re: [SR-Users] Integrate xmpp module with kamailio 3.0.1 > > Hello, > > take the latest git version from branch kamailio_3.0. There were > some fixes to xmpp module after 3.0.1. > > Cheers, > Daniel > > On 10/14/10 7:41 PM, Huy Nguyen wrote: > > Hi all, > > I'm trying to integrate xmpp module to kamailio 3.0.1. > > But i wondered about the server mode, if not use the local > xmpp server, how to establish the s2s connection with the > other xmpp server ? I think it must have the running port 5269 > on two xmpp servers to setup the connection. > > Thanks, > Huy Nguyen > > * > > * > ------------------------------------------------------------------------ > ------------------------------------------------------------------------ > www.htk-inc.com <http://www.htk-inc.com/> > > > > > > > _______________________________________________ > 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 > > > _______________________________________________ > 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 -- Daniel-Constantin Mierla http://www.asipto.com

1 0

how can string "0" be equal to int 0?
by Juha Heinanen 15 Oct '10

15 Oct '10

i have stored in htable a key_value=0 with key_type=1, i.e., string. then i have this piece of test code: if (defined($sht(htable=>$var(carrier_key)::id))) { xlog("L_INFO", "$sht(htable=>$var(carrier_key)::id) is defined\n"); } else { xlog("L_INFO", "$sht(htable=>$var(carrier_key)::id) is NOT defined\n"); }; $var(carrier_id) = $sht(htable=>$var(carrier_key)::id); if ($var(carrier_id) == (int)0) { xlog("L_INFO", "$sht(htable=>$var(carrier_key)::id) is not found\n"); } else { xlog("L_INFO", "$sht(htable=>$var(carrier_key)::id) is found\n"); }; and when the piece is executed, this is printed to syslog: Oct 15 12:20:33 squeeze /usr/sbin/trunk-proxy[2915]: INFO: 0 is defined Oct 15 12:20:33 squeeze /usr/sbin/trunk-proxy[2915]: INFO: 0 is not found i understand the first line, because the entry is in the htable, but how is it possible that string "0" is equal to int 0? -- juha

1 0

Re: [SR-Users] Integrate xmpp module with kamailio 3.0.1
by Huy Nguyen 15 Oct '10

15 Oct '10

Hi Daniel and list, If i use the different domain for kamailio and xmpp (e.g kamailio1.com and xmpp1.com, i'm testing in my local network). If from my xmpp client, e.g 109(a)xmpp1.com, i chat to sip client, e.g 101*kamailio1.com(a)xmpp1.com. I think it'll get the previous problem, because the xmpp server saw that the domain of the customer is xmpp1.com --> this is a user on its domain. How can xmpp server recognize that is the sip user from kamailio server and send the message to kamailio ? Have any wrong in my testing scenario ? Or we must change the order of sip buddy name: 101*xmpp1.com(a)kamailio1.com ? Thanks, Huy Nguyen www.htk-inc.com Date: Fri, 15 Oct 2010 10:31:28 +0200 From: miconda(a)gmail.com To: huy_quocnguyen(a)live.com CC: sr-users(a)lists.sip-router.org Subject: Re: [SR-Users] Integrate xmpp module with kamailio 3.0.1 Hello, it is the same domain you use for both xmpp and sip server? You have to use different ones, otherwise each server considers it is a local user and try to look it up in own location database. Cheers, Daniel On 10/15/10 7:27 AM, Huy Nguyen wrote: Hi, Thanks for your quickly support, but after recompile with the lastest git branch 3.0 version, I still meet the same problem. From my xmpp client, i chat to sip client (e.g 101*sip.htk.com(a)xmpp.htk.com), the xmpp server looking for its database and not found user 101*sip.htk.com. So, it show the error 503. I don't know why the xmpp server not send message to my sip server ? At the current time, can we use the server mode of xmpp module ? If yes, how can establish the connection between xmpp module (on kamailio server) and the xmpp server, if the xmpp module not running at port 5269 ? Thanks, Huy Nguyen www.htk-inc.com Date: Thu, 14 Oct 2010 19:57:57 +0200 From: miconda(a)gmail.com To: huy_quocnguyen(a)live.com CC: sr-users(a)lists.sip-router.org Subject: Re: [SR-Users] Integrate xmpp module with kamailio 3.0.1 Hello, take the latest git version from branch kamailio_3.0. There were some fixes to xmpp module after 3.0.1. Cheers, Daniel On 10/14/10 7:41 PM, Huy Nguyen wrote: Hi all, I'm trying to integrate xmpp module to kamailio 3.0.1. But i wondered about the server mode, if not use the local xmpp server, how to establish the s2s connection with the other xmpp server ? I think it must have the running port 5269 on two xmpp servers to setup the connection. Thanks, Huy Nguyen www.htk-inc.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://www.asipto.com

1 0

Re: [SR-Users] Integrate xmpp module with kamailio 3.0.1
by Huy Nguyen 15 Oct '10

15 Oct '10

Hi, Thanks for your quickly support, but after recompile with the lastest git branch 3.0 version, I still meet the same problem. >From my xmpp client, i chat to sip client (e.g 101*sip.htk.com(a)xmpp.htk.com), the xmpp server looking for its database and not found user 101*sip.htk.com. So, it show the error 503. I don't know why the xmpp server not send message to my sip server ? At the current time, can we use the server mode of xmpp module ? If yes, how can establish the connection between xmpp module (on kamailio server) and the xmpp server, if the xmpp module not running at port 5269 ? Thanks, Huy Nguyen www.htk-inc.com Date: Thu, 14 Oct 2010 19:57:57 +0200 From: miconda(a)gmail.com To: huy_quocnguyen(a)live.com CC: sr-users(a)lists.sip-router.org Subject: Re: [SR-Users] Integrate xmpp module with kamailio 3.0.1 Hello, take the latest git version from branch kamailio_3.0. There were some fixes to xmpp module after 3.0.1. Cheers, Daniel On 10/14/10 7:41 PM, Huy Nguyen wrote: Hi all, I'm trying to integrate xmpp module to kamailio 3.0.1. But i wondered about the server mode, if not use the local xmpp server, how to establish the s2s connection with the other xmpp server ? I think it must have the running port 5269 on two xmpp servers to setup the connection. Thanks, Huy Nguyen www.htk-inc.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://www.asipto.com

2 1

Regd: Kamailio + Asterisk
by jack nicolson 15 Oct '10

15 Oct '10

Hi, I am trying to use kamailio as SIP server, so that it receive SIP invite and rewrite to asterisk, the set up is working fine for audio stuff, however for video it is not working properly, For rtsp stream I am getting no media found in asterisk log Could help me in it. Thanks Jack

2 3

Record-route issue in Kamailio 3.0.3
by Morten Isaksen 14 Oct '10

14 Oct '10

Hi, I think I have found a issue with recourd-route in Kamailio 3.0.3. My old setup was: Microsoft OCS <--> (TCP) OpenSER (UDP) <---> (UDP) Mediant 2000 (ISDN). That worked fine. Now I have inserted a new server in this setup. Microsoft OCS <--> (TCP) OpenSER (UDP) <---> (UDP) Kamailio (UDP) <--> (UDP) Mediant 2000 (ISDN). When OpenSer sends the message to Kamailio the recourd-routes look like this: Record-Route: <sip:x.x.248.20;r2=on;lr;ftag=3d9e7d131b> Record-Route: <sip:x.x.248.20;transport=tcp;r2=on;lr;ftag=3d9e7d131b> But when the message comes back from Kamailio it is: Record-Route: <sip:x.x.248.7;lr=on;ftag=3d9e7d131b;did=584.1a683a45>,<sip:x.x.248.20;r2=on;lr;ftag=3d9e7d131b>,<sip:x.x.248.20;transport=tcp;r2=on;lr;ftag=3d9e7d131b> OpenSER forwards the message to the OCS with Record-route unchanged and the OCS gets confused and does not reply. I am not that strong in the SIP RFC's. Is it part of the SIP standard to compact the record-route into one line? Is it a bug in Kamailio or is it just a parameter that needs to be changed? Any help will be much appreciated. -- Morten Isaksen

4 7

Integrate xmpp module with kamailio 3.0.1
by Huy Nguyen 14 Oct '10

14 Oct '10

Hi all, I'm trying to integrate xmpp module to kamailio 3.0.1. But i wondered about the server mode, if not use the local xmpp server, how to establish the s2s connection with the other xmpp server ? I think it must have the running port 5269 on two xmpp servers to setup the connection. Thanks, Huy Nguyen www.htk-inc.com

2 1

where to look
by Anders 14 Oct '10

14 Oct '10

Hi, I was wondering if a message like this below in the syslog contains any lead as to where to look for the problem. The [23582] is a lead to where the problem occured? Oct 14 13:34:52 server1 /usr/local/sbin/kamailio[23582]: ERROR:db_mysql:db_mysql_submit_query: driver error on query: Unknown column 'xxxxx' in 'where clause' And concerning this specific issue, where I put 'xxxxx' appears the src_user. Problem in kamailio.cfg, right? Thanks! //Anders

2 3

redirect on register
by Nick Olkin 14 Oct '10

14 Oct '10

Hi all! I need to implement a kind of Location Server. The idea behind is to process the REGISTER messages and analyze the source IP address of the packet and reply with the "302 Moved Temporarily" message. What I want to achieve - provide the SIP client with the "nearest" point to use for registration and further communications. The possible approach is: 1. make some list of available registrars/SBC's mapped to appropriate IP (sub)networks. 2. configure SER to process REGISTER messages and reply with "302 Moved Temporarily" having the IP or name of "nearest" point in Contact field. Does anybody have any ideas on possible implementations (with SER or perhaps with some other tools)? Thanks a lot! PS. At this moment I can certainly send "302 Moved Temporarily" on REGISTER with SER, but I have no idea on how to insert correct Contact header, so my SIP client just stops registering.

1 0

Re: [SR-Users] Regarding SER Listening on two ports
by Saurabh Srivastava 14 Oct '10

14 Oct '10

Hello, We are facing a problem after doing the changes that you said to do in the configuration file. Now we are able to get the server listening on two ports such as 5060 & 5062 but now the problem is that the user agent which is registered from 5060 is able to communicate with the other user agent registered to 5060 and the same is true for the other port 5062 (i.e. the user agent which is registered from 5062 is able to communicate with the other user agent registered to 5062.). But there is no communication happening in case of one user is registered to 5060 and the other user is registered through 5062. Please help us in this regard. Thanks and Best Regards Saurabh Srivastava On Wed, Oct 13, 2010 at 5:59 PM, Saurabh Srivastava <saurabh(a)atc-labs.com>wrote: > Hello Andrei, > > Wwhere do we need to add this line > > "listen=eth0:5060 eth0:5062" > > in the ser.cfg file or somewhere else. Please guide me through the steps or > if it is possible can you please attach an example ser.cfg for listening on > 2 ports. > > Thanks and Best Regards > Saurabh Srivastava > > On Fri, Oct 8, 2010 at 1:28 PM, Andrei Pelinescu-Onciul < > andrei(a)iptel.org> wrote: > >> On Oct 06, 2010 at 11:43, Saurabh Srivastava <saurabh(a)atc-labs.com> >> wrote: >> > Hello SER Team, >> > >> > We are using SER as a SIP Server for our purpose. We wan to configure >> SER to >> > listen on two ports at the same time(i.e. 5060 & some other custom port >> such >> > as 5090 or something like that). Can you please assist us how can we >> achieve >> > this, means what changes we need to make such that SER listens on two >> ports >> > at the same time. >> >> Add >> listen=eth0:5060 eth0:5062 >> >> The generic format is: listen=[proto:]ip_hostname_or_interface[:port]. >> >> You can also do it form the command line. e.g.: >> ser -f yourconfig -l 127.0.0.1:5060 -l 127.0.0.1:5062 >> >> Andre > >

1 0

Use Kamailio
by michel freiha 14 Oct '10

14 Oct '10

*Dear All, I'm planning to use Kamailio in a high network traffic environment... I'm sure that it can handle several thousand of calls when using SIP over UDP but I have a doubt regarding TLS/TCPusage with Kamailio...I saw on the website that the latest version fixed that BUG that I had bad experience with and Kamailio is ready now for using TLS over TCP with no problem... My question is, how much TLS/TCP calls Kamailio can handle and does anyone had an experience with it? Regards*

2 1

forward sip register
by Borin 14 Oct '10

14 Oct '10

Hello, I am not good at kamailio. Could anyone tell me how can I forward register messages coming to kamailio to asterisk? I need users to be registered on kamailio and on asterisk at one time. I do in kamailio.cfg: if (is_method("REGISTER")) { save("location"); route(FWD_REG); exit; } route[FWD_REG] { rewritehost("Asterisk_IP:5060"); if(!t_relay()) sl_reply_error(); exit; } But it (kamailio ver 3.1) does not forward them.

3 9

[OT] Local root exploit for the x86_64 Linux kernel ia32syscall emulation vulnerability
by Iñaki Baz Castillo 14 Oct '10

14 Oct '10

In any Linux Kernel 64 bits < 2.6.36-rc4: -------------------------------------------- ibc@myserver :/tmp$ whoami ibc ibc@myserver:/tmp$ wget http://packetstormsecurity.org/1009-exploits/robert_you_suck.c ibc@myserver:/tmp$ gcc -o putada robert_you_suck.c ibc@myserver:/tmp$ ./putada resolved symbol commit_creds to 0xffffffff81092120 resolved symbol prepare_kernel_cred to 0xffffffff81091fa0 mapping at 3f80000000 UID 0, EUID:0 GID:0, EGID:0 sh-3.2# whoami root <----------- OPSSSS !!! -------------------------------------------- More info: http://packetstormsecurity.org/filedesc/robert_you_suck.c.html Fixed in Debian (kernel patch backported): http://security-tracker.debian.org/tracker/CVE-2010-3081 - lenny (security) 2.6.26-25lenny1 fixed - lenny-backports 2.6.32-23~bpo50+1 fixed - squeeze 2.6.32-23 fixed -- Iñaki Baz Castillo <ibc(a)aliax.net>

1 0

Best ASR with Kamailio
by Yury Naumovich 13 Oct '10

13 Oct '10

Good day. How with Kamailio improve ASR to ~100%, if we have 3-4 wholesale Voip providers? Each can guaranty only ASR 60-70%. Best regards, Yury

2 1

rtp proxy - where to invoke?
by Anders 13 Oct '10

13 Oct '10

Hi, I'm trying to get the RTP proxy to work with Kamailio (1.5), but right now I am getting this error here: Oct 13 16:54:44 vn1031 /usr/local/sbin/kamailio[13622]: ERROR:nathelper:force_rtp_proxy: incorrect port 0 in reply from rtp proxy There is probably something wrong in my cfg - for the use of force_rtp_proxy that part of my config is here below. See anything odd that could lead to the error above? route { ...... route(4); ...... } route[4]{ force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); } setflag(5); } return; } route[5] { if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy(); } if (!has_totag()) add_rr_param(";nat=yes"); return; } onreply_route[1] { xdbg("incoming reply\n"); if (!(status =~ "183" || status =~ "200")) exit; #fix_nated_contact(); force_rtp_proxy(); }

2 1

core:parse error
by Anders 13 Oct '10

13 Oct '10

Hi, I'm getting a bunch of these sequences in my syslog - and have absolutely no idea where to look for the problem. Any pointers, please? Oct 13 08:12:34 vn1031 /usr/local/sbin/kamailio[2821]: ERROR:core:parse_cseq: expecting CSeq EoL Oct 13 08:12:34 vn1031 /usr/local/sbin/kamailio[2821]: ERROR:core:parse_cseq: bad cseq Oct 13 08:12:34 vn1031 /usr/local/sbin/kamailio[2821]: ERROR:core:get_hdr_field: bad cseq Oct 13 08:12:34 vn1031 /usr/local/sbin/kamailio[2821]: ERROR:maxfwd:is_maxfwd_present: parsing MAX_FORWARD header failed! Thanks!! --Anders

2 1

kamailio-3.1.0+lenny3 + siptrace
by MÉSZÁROS Mihály 13 Oct '10

13 Oct '10

Hello all! I am getting the following problem with 3.1.0 kamailio: ERROR: <core> [sr_module.c:875]: init_mod(): Error while initializing module siptrace (/usr/lib/kamailio/modules_k/siptrace.so) config: loadmodule "siptrace.so" # ----- siptrace params ----- modparam("siptrace", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("siptrace", "table", "sip_trace") modparam("siptrace", "trace_flag", 22) modparam("siptrace", "traced_user_avp", "$avp(s:user)") modparam("siptrace", "trace_on", 1) modparam("siptrace", "trace_sl_acks", 1) modparam("siptrace", "trace_local_ip", "10.1.1.1:5064") Any help appreciated. Best Regards, Misi

3 15

Kamailio Scalability
by michel freiha 13 Oct '10

13 Oct '10

*Dear All, I'm planning to use Kamailio in a high network traffic environment... I'm sure that it can handle several thousand of calls when using SIP over UDP but I have a doubt regarding TLS/TCPusage with Kamailio...I saw on the website that the latest version fixed that BUG that I had bad experience with and Kamailio is ready now for using TLS over TCP with no problem... My question is, how much TLS/TCP calls Kamailio can handle and does anyone had an experience with it? Regards*

1 0

mtree memory comsumption question
by anthony thomas 13 Oct '10

13 Oct '10

Hello list, I have been testing the new mtree module but I am a little bit confused about the results. On my tests, I randomly generated 500K, 1M and 2M entries in a table in several trees. The size of the db (on disk) ranges from 120 megs to 270 megs and when Kamailio loads it in memory, it jumps from 1.0Gb to 2.8Gb. Is there any way to pre-calculate the total amount of memory that a tree will need? Thanks, A

2 1

Need some help with showing one of my ITSPs that they have a problem.
by Geoffrey Mina 13 Oct '10

13 Oct '10

Hello, I am currently going through an Inter-Op test with a new carrier and there have been some issues. I have my Kamailio server integrated with many tier-1 carriers, so I'm pretty sure I'm doing everything OK. The problem is in the BYE coming from the ITSP, the URI isn't valid, but I am not sure based on RFC3261 exactly why it isn't correct. If anyone can take a quick glance, I would really appreciate it. Thanks! >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>> OUT >>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> INVITE sip:16782247008@67.150.160.10:5060 SIP/2.0 Record-Route: <sip:X.X.X.90;lr;ftag=as4c62e0b9> Via: SIP/2.0/UDP X.X.X.90;branch=z9hG4bK2aa.c3961912.0 Via: SIP/2.0/UDP X.X.X.238:5060;received=X.X.X.238;branch=z9hG4bK10281a99;rport=5060 From: ""2026097878"" <sip:2026097878@mydomain.com<sip%3A2026097878(a)mydomain.com> >;tag=as4c62e0b9 To: <sip:+16782247008@X.X.X.90:5060> Contact: <sip:2026097878@X.X.X.238> Call-ID: 4199b75502736b01448ce4bf04876d7e(a)mydomain.com CSeq: 102 INVITE User-Agent: G-Tel v1.0 Max-Forwards: 69 Date: Mon, 11 Oct 2010 17:50:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces P-Charge-Info: ""4019921313"" <sip:4019921313@connectfirst.com<sip%3A4019921313(a)connectfirst.com> > Content-Type: application/sdp Content-Length: 244 v=0 o=root 10137 10137 IN IP4 X.X.X.238 s=session c=IN IP4 X.X.X.238 t=0 0 m=audio 11748 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< <<<<<<<<<<<<<<< IN <<<<<<<<<<<<<<<<<<<<<<<< <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< SIP/2.0 100 Trying Via: SIP/2.0/UDP X.X.X.90;branch=z9hG4bK2aa.c3961912.0 Via: SIP/2.0/UDP X.X.X.238:5060;received=X.X.X.238;branch=z9hG4bK10281a99;rport=5060 Record-Route: <sip:X.X.X.90;lr;ftag=as4c62e0b9> From: ""2026097878"" <sip:2026097878@mydomain.com<sip%3A2026097878(a)mydomain.com> >;tag=as4c62e0b9 To: <sip:+16782247008@X.X.X.90:5060> Call-ID: 4199b75502736b01448ce4bf04876d7e(a)mydomain.com CSeq: 102 INVITE Contact: <sip:16782247008@67.150.160.10 <sip%3A16782247008(a)67.150.160.10>> Content-Length: 0 <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< <<<<<<<<<<<<<<< IN <<<<<<<<<<<<<<<<<<<<<<<< <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< SIP/2.0 183 Session Progress Via: SIP/2.0/UDP X.X.X.90;branch=z9hG4bK2aa.c3961912.0 Via: SIP/2.0/UDP X.X.X.238:5060;received=X.X.X.238;branch=z9hG4bK10281a99;rport=5060 Record-Route: <sip:X.X.X.90;lr;ftag=as4c62e0b9> From: ""2026097878"" <sip:2026097878@mydomain.com<sip%3A2026097878(a)mydomain.com> >;tag=as4c62e0b9 To: <sip:+16782247008@X.X.X.90:5060>;tag=H3jDp3KZmHNja Call-ID: 4199b75502736b01448ce4bf04876d7e(a)mydomain.com CSeq: 102 INVITE Contact: <sip:16782247008@67.150.160.10 <sip%3A16782247008(a)67.150.160.10>> Content-Type: application/sdp Content-Length: 109 v=0 o=ATSI-B1 0 0 IN IP4 67.150.160.10 s=sip call c=IN IP4 67.150.160.10 t=0 0 m=audio 17308 RTP/AVP 0 <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< <<<<<<<<<<<<<<< IN <<<<<<<<<<<<<<<<<<<<<<<< <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< SIP/2.0 200 OK Via: SIP/2.0/UDP X.X.X.90;branch=z9hG4bK2aa.c3961912.0 Via: SIP/2.0/UDP X.X.X.238:5060;received=X.X.X.238;branch=z9hG4bK10281a99;rport=5060 Record-Route: <sip:X.X.X.90;lr;ftag=as4c62e0b9> From: ""2026097878"" <sip:2026097878@mydomain.com<sip%3A2026097878(a)mydomain.com> >;tag=as4c62e0b9 To: <sip:+16782247008@X.X.X.90:5060>;tag=H3jDp3KZmHNja Call-ID: 4199b75502736b01448ce4bf04876d7e(a)mydomain.com CSeq: 102 INVITE Contact: <sip:16782247008@67.150.160.10 <sip%3A16782247008(a)67.150.160.10>> Content-Type: application/sdp Content-Length: 109 v=0 o=ATSI-B1 0 0 IN IP4 67.150.160.10 s=sip call c=IN IP4 67.150.160.10 t=0 0 m=audio 17308 RTP/AVP 0 >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>> OUT >>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> ACK sip:16782247008@67.150.160.10 <sip%3A16782247008(a)67.150.160.10> SIP/2.0 Via: SIP/2.0/UDP X.X.X.238:5060;branch=z9hG4bK60b74898;rport Route: <sip:X.X.X.90;lr;ftag=as4c62e0b9> From: ""2026097878"" <sip:2026097878@mydomain.com<sip%3A2026097878(a)mydomain.com> >;tag=as4c62e0b9 To: <sip:+16782247008@X.X.X.90:5060>;tag=H3jDp3KZmHNja Contact: <sip:2026097878@X.X.X.238> Call-ID: 4199b75502736b01448ce4bf04876d7e(a)mydomain.com CSeq: 102 ACK User-Agent: G-Tel v1.0 Max-Forwards: 70 Content-Length: 0 <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< <<<<<<<<<<<<<<< IN <<<<<<<<<<<<<<<<<<<<<<<< <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< BYE sip:X.X.X.90;lr;ftag=as4c62e0b9 SIP/2.0 Via: SIP/2.0/UDP 67.150.160.10;branch=z9hG4bKvNBHaK6ZByQmg Max-Forwards: 70 From: <sip:+16782247008@X.X.X.90:5060>;tag=H3jDp3KZmHNja To: ""2026097878"" <sip:2026097878@mydomain.com<sip%3A2026097878(a)mydomain.com> >;tag=as4c62e0b9 Call-ID: 4199b75502736b01448ce4bf04876d7e(a)mydomain.com CSeq: 102 BYE Content-Length: 0 >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>> OUT >>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> SIP/2.0 404 Not here Via: SIP/2.0/UDP 67.150.160.10;branch=z9hG4bKvNBHaK6ZByQmg From: <sip:+16782247008@X.X.X.90:5060>;tag=H3jDp3KZmHNja To: ""2026097878"" <sip:2026097878@mydomain.com<sip%3A2026097878(a)mydomain.com> >;tag=as4c62e0b9 Call-ID: 4199b75502736b01448ce4bf04876d7e(a)mydomain.com CSeq: 102 BYE Server: Kamailio Content-Length: 0

2 1

failure_route does not fire up after
by Mino Haluz 12 Oct '10

12 Oct '10

Hi, I've been struggling with this problem very long time and I am starting to be quite desperate. I'm trying to implement voicemail redirection after 15 secs of ringing. So I set fr_inv_timer, set the event handler with t_on_failure. But after 15secs of ringing the failture_route block does not execute at all, the transaction is lost and after these 15 secs it has strange behaviour (the phone is still ringing, even after the other side has cancelled the call). I'm attaching the configuration without sensitive data. Please tell me whether problem is in the configuration or somewhere outside of kamailio. Thanks for any suggestion. #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_call ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif # ----------------------------------------------------------------------------------------- #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_NAT # ----------------------------------------------------------------------------------------- ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL7 fork=yes #fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ #auto_aliases=no /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ listen=udp:1.1.1.1:5060 # ----------------------------------------------------------------------------------------- alias=test # ----------------------------------------------------------------------------------------- port=5060 ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # ####### Modules Section ######## #set module path #mpath="/usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/" mpath="/lib/kamailio/modules_k/:/lib/kamailio/modules/" /* uncomment next line for MySQL DB support */ #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "uri_db.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "mi_rpc.so" loadmodule "acc.so" # Addition modules # ----------------------------------------------------------------------------------------- loadmodule "avpops.so" loadmodule "regex.so" loadmodule "sqlops.so" loadmodule "dialog.so" loadmodule "call_control.so" loadmodule "mediaproxy.so" loadmodule "siptrace.so" loadmodule "mi_datagram.so" loadmodule "group.so" loadmodule "permissions.so" # ----------------------------------------------------------------------------------------- modparam("mi_datagram", "socket_name", "/var/run/kamailio/kamailio.sock") # The name of a UNIX SOCKET used by Callcontrol modparam("dialog", "dlg_flag", 4) #modparam("call_control", "disable", 0) #modparam("call_control", "signaling_ip_avp", "$avp(s:signaling_ip)") # ----------------------------------------------------------------------------------------- modparam("sqlops","sqlcon","ca=>mysql:/") # The definition of a DB connection modparam("mediaproxy", "mediaproxy_socket", "/var/run/mediaproxy/dispatcher.sock") # Path to the filesystem socket where the mediaproxy dispatcher listens for commands # ----------------------------------------------------------------------------------------- modparam("tm", "ruri_matching", 1) modparam("tm", "restart_fr_on_each_reply", 0) # Final response timer will be re-triggered for each received provisional reply #modparam("tm", "fr_timer", 15000) # Timer which hits if no final reply for an INVITE arrives (15 sec) modparam("tm", "fr_inv_timer", 15000) # Timer which hits if no final reply for an INVITE arrives (15 sec) # ----------------------------------------------------------------------------------------- modparam("siptrace", "db_url", "") # Database URL modparam("siptrace", "traced_user_avp", "$avp(s:traced_user)") modparam("siptrace", "trace_on", 1) modparam("siptrace", "trace_flag", 22) # Flag is used to mark messages to trace # ----------------------------------------------------------------------------------------- # Radius settings modparam("acc", "failed_transaction_flag", 1) modparam("acc", "report_cancels", 0) modparam("acc", "report_ack", 0) modparam("acc", "early_media", 0) modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 1) modparam("acc", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") # This is the location of the configuration file of radius client modparam("acc", "radius_flag", 1) modparam("acc", "radius_missed_flag", 1) modparam("acc", "radius_extra", "User-Name=$Au; \ Calling-Station-Id=$from; \ Called-Station-Id=$to; \ Sip-Translated-Request-URI=$avp(s:translated_uri); \ Sip-RPid=$avp(s:rpid); \ Source-IP=$si; \ Source-Port=$sp; \ Canonical-URI=$avp(s:can_uri); \ Billing-Party=$avp(s:billing_party); \ Divert-Reason=$avp(s:divert_reason); \ X-RTP-Stat=$hdr(X-RTP-Stat); \ Contact=$hdr(contact); \ Event=$hdr(event); \ SIP-Proxy-IP=$avp(s:sip_proxy_ip); \ ENUM-TLD=$avp(s:enum_tld)") # ----------------------------------------------------------------------------------------- #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!endif /* uncomment next line for aliases support NOTE: a DB (like db_mysql) module must be also loaded */ #loadmodule "alias_db.so" /* uncomment next line for multi-domain support NOTE: a DB (like db_mysql) module must be also loaded NOTE: be sure and enable multi-domain support in all used modules (see "multi-module params" section ) */ loadmodule "domain.so" #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" #!endif loadmodule "uac.so" # ----------------- setting module-specific parameters --------------- # --- enable CLI with append_rpid_hf --- modparam("auth", "rpid_prefix", "<sip:") modparam("auth", "rpid_suffix", "@1.1.1.1;user=phone>;party=calling; screen=yes; privacy=off") #modparam("auth", "rpid_avp", "$avp(s:rpid)") # ----- mi_fifo params ----- # ----------------------------------------------------------------------------------------- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----------------------------------------------------------------------------------------- # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 1) # ----- rr params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # ----- uri_db params ----- /* by default we disable the DB support in the module as we do not need it in this configuration */ modparam("uri_db", "use_uri_table", 0) modparam("uri_db", "db_url", "mysql://") modparam("domain", "db_url", "mysql://") # ----- acc params ----- /* what sepcial events should be accounted ? */ modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 2) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) modparam("acc", "db_url", "mysql://") modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", "mysql://") #!endif # ----- auth_db params ----- /* enable the DB based authentication */ #!ifdef WITH_AUTH modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "db_url", "mysql://") modparam("auth_db", "load_credentials", "rpid") #!endif # -- group params -- /* for grp mysql table usage - groups - siptrace, blocked*/ modparam("group", "db_url", "mysql://") modparam("group", "table", "grp") modparam("group", "user_column", "username") modparam("group", "domain_column", "domain") modparam("group", "group_column", "grp") # -- permissions params -- /* for trusted table usage and other functions - other modparam are default */ modparam("permissions", "db_url", "mysql://") modparam("permissions", "db_mode", 1) # ----- alias_db params ----- /* uncomment the following lines if you want to enable the DB based aliases */ #modparam("alias_db", "db_url", # "mysql://") # ----- domain params ----- /* uncomment the following lines to enable multi-domain detection support */ #modparam("domain", "db_url", # "mysql://") #modparam("domain", "db_mode", 1) # Use caching # ----- multi-module params ----- /* uncomment the following line if you want to enable multi-domain support in the modules (dafault off) */ #modparam("alias_db|auth_db|usrloc|uri_db", "use_domain", 1) # ----- presence params ----- /* enable presence server support */ #!ifdef WITH_PRESENCE modparam("presence|presence_xml", "db_url", "mysql://") modparam("presence_xml", "force_active", 1) modparam("presence", "server_address", "sip:10.0.0.10:5060") #!endif # ----- nathelper ----- #!ifdef WITH_NAT #modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1:7722") #modparam("nathelper", "natping_interval", 30) #modparam("nathelper", "ping_nated_only", 1) #modparam("nathelper", "sipping_bflag", 7) #modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org<sip%3Apinger(a)kamailio.org> ") modparam("registrar|nathelper", "received_avp", "$avp(i:80)") modparam("usrloc", "nat_bflag", 6) #!endif ####### Routing Logic ######## # main request routing logic route{ #------------SANITY CHECK---------------------------------------------------------------- xlog("L_INFO","XLOG ****************START*($rm)*S_IP($si)*CallID($ci)*********************"); if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); xlog("L_INFO","XLOG: Too many hops man"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } if(!sanity_check("1511", "7")) { xlog("L_INFO","XLOG: Malformed SIP message"); xlog("Malformed SIP message from $si:$sp\n"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } # NAT detection xlog("L_INFO", "XLOG: Performing NAT detection"); route(NAT); if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { xlog("L_INFO", "XLOG: loose_route()"); if (is_method("BYE")) { xlog('L_INFO', 'XLOG: Method is BYE'); setflag(1); # do accounting ... setflag(3); # ... even if the transaction fails # end_media_session(); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { xlog('L_INFO', 'XLOG: Method is SUBSCRIBE and uri == myself'); # in-dialog subscribe requests route(PRESENCE); #!ifndef WITH_PRESENCE #added by yenkee sl_send_reply("403", "Not Allowed"); #!endif xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } if ( is_method("ACK") ) { xlog('L_INFO', 'XLOG: Method is ACK'); if ( t_check_trans() ) { # non loose-route, but stateful ACK; must be an ACK after a 487 or e.g. 404 from upstream server xlog('L_INFO', 'XLOG: t_check_trans()'); t_relay(); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } else { xlog('L_INFO', 'XLOG: ACK without matching transaction ... ignore and discard.'); #added by yenkee sl_send_reply("403", "Not Allowed"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } } sl_send_reply("404","Not here"); xlog("L_INFO", "XLOG: 404 Not Here ----------------------------------"); } xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } #initial requests # CANCEL processing if (is_method("CANCEL")) { xlog('L_INFO', 'XLOG: Method is CANCEL'); if (t_check_trans()) t_relay(); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } t_check_trans(); # authentication route(AUTH); setflag(1); # 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")) { xlog("L_INFO","XLOG: Method is INVITE"); setflag(1); # do accounting # ----------------------------------------------------------------------------------------- $avp(s:can_uri) = $ru; # Canonical-URI will be used for rating the session xlog("L_INFO", "XLOG: Engaging Media Proxy"); engage_media_proxy(); # Automatically call use_media_proxy / end_media_proxy on every request and reply that belongs to the dialog # ----------------------------------------------------------------------------------------- } if (!uri==myself) /* replace with following line if multi-domain support is used */ ##if (!is_uri_host_local()) { xlog("L_INFO","XLOG: Appeding Hf P-hing: outbound"); append_hf("P-hint: outbound\r\n"); route(RELAY); } # requests for my domain if( is_method("PUBLISH|SUBSCRIBE")) route(PRESENCE); if (is_method("REGISTER")) { if(isflagset(5)) { setbflag("6"); # uncomment next line to do SIP NAT pinging ## setbflag("7"); } if (!save("location")) sl_reply_error(); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } # ---------ENABLING SIP TRACE-------------------------------------------------------------- # Enable sip tracing for users in grp siptrace in mysql db ( nema vplyv na zapisovanie acc informacii ) if (is_user_in("From", "siptrace") ) { xlog("L_INFO","XLOG: Starting sip_trace for From-URI:($fu)"); setflag(22); # sip_trace(); # mne sip_trace nefungoval iba setflag , ak sa nastavi flag na hodnotu v modparam bude robit cdr tool sip tracy } # route(PSTN); # ----------------------------------------------------------------------------------------- # apply DB based aliases (uncomment to enable) ##alias_db_lookup("dbaliases"); if (is_method("INVITE")) { #------------BLOCK NUMBER----------------------------------------------------------------- # Block username if ( is_user_in("From", "blocked")) { xlog("L_INFO", "XLOG: [main route] Account ($fu) to ($ru) is blocked"); sl_send_reply("403", " Account blocked "); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } # ----------------------------------------------------------------------------------------- # Call control allows one to limit the duration of calls and automatically end them when they exceed the imposed limit xlog("L_INFO","XLOG: Before call_control() var rm = ($rm)"); call_control(); switch ($retcode) { case 2: # Call with no limit xlog("L_INFO", "XLOG: [call_control] Call with no limit"); break; case 1: # Call with a limit under callcontrol management (either prepaid or postpaid) xlog("L_INFO", "XLOG: [call_control] Call with a limit under callcontrol management"); break; case -1: # Not enough credit (prepaid call) xlog("L_INFO", "XLOG: Call control: not enough credit for prepaid call"); acc_rad_request("402"); sl_send_reply("402", "Not enough credit"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; break; case -2: # Locked by call in progress (prepaid call) xlog("L_INFO", "XLOG: Call control: prepaid call locked by another call in progress"); acc_rad_request("403"); sl_send_reply("403", "Call locked by another call in progress"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; break; default: # Internal error (message parsing, communication, ...) xlog("L_INFO", "XLOG: Call control: internal server error"); acc_rad_request("500"); sl_send_reply("500", "Internal server error"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } # ----------------------------------------------------------------------------------------- # ----------------------------------------------------------------------------------------- # Return error message when you are calling on your own number # fU - From username # tU - To username if ($fU==$tU) { sl_send_reply("403","Forbidden calling own number"); xlog("L_INFO","XLOG: Call on your own number: Request-URI:($ru) From-URI:($fu). Sending 403 Forbidden"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } # ----------------------------------------------------------------------------------------- } # ----------------------------------------------------------------------------------------- # ----------------------------------------------------------------------------------------- # ----------------------------------------------------------------------------------------- # ----------------------------------------------------------------------------------------- xlog("L_INFO","XLOG: Performing lookup( location) "); if (!lookup("location")) { switch ($rc) { case -1: case -3: t_newtran(); t_reply("404", "[local failed] Not Found"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; break; # commented - not used with PSTN case -2: sl_send_reply("405", "Method Not Allowed"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } # ----------------------------------------------------------------------------------------- # ----------------------------------------------------------------------------------------- # ----------------------------------------------------------------------------------------- # Replacing plus sign with 00 if ($ru=~".*sip:\+.*@.*") { # Check if URI contains plus sign $avp(s:to)=$ru; avp_subst("$avp(s:to)", "/(sip:)\+/sip:00/"); # Replace plus sign with 00 avp_pushto("$ru", "$avp(s:to)"); # Push the value of AVP with name into the SIP message as Request-URI xlog("L_INFO","XLOG: Replacing plus sign in Request-URI ($ru)"); } # ----------------------------------------------------------------------------------------- # ----------------------------------------------------------------------------------------- } # when routing via usrloc, log the missed calls also setflag(2); route(RELAY); } route[RELAY] { xlog("L_INFO", "XLOG: [RELAY] Jumped to route(RELAY)"); #!ifdef WITH_NAT if (check_route_param("nat=yes")) { xlog("L_INFO","XLOG: [RELAY] check_route_param('nat=yes') == 1"); setbflag("6"); } if (isflagset(5) || isbflagset("6")) { xlog("L_INFO","XLOG: [RELAY] isflagset(5) && isbflagset(6)"); route(RTPPROXY); } #!endif # ----------------------------------------------------------------------------------------- /* example how to enable some additional event routes */ if (is_method("INVITE")) { # $avp(s:callee_fr_inv_timer) = 10; xlog("L_INFO", "XLOG: [RELAY] Routing of failure and reply messages"); # t_on_branch("BRANCH_ONE"); t_set_fr(10000); t_on_reply("REPLY_ONE"); t_on_failure("1"); # Registering failure route } if (!t_relay()) { sl_reply_error(); } xlog("L_INFO","XLOG: exiting FROM RELAY ------------------------------------"); exit; } # Presence server route route[PRESENCE] { #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } return; } # Authentication route route[AUTH] { xlog("L_INFO", "XLOG: Jumped to route[AUTH]"); #!ifdef WITH_AUTH if (is_method("REGISTER")) { xlog("L_INFO","XLOG: [AUTH] Register pred ifom From-URI:($fu)"); # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("", "subscriber")) { xlog("L_INFO", "XLOG: [AUTH] www_challenge()"); www_challenge("", "0"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } if ($au!=$tU) { xlog("L_INFO", "XLOG: [AUTH] Forbidden auth ID From-URI:($fu)"); sl_send_reply("403","Forbidden auth ID"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } } else { # authenticate if from local subscriber (uncomment to enable auth) if (from_uri==myself) { xlog("L_INFO","XLOG: [AUTH] Message is not REGISTER, and from_uri == myself"); if (!proxy_authorize("", "subscriber")) { xlog("L_INFO", "XLOG: [AUTH] proxy_challenge()"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); proxy_challenge("", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } } xlog("L_INFO","XLOG: [AUTH] consumer_credentials()"); consume_credentials(); # caller authenticated } } #!endif return; } # Caller NAT detection route route[NAT]{ #!ifdef WITH_NAT force_rport(); xlog("L_INFO", "XLOG: Checking nat_uac_test('19')"); if (nat_uac_test("19")) { if (method=="REGISTER") { xlog("L_INFO", "XLOG: fix_nated_register()"); fix_nated_register(); } else { xlog("L_INFO", "XLOG: fix_nated_contact()"); fix_nated_contact(); } setflag(5); } #!endif return; } # RTPProxy control route[RTPPROXY] { xlog("L_INFO","XLOG: Jumped to 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"); xlog("L_INFO","XLOG: Adding RR param(';nat=yes')"); } #!endif return; } # 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"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); route(RELAY); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; #!endif return; } # Sample branch router branch_route[BRANCH_ONE] { xlog("L_INFO","XLOG: BRANCH ROUTE "); exit; } # Sample onreply route onreply_route[REPLY_ONE] { #xdbg("incoming reply\n"); xlog("L_INFO","XLOG: REPLY ROUTE "); #!ifdef WITH_NAT # if ((isflagset(5) || isbflagset("6")) && status=~"(183)|(2[0-9][0-9])") { # force_rtp_proxy(); # } if (isbflagset("6")) { fix_nated_contact(); } #!endif } # Sample failure route failure_route[1] { #!ifdef WITH_NAT # if (is_method("INVITE") # && (isbflagset("6") || isflagset(5))) { # unforce_rtp_proxy(); # } #!endif xlog("L_INFO","XLOG ****************FAILROUTE*($rm)*S_IP($si)**CallID($ci)********************"); if (t_is_canceled()) { xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } # ----------------------------------------------------------------------------------------- # Forward call on Voicemail when spefified SIP Status Code is returned. Failure route registered using t_on_failure. # 486 - Busy Here # 408 - Request Timeout # 480 Temporarily Unavailable # if (t_check_status("486|408|480")) { if (t_check_status("486|408")) { xlog("L_INFO","XLOG: [failure route] Forwarding on Voicemail"); } # ----------------------------------------------------------------------------------------- }

2 1

Caller ID issue
by Lucas Alvarez 12 Oct '10

12 Oct '10

Hi, I'm having a problem with the caller ID, I have implemented an integration between asterisk and kamailio following this tutorial: http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb and the problem is that when I call from extension, let's say 1000, to another extension, let's say 2000, the callerid number is always the number I'm calling, in this case 2000. Using xlog and printing $fu, $fU variables I realize that when the call came from asterisk to the destination number, kamailio changes the "From" headers. I will appreciate any kind of help. Regards. Lucas

2 3

Stateful vs. stateless replies from script
by Alex Balashov 12 Oct '10

12 Oct '10

Now that K v3.1.0 provides send_reply(), what is the preferred ideology about whether to send stateful or stateless negative error replies? I mean in general, not in specific cases like digest authentication in the new 'auth' module, where, according to the docs, a transaction and stateful replies are required in order for enhanced nonce replay protection. The traditional wisdom in past bodies of scripts and knowledge in the public realm has been, in my interpretation, that stateless replies are acceptable for final negative responses from script[1]. After all, a reply like "404 Not Found" from the proxy is basically fire-and-forget; why devote additional transactional memory to it while the transaction hold-down timer expires? We do not care if we receive an ACK, nor want anything further to do with this branch. On the other hand, there is a point of view out there that mixing stateless and stateful replies is not good, despite the somewhat increased overhead and resource usage associated with making all replies stateful. This perspective emphasises that if the proxy is fundamentally operating in a stateful manner, it should do so with respect to all messages, rather than allowing a large category of replies to slip under the radar of TM and have possibly inconsistent results. For now, I have changed my code to use send_reply() in all cases where sl_send_reply() was used in the past; it seems easier to let Kamailio/SR make the decision. However, in practice, most requests will result in the creation of a transaction, so it is like replacing sl_send_reply() with t_reply() in the majority of cases. Am I missing anything? Any salient considerations I missed? I am curious as to the consensus about the best practice here from the vantage point of the developers. Thanks, -- Alex [1] Except in FAILURE-ROUTEs, from which only stateful replies may be issued. -- Alex Balashov - Principal Evariste Systems LLC 1170 Peachtree Street 12th Floor, Suite 1200 Atlanta, GA 30309 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

3 6

Seg fault when executing kamailio 3.1.0
by Mino Haluz 12 Oct '10

12 Oct '10

These are some lines from syslog. Where could I find the problem ? Oct 12 15:13:15 nod2 /usr/local/sbin/kamailio[23102]: DEBUG: <core> [sr_module.c:625]: find_export_record: found <load_tm> in module tm [/usr/local/lib/kamailio/modules/tm.so] Oct 12 15:13:15 nod2 /usr/local/sbin/kamailio[23102]: DEBUG: <core> [sr_module.c:625]: find_export_record: found <t_newtran> in module tm [/usr/local/lib/kamailio/modules/tm.so] Oct 12 15:13:15 nod2 /usr/local/sbin/kamailio[23102]: DEBUG: <core> [sr_module.c:625]: find_export_record: found <t_relay_to_tcp> in module tm [/usr/local/lib/kamailio/modules/tm.so] Oct 12 15:13:15 nod2 /usr/local/sbin/kamailio[23102]: DEBUG: <core> [sr_module.c:625]: find_export_record: found <t_relay_to_udp> in module tm [/usr/local/lib/kamailio/modules/tm.so] Oct 12 15:13:15 nod2 /usr/local/sbin/kamailio[23102]: DEBUG: <core> [sr_module.c:625]: find_export_record: found <t_relay> in module tm [/usr/local/lib/kamailio/modules/tm.so] Oct 12 15:13:15 nod2 /usr/local/sbin/kamailio[23102]: DEBUG: <core> [sr_module.c:625]: find_export_record: found <t_forward_nonack> in module tm [/usr/local/lib/kamailio/modules/tm.so] Oct 12 15:13:15 nod2 /usr/local/sbin/kamailio[23102]: DEBUG: <core> [sr_module.c:625]: find_export_record: found <t_release> in module tm [/usr/local/lib/kamailio/modules/tm.so] Oct 12 15:13:15 nod2 /usr/local/sbin/kamailio[23102]: <core> [mem/q_malloc.c:366]: qm_malloc(0xb4ff8000, 24) called from tm: t_hooks.c: insert_tmcb(159) Oct 12 15:13:15 nod2 /usr/local/sbin/kamailio[23102]: <core> [mem/q_malloc.c:406]: qm_malloc(0xb4ff8000, 24) returns address 0xb516d824 frag. 0xb516d80c (size=24) on 1 -th hit Oct 12 15:13:15 nod2 /usr/local/sbin/kamailio[23102]: DEBUG: <core> [sr_module.c:625]: find_export_record: found <bind_sl> in module sl [/usr/local/lib/kamailio/modules/sl.so] Oct 12 15:13:15 nod2 kernel: [693453.525967] kamailio[23102]: segfault at 0 ip (null) sp bfe37c8c error 4 in kamailio[8048000+1d1000]

2 1

kamailio error in parse message
by hala alramli 12 Oct '10

12 Oct '10

Oct 12 09:15:50 localhost /usr/local/kamailio-1.4-pp/sbin/kamailio[3178]: ERROR:core:parse_method: invalid character øOct 12 09:15:50 localhost /usr/local/kamailio-1.4-pp/sbin/kamailio[3178]: INFO:core:parse_first_line: failed to parse the methodOct 12 09:15:50 localhost /usr/local/kamailio-1.4-pp/sbin/kamailio[3178]: INFO:core:parse_first_line: bad messageOct 12 09:15:50 localhost /usr/local/kamailio-1.4-pp/sbin/kamailio[3178]: ERROR:core:parse_msg: message=<ø^S1^\^B }^S ¡6 } \216^F^A^O\206^D¡½>Oct 12 09:15:50 localhost /usr/local/kamailio-1.4-pp/sbin/kamailio[3178]: ERROR:core:receive_msg: parse_msg failed this error keeping appear in my log file ,what is the reason of that ? or what is cussing that?how can i fix that ? thanks for any help

2 1

Development builds for debian and ubuntu available
by Jon Bonilla 12 Oct '10

12 Oct '10

Starting today, Kamailio's debian repository offers nightly development versions of "master" and "3.1" branches for Ubuntu and Debian distributions. Brief summary of the repository status at the moment: * Flavours: Kamailio * Branches: 3.0, 3.1, master * Distributions: Etch (3.0 stable only), Lenny, Squeeze, Lucid * Archs: amd64, i386 * Sources (deb-src) provided for stable releases * Nightly builds of supported branches All the instructions and status updates can be found here: http://www.kamailio.org/dokuwiki/doku.php/packages:debs We hope you will find these dev builds usefull for testing or playing. Sipdoc Team

2 1

error installing ser
by Meftah Tayeb 11 Oct '10

11 Oct '10

hi all, i have error installing ser in centOs 5.5 i am not getting serctl and serdbctl, only sercmd and ser any idea is welcome thank you -- Meftah Tayeb Sales Manager Ove Telecom direct: +13477595883 INUM: +883510001289101 MSN: sqlsrvx86(a)hotmail.com yahoo: vsdevx86 OVETEL USA: +1-212-401-0707 OVETEL UK: +44-20-34110486 OVETEL Ukraine: +380-44-3607309 OVETEL Jordan: +962-6-2508905 OVETEL South Africa: +27-11-4613345 http://www.ovetel.com follow us on Twitter for the latest price updates: http://twitter.com/ovetel

5 11

Problem with ser - INVITE size problem tcp->udp?
by Simone Felici 11 Oct '10

11 Oct '10

Hello to all! I need a little help with our ser installation (ser-2.0.0-rc1). The continuous groving up of our infrastructure and using even more codecs, causes the INVITE (udp) to be over 1500bytes. An external->incoming call to our proxy sip comes in with a size of ~1300 and will be forwarded to the end device (sua) over 1500. Ser is adding more than 200bytes in the udp packet. I can see 'INVITE' from ser to customer, but the INVITE is not answerred due the size. I'm sure about that, because in certain situations, deleting codec list the packet-size goes under the 1500bytes and all works. To main issue is from external calls to ser server to our customers. Calls started from customers to ser and then out are all ok. Now the question: Could the passage from UDP to TCP solve the issue? Could someone give me an example and how implement it into our ser configuration? The atteched configuration, ser on a mysql basis, is working well in expect the issue described. Thank's a lot for your help! Simon

2 5

Size of an mqueue
by Alex Balashov 11 Oct '10

11 Oct '10

According to the documentation, Optional attribute 'size' specifies the maximum number of items in queue, if it is execeeded the oldest one is removed Is this meant to imply that by default, absent this attribute, queues are boundless? Or is there a default size anyway? Thanks, -- Alex Balashov - Principal Evariste Systems LLC 1170 Peachtree Street 12th Floor, Suite 1200 Atlanta, GA 30309 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

2 1

Question about mqueue
by Alex Balashov 11 Oct '10

11 Oct '10

First, thank you, Ramona, for implementing 'mqueue', and the development team more generally. I got away from 'acc' over the weekend and how use mqueue + rtimer + sqlops for asynchronous CDR logging. Classy! The question is, there a clean, fast way of checking whether there are more items left in the queue to consume? I would have thought mq_fetch() would return when there's nothing else left in the queue, but it doesn't; instead, it throws the script into an infinite loop. :-) So, I am using: route[RECORD_CONSUMER] { while(mq_fetch("somequeue")) { if(! defined $mqk(somequeue)) break; # Do things with $mqk(...) and $mqv(...) } } Is 'defined' a fast-acting operator within route script? Is there a better way? -- Alex Balashov - Principal Evariste Systems LLC 1170 Peachtree Street 12th Floor, Suite 1200 Atlanta, GA 30309 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

1 2

git repository open for development
by Daniel-Constantin Mierla 11 Oct '10

11 Oct '10

Hello, just in case someone was waiting for a more formal announcement regarding the close of frozen state for GIT repository, then it is now open for new development. Thanks again to all that helped to make stable branch v3.1.x happen. Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

Re: [SR-Users] 2 instances of openser
by Daniel-Constantin Mierla 09 Oct '10

09 Oct '10

Hello, don't forget to keep the list cc-ed. Further private emails are ignored. On 10/9/10 9:54 PM, Erico Oliveira wrote: > Hi Daniel, > > First, tanks for the help ! > Look... I tried do it with openser, and I got a error (don't ask > me because I don't remember).... Anyway, I will try again and will go > download the kamalio. > After, I return a feedback to you ! ok. Running several instances of sip proxy was possible since ser 0.x.y, year 2002. Cheers, Daniel > Really tanks ! > > Regard's > > On Sat, Oct 9, 2010 at 3:43 PM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > I recommend the same of installing last kamailio version, 1.1.1 > is really really old. > > However, I don't understand why you need to edit the binary with > GHex. You can run kamailio (or older versions that had openser > name) as many times as you want with different configuration > files, just be sure you don't conflict on listening sockets. > > For example: > > /usr/local/sbin/kamailio -f /usr/local/etc/kamailio/kamailio1.cfg > /usr/local/sbin/kamailio -f /usr/local/etc/kamailio/kamailio2.cfg > > Cheers, > Daniel > > > On 10/9/10 4:58 PM, Juha Heinanen wrote: > > Erico Oliveira writes: > > Now, I have almost everything functional (like > freeradius and mediaproxy > and mysql requests), but the Openser can't startup..... > For a while, I copy > his binary (*/usr/local/sbin/openser*) to* > /usr/local/sbin/openserhost* and > change the openser file's content -- where was openser.cfg > now is > openserhost.cfg) -- with a hexadecimal editor (GHex). I > created a > */etc/init.d/openserhost > , /etc/default/openserhost* and a* > /usr/local/etc/openserhost/ *too. > > i suggest that you install latest kamailio package for > distribution of > your choice. then, if your problems still persist, send > another email to > the list. > > -- juha > > _______________________________________________ > 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 > > -- Daniel-Constantin Mierla http://www.asipto.com

1 0

Failed to start kamailio with permissions module
by Shrouk Khan 09 Oct '10

09 Oct '10

hi all, i am trying to load the permission module in the following manner: loadmodule "permissions.so" #---- permission params ------ modparam("permissions", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("permissions", "db_mode",1) modparam("permissions", "trusted_table", "trusted") and i am getting the following error: ct 9 04:44:25 ubuntu /sbin/kamailio[1469]: WARNING: permissions [permissions.c:604]: default allow file (//etc/kamailio/permissions.allow) not found => empty rule set Oct 9 04:44:25 ubuntu /sbin/kamailio[1469]: WARNING: permissions [permissions.c:613]: default deny file (//etc/kamailio/permissions.deny) not found => empty rule set Oct 9 04:44:25 ubuntu /sbin/kamailio[1469]: ERROR: permissions [address.c:156]: database problem Oct 9 04:44:25 ubuntu /sbin/kamailio[1469]: CRITICAL: permissions [address.c:233]: reload of address table failed Oct 9 04:44:25 ubuntu /sbin/kamailio[1469]: ERROR: permissions [address.c:243]: no more shm memory Oct 9 04:44:25 ubuntu /sbin/kamailio[1469]: ERROR: permissions [permissions.c:627]: failed to initialize the allow_address function Oct 9 04:44:25 ubuntu /sbin/kamailio[1469]: ERROR: <core> [sr_module.c:874]: init_mod(): Error while initializing module permissions any idea what is happening ? -- Regards Shrouk Khan (Khan) System Administrator / Telecommunication System Developer Office: +354 4400807 (Reykjavik) +44 2031370800 (London) Mobile: +66 875049439 (Bangkok) Web: www.softverk.is Reykjavik, Iceland // London, UK // Bangkok, Thailand

4 5

2 instances of openser
by Erico Oliveira 09 Oct '10

09 Oct '10

Hi guys ! I'm new here and I came to the list to try solve a problem first. =) Good.... I have a SIPProxy installed in my machine working with Openser 1.1.1. Now I need to do this SIPProxy answer for 2 different domains and to do something depending of where request came. Now, I have almost everything functional (like freeradius and mediaproxy and mysql requests), but the Openser can't startup..... For a while, I copy his binary (*/usr/local/sbin/openser*) to* /usr/local/sbin/openserhost* and change the openser file's content -- where was openser.cfg now is openserhost.cfg) -- with a hexadecimal editor (GHex). I created a */etc/init.d/openserhost , /etc/default/openserhost* and a* /usr/local/etc/openserhost/ *too. Finally, I edit the "port" (to 5070) and de informations to database connect in new config file. When I try start the software, I get a error like this: *siphost01:/usr/local/etc/openser# /etc/init.d/openserhost start Not starting openserhost: invalid configuration file! siphost01:/usr/local/etc/openser# * Continuing, looking for a log, I see the /var/log/openser/openserhost.log and got: * Oct 9 09:57:17 siphost01 kernel: openserhost[12020]: segfault at 0000000000000004 rip 00000000f7f6696e rsp 00000000ffcccaec error 4 Oct 9 10:04:33 siphost01 kernel: openserhost[12207]: segfault at 0000000000000004 rip 00000000f7fa096e rsp 00000000ffbc49ec error 4 Oct 9 10:05:36 siphost01 kernel: openserhost[12241]: segfault at 0000000000000004 rip 00000000f7f8096e rsp 00000000fff58d7c error 4 Oct 9 10:54:39 siphost01 kernel: openserhost[13443]: segfault at 0000000000000004 rip 00000000f7f7b96e rsp 00000000ffc52a6c error 4* And that is All..... Sorry if I'm doing somethig stupid or if my english is very very bad.... I'm "new" in "Sr world" and I'm brazilian... =) Wait someone helpme with this.... Regards...... Érico Oliveira Solution Manager

3 2

2 instances of OpenSer
by Erico 09 Oct '10

09 Oct '10

Hi guys ! I'm new here and I came to the list to try solve a problem first. =) Good.... I have a SIPProxy installed in my machine working with Openser 1.1.1. Now I need to do this SIPProxy answer for 2 different domains and to do something depending of where request came. Now, I have almost everything functional (like freeradius and mediaproxy and mysql requests), but the Openser can't startup..... For a while, I copy his binary (//usr/local/sbin/openser/) to/ /usr/local/sbin/openserhost/ and change the openser file's content -- where was openser.cfg now is openserhost.cfg) -- with a hexadecimal editor (GHex). I created a //etc/init.d/openserhost , /etc/default/openserhost/ and a/ /usr/local/etc/openserhost/ /too. Finally, I edit the "port" (to 5070) and de informations to database connect in new config file. When I try start the software, I get a error like this: *siphost01:/usr/local/etc/openser# /etc/init.d/openserhost start Not starting openserhost: invalid configuration file! siphost01:/usr/local/etc/openser# * Continuing, looking for a log, I see the /var/log/openser/openserhost.log and got: * Oct 9 09:57:17 siphost01 kernel: openserhost[12020]: segfault at 0000000000000004 rip 00000000f7f6696e rsp 00000000ffcccaec error 4 Oct 9 10:04:33 siphost01 kernel: openserhost[12207]: segfault at 0000000000000004 rip 00000000f7fa096e rsp 00000000ffbc49ec error 4 Oct 9 10:05:36 siphost01 kernel: openserhost[12241]: segfault at 0000000000000004 rip 00000000f7f8096e rsp 00000000fff58d7c error 4 Oct 9 10:54:39 siphost01 kernel: openserhost[13443]: segfault at 0000000000000004 rip 00000000f7f7b96e rsp 00000000ffc52a6c error 4* And that is All..... Sorry if I'm doing somethig stupid or if my english is very very bad.... I'm "new" in "Sr world" and I'm brazilian... =) Wait someone helpme with this.... Regards...... Érico Oliveira Solution Manager

1 0

Re: [SR-Users] Failed to start kamailio with permissions module
by Shrouk Khan 09 Oct '10

09 Oct '10

On Sat, Oct 9, 2010 at 7:22 PM, Shrouk Khan <shroukkhan(a)softverk.is> wrote: > hi, > actually if i make the address table empty , it works. but if i add > any entry to it , it stops working . > my address table looks like this: > > > id grp ip_addr mask port > 3 0 192.168.189.1 32 5060 > > so the mask is the issue ? or the ip address . I am not great with > networking . i am just trying to allow 192.168.189.1 to be able to > send INVITE msg. > > > On Sat, Oct 9, 2010 at 7:13 PM, Iñaki Baz Castillo <ibc(a)aliax.net> wrote: >> >> 2010/10/9 Shrouk Khan <shroukkhan(a)softverk.is>: >> > its kamailio 3.0.3 from : http://www.kamailio.org/pub/kamailio/3.0.3/src/ >> >> And have you checked the netries in "address" table as I have suggested? >> >> >> -- >> Iñaki Baz Castillo >> <ibc(a)aliax.net> > > > > -- > Regards > > Shrouk Khan (Khan) > System Administrator / Telecommunication System Developer > Office: +354 4400807 (Reykjavik) > +44 2031370800 (London) > Mobile: +66 875049439 (Bangkok) > > Web: www.softverk.is > > Reykjavik, Iceland // London, UK // Bangkok, Thailand > -- Regards Shrouk Khan (Khan) System Administrator / Telecommunication System Developer Office: +354 4400807 (Reykjavik) +44 2031370800 (London) Mobile: +66 875049439 (Bangkok) Web: www.softverk.is Reykjavik, Iceland // London, UK // Bangkok, Thailand

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Permission module error
by Shrouk Khan 09 Oct '10

09 Oct '10

hi, i have been trying to use the allow_trusted() to allow or disallow invite request from certain IP address. This is the block that i am using: loadmodule "permission.so" ...... ........ if(is_method("INVITE")){ if(!allow_trusted()){ sl_send_reply("403","Address Not allowed"); exit; } } but trying to restart kamailio after this fails. and from the syslog it says: Oct 9 03:19:49 ubuntu kamailio: ERROR: <core> [sr_module.c:378]: ERROR: load_module: could not find module <permission> in <//lib/kamailio/modules_k/://lib/kamailio/modules/> Oct 9 03:19:49 ubuntu kamailio: : <core> [cfg.y:3330]: parse error in config file //etc/kamailio/kamailio.cfg, line 151, column 12-26: failed to load module Oct 9 03:19:49 ubuntu kamailio: : <core> [cfg.y:3333]: parse error in config file //etc/kamailio/kamailio.cfg, line 356, column 21: unknown command, missing loadmodule?#012 I think i forgot to compile kamailio with the support of permission module . but i do not see it anywhere in the ignore list either. can anyone tell me what could be the problem -- Regards Shrouk Khan (Khan) System Administrator / Telecommunication System Developer Office: +354 4400807 (Reykjavik) +44 2031370800 (London) Mobile: +66 875049439 (Bangkok) Web: www.softverk.is Reykjavik, Iceland // London, UK // Bangkok, Thailand

1 1

Evaluate Perl Return Value in Routing Logic
by "Nicolas Rüger" 08 Oct '10

08 Oct '10

Hello, is there any way to use or evaluate the return-statements of a perl-script in kamailio.cfg? I tried the following in the Routing Logic in kamailio.cfg: if (is_method("INVITE")) { if (perl_exec("my_perl_subroutine") == -1){ xlog("PERL returns -1 \n"); } } Doesn't work. The bad thing is that the return-values of the - perl_exec("XXX") - call are not the same that the perl subroutine "XXX" returns. That's at least what I strongly believe after testing. I do need to read/evaluate the return-value of the perl-subroutine in the Routing Logic to define different routes depending on what the perl-subroutine returns. Is there any way to do that??? Thank you for your help in advance. Regards, Nicolas -- GRATIS für alle GMX-Mitglieder: Die maxdome Movie-FLAT! Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01

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Regarding SER Listening on two ports
by Saurabh Srivastava 08 Oct '10

08 Oct '10

Hello SER Team, We are using SER as a SIP Server for our purpose. We wan to configure SER to listen on two ports at the same time(i.e. 5060 & some other custom port such as 5090 or something like that). Can you please assist us how can we achieve this, means what changes we need to make such that SER listens on two ports at the same time. Thanks and Best Regards Saurabh Srivastava

2 1

how to check if t_newtran() has already been called?
by Juha Heinanen 07 Oct '10

07 Oct '10

can t_check_trans() be also used to check if t_newtran() has been called on the request earlier in the script? if so, is it so heavyweight thing that it would be better to set a flag when t_newtran() is called and then test the flag? -- juha

1 0

t_relay_cancel usage
by Juha Heinanen 07 Oct '10

07 Oct '10

in tm readme there is this kind of example regarding use of t_relay_cancel: if (method == CANCEL) { if (!t_relay_cancel()) { # implicit drop if relaying was successful, # nothing to do # corresponding INVITE transaction found but error occurred sl_reply("500", "Internal Server Error"); drop; } # bad luck, corresponding INVITE transaction is missing, # do the same as for INVITEs } if proxy is processing all invites statefully, why anything needs to be done when invite transaction corresponding the to cancel is missing? isn't it a case of unmatched cancel and the cancel could just be dropped? -- juha

3 5

www_challenge(realm, flags) question
by Juha Heinanen 07 Oct '10

07 Oct '10

does www_challenge(realm, flags) know to send its challenge reply statefully or statelessly depending on if t_newtran() has been called or not? -- juha

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Question about early media
by Asim Riaz 07 Oct '10

07 Oct '10

Hi List, Phone-1 ------->Proxy1 (kamailio)-------------->Proxy-2(3rd party) -------------> phone 2 Calls from phone-1 to phone-2, proxy-2 is sending early media without sdp in provisional reply, this early media is sourced from different port than response (200), what port proxy-1 will use to transmit actual voice, the source port in early media or source port in response(200) which is different than early media. Thanks, Asim

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WARNING: script writer didn't release transaction
by Juha Heinanen 07 Oct '10

07 Oct '10

after playing with t_newtran(), i started to get these to syslog: Oct 7 15:39:11 sip /usr/sbin/sip-proxy[26119]: WARNING: tm [t_lookup.c:1543]: WARNING: script writer didn't release transaction as far as i remember, in k these messages were not printed, because transactions were freed automatically when script ended. also, example in section 1.3.3. nonce_count of auth module does not include any t_release() calls. where exactly should they be put? -- juha

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auth module README t_newtran() issue
by Juha Heinanen 07 Oct '10

07 Oct '10

in auth module readme, there is this kind of example code in section 1.3.3. nonce_count: # go stateful and catch retransmissions if (!t_newtran()) drop; # retransmission based on tm source comment, the #retransmission comment is not correct: /** if no transaction already exists for the message, create a new one. * atomic "new_tran" construct; it returns: * * @return <0 on error * +1 if a request did not match a transaction * 0 on retransmission that is, if t_newtran() returns false, there has been an error. drop happens automatically in case of retransmission. -- juha

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kamailio 3.0.3 tls problems
by Rouskol Andrey 07 Oct '10

07 Oct '10

Hello, I have a problem running kamailio with tls support. I can successfully register only for the first time, all further registration fails with the following message in the log: Sep 28 11:53:55 siptest /usr/sbin/kamailio[16963]: ERROR: tls [tls_server.c:392]: SSL error:error:07064041:memory buffer routines:BUF_MEM_grow:malloc failure The system is a default kamailio installation on a Debian 5.0.6 system. Does somebody know what could cause the problem? Regards, Andrey.

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authenticating users based on IP address and limiting the number of channels for the users using Kamailio 3.0 and Siremis 1.0
by Shrouk Khan 07 Oct '10

07 Oct '10

hi, i have been looking for a way to authenticate users based on the IP from they come from . Also i would like to limit the number of channels ( simultaneous calls) that each user can make . is their any way to do this using kamailio 3.0 and siremis 1.0 ? my configuration is like this: sipphones --->UserGroup1(Asterisk PBX)---->KamailioSipProxy--->AsteriskPSTNGateway--->PSTN 1. there can be many softphones on the user group 1 and there can be multiple calls coming from there at any time. 2. UserGroup1 is actually another SIP proxy [asterisk] with a fixed public IP 3. Kamailio Sip proxy is the one i am trying to configure . I would like to authenticate UserGroup1 based on its public IP address 4. From kamailio , calls will be forwarded to a asterisk pstn gateway. I would like to put restriction on this one : " User group 1 can not make more than 5 concurrent calls at a time" can anyone give me some suggestion or point me in some direction where I can do some reading on this one. Aslo , what would be best way to bill this UserGroup1 ? Using FreeRadius and CDRTool ? -- Regards Shrouk Khan (Khan) System Administrator / Telecommunication System Developer Office: +354 4400807 (Reykjavik) +44 2031370800 (London) Mobile: +66 875049439 (Bangkok) Web: www.softverk.is Reykjavik, Iceland // London, UK // Bangkok, Thailand

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force_rtp_proxy assistance
by Stagg Shelton 06 Oct '10

06 Oct '10

I'm working on a setup where we have rtpproxy on a machine with eth0 IP 10.10.5.141/19 and eth1 IP 10.10.10.78/24. When using force_rtp_proxy("","10.10.5.141"); the connection information (c) field in the SDP is not being set to the ip 10.10.5.141 It is instead being set to 10.10.10.78 We are starting to run out of ideas on getting the proper contact address into the sdp based on call source signaling ip. Does anyone have any ideas what the problem could be? Thanks Stagg

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Re: [SR-Users] add fromuser to sip request
by JR Richardson 06 Oct '10

06 Oct '10

>> Hi Alex. I use a general context in asterisk to route calls already >> and wanted to get away from pattern matches. I'm using asterisk >> realtime and want to decrease the load. I'll play around with the >> 'uac_replace_from()' and 'append_hf' to see what works best for me. > > Well, you don't have to use pattern matches. :-) You can always > direct to fixed extensions, a la: > > exten => moh,1,Answer > exten => moh,n,MusicOnHold > exten => moh,n,Hangup > > Those should be pretty snappy, even in realtime. > So the first thing I notice with using 'uac_replace_from()' is the caller ID number is replaced with the replaced value, so I guess this is not what I need for my particular application. I'll fool around with the 'append_hf' and some pattern matches. Thanks. JR -- JR Richardson Engineering for the Masses

2 1

Kamailio v3.1.0 Released
by Daniel-Constantin Mierla 06 Oct '10

06 Oct '10

Kamailio (OpenSER) v3.1.0 is out – major release with impressing number of new features and improvements. This release is a result of more than 8 months of development and testing from the teams of Kamailio (OpenSER) and SIP Express Router (SER) projects. Backed up by a solid development group, we are proud to announce that this release brings a large set of features, many for first time on the SIP server market, such as asynchronous TLS, UDP raw sockets, embedded HTTP and XCAP servers, embedded Lua, configuration file debugger. All together, there are over 15 new modules and countless improvements to old components. Since last major release (version 3.0.0, which was out in January 10, 2010), the two SIP servers, Kamailio and SER, are practically the same application, the name making the difference regarding the database structure and the extensions used for certain features, such as user database based authentication or location service. Therefore another development direction was towards smooth integration of Kamailio and SER extensions, previously duplicated modules such as auth, sl, ratelimit or sms were merged during this development cycle. Continue reading the release notes at: http://www.kamailio.org/w/kamailio-openser-v3.1.0-release-notes/ Many thanks to all developers and community members that made possible this release. -- Daniel-Constantin Mierla http://www.asipto.com

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General question
by Uriel Rozenbaum 06 Oct '10

06 Oct '10

Guys, This mail has nothing to do with Kamailio or SR, so feel free to discard it. I'm trying to find a command-line tool to grab updates from an RSS feed. I tried to install rsstail but it has a lot of dependencies and cannot make it work. Does anyone have some software I can download and use in Centos 5.2? Thanks! Uriel

1 0

Problem with debug memory: kill -SIGUSR1 doesn't work fine
by César Pinto Magán 06 Oct '10

06 Oct '10

Hello, We had been experiencing memory leaks issues with our Kamailio deployment. Some of the process started to use all their assigned memory and arises allocation memory errors. Using the "kill -SIGUSR1" as explained in the Kamailio wiki and found the problem: Sep 8 11:34:04 localhost /sbin/kamailio[17275]: INFO: <core> [main.c:788]: INFO: signal 15 received Sep 8 11:34:04 localhost /sbin/kamailio[17275]: INFO: <core> [main.c:799]: Memory status (pkg): Sep 8 11:34:04 localhost /sbin/kamailio[17275]: INFO: qm_status: (0x885a60): Sep 8 11:34:04 localhost /sbin/kamailio[17275]: INFO: qm_status: heap size= 67108864 Sep 8 11:34:04 localhost /sbin/kamailio[17275]: INFO: qm_status: used= 60641440, used+overhead=67096424, free=12440 Sep 8 11:34:04 localhost /sbin/kamailio[17275]: INFO: qm_status: max used (+overhead)= 67102896 Sep 8 11:34:04 localhost /sbin/kamailio[17275]: INFO: qm_status: dumping all alloc'ed. fragments: Sep 8 11:34:04 localhost /sbin/kamailio[17275]: INFO: qm_status: 0. N address=0x8baf78 frag=0x8baf48 size=16 used=1 Sep 8 11:34:04 localhost /sbin/kamailio[17275]: INFO: qm_status: alloc'd from <core>: rvalue.c: mk_rval_expr_v(2452) Sep 8 11:34:04 localhost /sbin/kamailio[17275]: INFO: qm_status: start check=f0f0f0f0, end check= c0c0c0c0, abcdefed ... Sep 8 11:34:06 localhost /sbin/kamailio[17275]: INFO: qm_status: 1618. N address=0x95e920 frag=0x95e8f0 size=936 used=1 Sep 8 11:34:06 localhost /sbin/kamailio[17275]: INFO: qm_status: alloc'd from <core>: rvalue.c: rval_new_empty(233) Sep 8 11:34:06 localhost /sbin/kamailio[17275]: INFO: qm_status: start check=f0f0f0f0, end check= c0c0c0c0, abcdefed Sep 8 11:34:06 localhost /sbin/kamailio[17275]: INFO: qm_status: 1633. N address=0x95f528 frag=0x95f4f8 size=936 used=1 Sep 8 11:34:06 localhost /sbin/kamailio[17275]: INFO: qm_status: alloc'd from <core>: rvalue.c: rval_new_empty(233) Sep 8 11:34:06 localhost /sbin/kamailio[17275]: INFO: qm_status: start check=f0f0f0f0, end check= c0c0c0c0, abcdefed Sep 8 11:34:06 localhost /sbin/kamailio[17275]: INFO: qm_status: 1634. N address=0x95f930 frag=0x95f900 size=936 used=1 Sep 8 11:34:06 localhost /sbin/kamailio[17275]: INFO: qm_status: alloc'd from <core>: rvalue.c: rval_new_empty(233) Sep 8 11:34:06 localhost /sbin/kamailio[17275]: INFO: qm_status: start check=f0f0f0f0, end check= c0c0c0c0, abcdefed Sep 8 11:34:06 localhost /sbin/kamailio[17275]: INFO: qm_status: 1696. N address=0x962298 frag=0x962268 size=944 used=1 Sep 8 11:34:06 localhost /sbin/kamailio[17275]: INFO: qm_status: alloc'd from <core>: rvalue.c: rval_new_empty(233) Sep 8 11:34:06 localhost /sbin/kamailio[17275]: INFO: qm_status: start check=f0f0f0f0, end check= c0c0c0c0, abcdefed . . (the rvalue.c repeats all the time) . Sep 8 11:34:23 localhost /sbin/kamailio[17275]: INFO: qm_status: 64963. N address=0x4885118 frag=0x48850e8 size=944 used=1 Sep 8 11:34:23 localhost /sbin/kamailio[17275]: INFO: qm_status: alloc'd from <core>: rvalue.c: rval_new_empty(233) Sep 8 11:34:23 localhost /sbin/kamailio[17275]: INFO: qm_status: start check=f0f0f0f0, end check= c0c0c0c0, abcdefed Sep 8 11:34:23 localhost /sbin/kamailio[17275]: INFO: qm_status: 64964. N address=0x4885528 frag=0x48854f8 size=1288 used=1 Sep 8 11:34:23 localhost /sbin/kamailio[17275]: INFO: qm_status: alloc'd from <core>: rvalue.c: rval_new_empty(233) Sep 8 11:34:23 localhost /sbin/kamailio[17275]: INFO: qm_status: start check=f0f0f0f0, end check= c0c0c0c0, abcdefed We saw there was several core memory leaks resolved in the 3.0.3 version (we had the 3.0.2), so we decided to upgrade to that version. Now I was trying to check the memory status of the process (in order to see if the problem is arising again) with "kill -SIGUSR1" but now only show the memory status of the shm and the memory of the attendant process (the one who launch the other ones). I had tried to send the signal to each of the son process but I see nothing in the log. Is this a new bug or is that there is other new way to see the memory status of the processes? Thanks in advance, http://www.alhambra-eidos.es/CO/1.gif<http://www.alhambra-eidos.es/CO/1.gif> César Pinto. ICM Team +34 91 787 23 00 http://www.alhambra-eidos.es/CO/3.gif <http://www.alhambra-eidos.es/>

2 2

Fwd: lua.h / lauxlib.h issue while making deb pkgs ! (for 3.1 master branch i.e)
by X- x -X 06 Oct '10

06 Oct '10

Shouldn't me doing " git pull " today /even now have fixed it ? (it still not doing it for me though). I still get that error. of not finding those lua.h etc.. I do see it being upgrade to 3.1 during the compile ........................... dpkg-buildpackage: source package kamailio dpkg-buildpackage: source version 3.1.0 dpkg-buildpackage: source changed by Daniel-Constantin Mierla < miconda(a)gmail.com> dpkg-buildpackage: host architecture i386 .......................... On Wed, Oct 6, 2010 at 6:03 AM, Jon Bonilla <manwe(a)aholab.ehu.es> wrote: > El Wed, 6 Oct 2010 04:55:55 -0400 > X- x -X <myonlinedisk(a)gmail.com> escribió: > > > Hey Jony, > > > > while during "make deb" it complains of lua files > > > > Make sure you have an updated 3.1 branch. There was a fix for this > yesterday by > Daniel. > >

1 0

start preparing 3.1.0 release in 2 hours
by Daniel-Constantin Mierla 06 Oct '10

06 Oct '10

Hello, in order to build the packages for 3.1.0 without conflicts, please make your commits before 14:00CEST (Madrid, Paris, Berlin - UTC+2). If you still have some to do afterwards, please ask on sr-dev(a)lists.sip-router.org before committing. Thanks, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

lua.h / lauxlib.h issue while making deb pkgs ! (for 3.1 master branch i.e)
by X- x -X 06 Oct '10

06 Oct '10

Hey Jony, while during "make deb" it complains of lua files output .................................. In file included from app_lua_api.c:37: app_lua_api.h:27:17: warning: lua.h: No such file or directory app_lua_api.h:28:21: warning: lauxlib.h: No such file or directory app_lua_api.h:29:20: warning: lualib.h: No such file or directory In file included from app_lua_api.c:37: app_lua_api.h:35: error: expected specifier-qualifier-list before ‘lua_State’ app_lua_api.h:58: error: expected ‘)’ before ‘*’ token --------------------------------- I have all the 3 versions devs of lua installed debbox:~/kamailio# apt-file search lualib.h liblua5.1-0-dev: /usr/include/lua5.1/lualib.h liblualib40-dev: /usr/include/lua40/lualib.h liblualib50-dev: /usr/include/lua50/lualib.h dpkg - L | grep -i liblua debbox:~/kamailio# dpkg -l |grep lua ii liblua40 4.0-13 Main interpreter library for the Lua 4.0 programming language ii liblua40-dev 4.0-13 Main interpreter library for lua40: static library and headers ii liblua5.1-0 5.1.3-1 Simple, extensible, embeddable programming language ii liblua5.1-0-dev 5.1.3-1 Simple, extensible, embeddable programming language ii liblua50 5.0.3-3 Main interpreter library for the Lua 5.0 programming language ii liblua50-dev 5.0.3-3 Main interpreter library for Lua 5.0: static library and heade ii liblualib40 4.0-13 Extension library for the Lua 4.0 programming language ii liblualib40-dev 4.0-13 Extension library for Lua 4.0: static and headers ii liblualib50 5.0.3-3 Extension library for the Lua 5.0 programming language ii liblualib50-dev 5.0.3-3 Extension library for Lua 5.0: static and headers ii lua50 5.0.3-3 Small embeddable language with simple procedural syntax I reckon the path is not correct in one of the header files. vi modules/app_lua/app_lua_api.h ..................... #include <lua.h> #include <lauxlib.h> #include <lualib.h> ........................ could you enlighten me ? Thanx.

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