- sr-users - kamailio.org

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

31 Aug '11

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

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

30 Aug '11

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

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

30 Aug '11

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

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

30 Aug '11

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

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SEMS 1.4.2 released
by Stefan Sayer 29 Aug '11

29 Aug '11

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

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

29 Aug '11

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

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

29 Aug '11

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

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

29 Aug '11

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

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enum_pv_query
by Andrew Melton 28 Aug '11

28 Aug '11

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

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Persistent dialog
by Jon Bonilla 27 Aug '11

27 Aug '11

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

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insert_hf in branch_route
by Gustavo Garcia Bernardo 26 Aug '11

26 Aug '11

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

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

26 Aug '11

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

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

26 Aug '11

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

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

26 Aug '11

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

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

25 Aug '11

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

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

25 Aug '11

Spam detection software, running on the system "mail.iptel.org", has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: Dear user serusers(a)lists.iptel.org, mail server administrator of lists.iptel.org would like to let you know the following, We have detected that your e-mail account has been used to send a huge amount of junk email messages during this week. Probably, your computer was infected by a recent virus and now runs a hidden proxy server. [...] Content analysis details: (14.0 points, 5.0 required) pts rule name description ---- ---------------------- -------------------------------------------------- 0.5 FSL_XM_419 Old OE version in X-Mailer only seen in 419 spam 0.4 RCVD_IN_XBL RBL: Received via a relay in Spamhaus XBL [87.235.117.21 listed in zen.spamhaus.org] 0.0 DEAR_EMAIL BODY: Message contains Dear email address 0.8 BAYES_50 BODY: Bayes spam probability is 40 to 60% [score: 0.5000] 1.4 RCVD_IN_BRBL_LASTEXT RBL: RCVD_IN_BRBL_LASTEXT [87.235.117.21 listed in bb.barracudacentral.org] 0.3 FSL_UA FSL_UA 0.8 RDNS_NONE Delivered to internal network by a host with no rDNS 2.6 MSOE_MID_WRONG_CASE MSOE_MID_WRONG_CASE 1.2 TO_NO_BRKTS_MSFT To: misformatted and supposed Microsoft tool 1.9 FORGED_MUA_OUTLOOK Forged mail pretending to be from MS Outlook 2.5 DOS_OE_TO_MX Delivered direct to MX with OE headers 1.5 TO_NO_BRKTS_DIRECT To: misformatted and direct-to-MX The original message was not completely plain text, and may be unsafe to open with some email clients; in particular, it may contain a virus, or confirm that your address can receive spam. If you wish to view it, it may be safer to save it to a file and open it with an editor.

1 0

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

25 Aug '11

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

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

25 Aug '11

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

2 1

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

25 Aug '11

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

1 0

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

25 Aug '11

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

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

25 Aug '11

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

1 0

Float Comparison
by Brandon Armstead 25 Aug '11

25 Aug '11

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

4 9

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

25 Aug '11

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

2 5

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

25 Aug '11

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

4 5

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

25 Aug '11

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

3 2

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

24 Aug '11

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

3 2

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

23 Aug '11

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

2 4

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

23 Aug '11

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

2 2

max branches
by Jon Bonilla 23 Aug '11

23 Aug '11

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

4 4

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

23 Aug '11

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

1 0

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

23 Aug '11

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

1 0

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

23 Aug '11

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

2 1

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

22 Aug '11

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

3 11

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

22 Aug '11

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

1 0

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

22 Aug '11

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

2 2

min expires question
by Fabian Borot 22 Aug '11

22 Aug '11

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

3 4

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

22 Aug '11

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

2 5

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

22 Aug '11

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

1 0

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

22 Aug '11

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

1 0

masking carriers
by Anders 20 Aug '11

20 Aug '11

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

2 2

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

19 Aug '11

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

2 2

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

19 Aug '11

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

3 2

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

19 Aug '11

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

5 7

LCR Modules Comparison
by Spencer Thomason 19 Aug '11

19 Aug '11

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

4 3

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

19 Aug '11

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

2 1

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

19 Aug '11

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

5 7

decode_mime_type error
by Jijo 18 Aug '11

18 Aug '11

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

2 2

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

18 Aug '11

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

3 5

lookup for specified user
by Efelin Novak 17 Aug '11

17 Aug '11

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

3 5

siptrace loosing an ACK
by Timo Klecker 17 Aug '11

17 Aug '11

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

3 3

Error message
by Jon Bonilla 16 Aug '11

16 Aug '11

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

2 1

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

15 Aug '11

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

4 5

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

15 Aug '11

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

3 2

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

15 Aug '11

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

1 0

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

15 Aug '11

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

1 1

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

15 Aug '11

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

2 1

pua_xmpp questions
by laura testi 13 Aug '11

13 Aug '11

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

2 8

subscriber table reload
by Fabian Borot 12 Aug '11

12 Aug '11

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

4 5

mtree question
by Juha Heinanen 12 Aug '11

12 Aug '11

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

1 0

kamailio - proxy/relay
by Sunny 11 Aug '11

11 Aug '11

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

3 4

time for acc records
by Daniel-Constantin Mierla 11 Aug '11

11 Aug '11

Hello, it seems that is some parts of the world it is required to have call duration with two decimals precision. By default acc module does the timestamp in seconds. Thinking of adding an option to save the milliseconds, I see two option: - a new column to store the milliseconds part as integer - a new column to store the seconds.milliseconds as double For those using various billing engines, what would be easier to digest when rating the CDRs? There is already the option of doing db extra column to store $TV(Sn), but this value is stored in a string (varchar) format, giving a little more troubles to convert to numeric format before being able to compute duration, so it might just worth doing the C coding to store in right data type. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

4 5

Re: [SR-Users] Default kamailio installation, syntax errors with Defined Values
by Gertjan Wolzak 11 Aug '11

11 Aug '11

Sorry, my bad.... Missing dependencies Regards, Gertjan Wolzak

1 0

Default kamailio installation, syntax errors with Defined Values
by Gertjan Wolzak 11 Aug '11

11 Aug '11

Hello, Just been installing Kamailio, version 3.1.x via Git, but when I use the default kamailio.cfg, Kamailio will not start and the debug gives the following errors. 0(25876) : <core> [cfg.y:3330]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 285, column 29-35: syntax error 0(25876) : <core> [cfg.y:3330]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 285, column 29-35: Invalid arguments 0(25876) : <core> [cfg.y:3333]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 285, column 36: ERROR: bad config file (4 errors) When I check the line, column it is the following line of the acc params: modparam("acc", "log_flag", FLT_ACC) I assume the variable FLT_ACC is not set or read correctly. But as it is the default kamailio.cfg that seems strange to me. The defined values are: ####### 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://openser:openserrw@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 Does anyone has an idea why te values are not loaded into the variables.? With regards, Gertjan Wolzak

1 0

Using Kamailio to convert TCP to TLS
by aur.hoffmann＠free.fr 10 Aug '11

10 Aug '11

Hello, I would like to use Kamailio as a TCP <=> TLS convertor ? Do you think it's possible ? I have developed a software (which simulate a SIP phone) which can use UDP or TCP (not TLS). I need a SIP proxy to convert TCP to TLS. For example: UA => TCP OPTION (5060) => Kamailio => TLS OPTION (5061) => OXE UA <= TCP 200 OK (5060) <= Kamailio <= TLS 200 OK (5061) <= OXE Do you think Kamailio can do that ? Thanks in advance, Regards

3 2

$hu is not documented
by Juha Heinanen 10 Aug '11

10 Aug '11

xcap_server examples refer to pseudo variable $hu, which is not documented on pseudo variables wiki page: http://sip-router.org/wiki/cookbooks/pseudo-variables/devel -- juha

2 2

Cancel issue
by Carl Wagner 09 Aug '11

09 Aug '11

Hi, I am having an issue getting Kamailio to properly cancel a call. I initiate the call with an INVITE from Asterisk and then hang up. According to rfc3665 the call flow should be: 3.8. Unsuccessful No Answer Alice Proxy 1 Proxy 2 Bob | | | | | INVITE F1 | | | |--------------->| INVITE F2 | | | 100 F3 |--------------->| INVITE F4 | |<---------------| 100 F5 |--------------->| | |<---------------| | | | | 180 F6 | | | 180 F7 |<---------------| | 180 F8 |<---------------| | |<---------------| | | | CANCEL F9 | | | |--------------->| | | | 200 F10 | | | |<---------------| CANCEL F11 | | | |--------------->| | | | 200 F12 | | | |<---------------| | | | | CANCEL F13 | | | |--------------->| | | | 200 F14 | | | |<---------------| | | | 487 F15 | | | |<---------------| | | | ACK F16 | | | 487 F17 |--------------->| | |<---------------| | | | ACK F18 | | | 487 F19 |--------------->| | |<---------------| | | | ACK F20 | | | |--------------->| | | In my case I only have one proxy - the Kamailio box, and the call is coming from Asterisk instead of Alice. The is the call flow that I am seeing: Asterisk Kamailio SIP_phone | | | | INVITE | | |--------------->| INVITE | | 100 |--------------->| |<---------------| 100 | | |<---------------| | | | | | 180 | | 180 |<---------------| |<---------------| | | CANCEL | | |--------------->| | | | CANCEL | | |--------------->| | 200 Canceling | | |<---------------| | | | 200 | | |<---------------| | | | | | | | | 487 | | |<---------------| | | ACK | | 487 |--------------->| |<---------------| | | ACK | | |--------------->| | | CANCEL | | |--------------->| | | 200 Canceling | | |<---------------| | | CANCEL | | |--------------->| | | 200 Canceling | | |<---------------| | It looks like the problem is that that "200 Canceling" has a To: tag and it shouldn't. ====================================== # cancel from Asterisk: U 2011/05/24 15:18:56.123714 3.3.3.3:5060 -> 2.2.2.2:5060 CANCEL sip:SIP_13034445555@2.2.2.2 SIP/2.0. Via: SIP/2.0/UDP 3.3.3.3:5060;branch=z9hG4bK51c9602a;rport. From: "+13032223333" <sip:+13032223333@3.3.3.3>;tag=as4c415d87. To: <sip:SIP_13034445555@2.2.2.2>. Call-ID: 605b8de40a05924177085d71154bda77(a)3.3.3.3. CSeq: 102 CANCEL. User-Agent: VerbalWorld. Max-Forwards: 70. Content-Length: 0. # 200 canceling from Kamailio to asterisk: U 2011/05/24 15:18:56.127349 2.2.2.2:5060 -> 3.3.3.3:5060 SIP/2.0 200 canceling. Via: SIP/2.0/UDP 3.3.3.3:5060;branch=z9hG4bK51c9602a;rport=5060. From: "+13032223333" <sip:+13032223333@3.3.3.3>;tag=as4c415d87. To: <sip:SIP_13034445555@2.2.2.2>;tag=4ed8ab9f3ed29a77613bb14b3431ff43-2f34. Call-ID: 605b8de40a05924177085d71154bda77(a)3.3.3.3. CSeq: 102 CANCEL. Server: kamailio (3.1.3 (i386/linux)). Content-Length: 0. ====================================== Why is kamialio adding a To: tag? Portion of kamailio.cfg ===================== ... route { ... route(WITHINDLG); ### only initial requests (no To: tag) from here on if (is_method("CANCEL")) { if (t_check_trans()) { xlog("L_INFO", "MAIN: Recieved Cancel, t_check_trans is true\n"); ... t_relay(); exit; } ... Thanks! Carl

3 9

Modifying $rU after lookup()
by Efelin Novak 09 Aug '11

09 Aug '11

Hi Folks, I would like to ask how can I change the $rU (user part of uri) in multiple appended branches? In my scenario I do lookup("location") and after this I try to modify an username using the $rU variable. This works fine until there are several records in the location table for the given user (registered from different locations). In this case only one branch is changed and unchanged uri is sent to an another location. Is there a way to change it in all branches together? Thanks for the answer. Regards, Efelin

5 7

Re: [SR-Users] Using SR for Number Translation & Number Portabilty Applications
by Henning Westerholt 09 Aug '11

09 Aug '11

On Monday 08 August 2011, Yaron Nachum wrote: > I have seen the match_mode but I haven't seen how is it possible to set it > up per domain. > > I would appreciate if you could explain how. Hi Yaron, please always CC to the mailling list. Its not possible to set the match_mode per domain, its a global parameter. With regards to the domains, this was another option you could use, e.g. use domain 0 for calls without the 'D' prefix, and then domain D for calls with 'D' prefix. Best regards, Henning

1 0

Re: [SR-Users] Using SR for Number Translation & Number Portabilty Applications
by Henning Westerholt 08 Aug '11

08 Aug '11

> Yaron wrote: > Thanks for the quick response. I have already played with the sqlops - > this is an obvious way to do it. I will further investigate cr and sqlops > and decide which is best for my purposes. Hello Yaron, > I liked the idea of having several instances of SIP-ROUTERs which are > independent of each other- I understand that cr answers it. Indeed, this is one advantage of using an in-memory cache. But you probably need some more DB access on the servers for things like authentication. > I have another question - I was trying to add alpha-numeric character > into the cr prefix and it seems it is not supported. We are using 'D' prefixes > in our network in order to route between different elements in the network. > Is there a way to support it? Maybe another routing module? The cr module support also a extended matching mode, but it will be probably not really feasible with millions of routes: http://kamailio.org/docs/modules/stable/modules/carrierroute.html#id2651893 Another thing you could do is use another "domain" for the D prefix, e.g. one for the default routing, and then another one for the D case. The module support matching for the domain as well. You could maybe also use flags for matching. Other possible modules for PSTN routing are lcr and mtree. Regards, Henning

1 0

Using SR for Number Translation & Number Portabilty Applications
by Yaron Nachum 08 Aug '11

08 Aug '11

Hi Everyone, I have downloaded SR r 3.1.4 and have been playing around with it. I am considering implementing Number Translation services (1800...) and Number Portability Services on it. We have around 20K entries on the Number Translation Service and around 3M entries on the Number portability Service. The volume is around 100 caps. I have started working with the carrierroute module - it seems perfect for the Number translation Service, but I am not sure about the Number portability. I have several questions: 1. Is it possible to use this module with several millions of entries? 2. What is the impact of loading the table? Is it possible to do it during traffic hours? I have read about PDB module, but I haven't understood how to setup the PDB server. I would appreciate for any guidance. Would you recommend using the PDB module for the Number Portability or the carrierroute? Thanks.

2 1

suggested number of TCP / UDP workers for version 3.1.x
by Francesco Castellano 08 Aug '11

08 Aug '11

Dear sirs, searching on the web and in old ML threads about the suggested number of TCP and UDP workers, I just found: - for TCP children (i.e., TCP receivers, isn't it?) "As a rule of thumb, (maximum simultaneous connections)/2000 should be OK" (from doc/tcp_tunning.txt) - In general for workers (from a past thread by Henning Westerholt): "In my experience the number of children is not that important for performance, you may just choose the default size of 8." In the last post there was, moreover, a suggestion that the number of children should be chosen with respect to RAM and CPU cores (that makes sense to me). The only thing I can think about the rule of thumb above comes from the possibility that a number of processes may be unavailable in blocking operations. So my question is: - with asynchronous TCP/TLS is it safe to assume that more children than CPU cores are more or less useless? - are there other blocking operations for which the number of UDP/TCP children should be incremented above the CPU cores number? Thank you! Best regards, Francesco Castellano

2 2

Kamailio Redundant Systems
by HM Kias 08 Aug '11

08 Aug '11

Hi, Anyone tried pacemaker, heartbeat or OpenAIS for building redundant kamailio systems? Please share your experiences. Also auggest a better solution. Regards, -- HM Kias 91-9443467600

2 1

Homer Project
by Alexandr Dubovikov 05 Aug '11

05 Aug '11

Hello sr-users, I would like to present a new project called "Homer". What is Homer ? Homer is a capture solution for SIP logging. How does it work? Homer has 4 components: 1. Capture Agent 2. Capture Node 4. Capture DB 3. Web GUI. The capture agent takes your local SIP traffic and sends it to the defined capture node in HEP protocol. If you are user of FreeSWITCH or Kamailio, use build-in capture agent. In other case, you can use an external generic daemon, but only for SIP/UDP. The capture node is based on kamailio, can receive SIP traffic encapsulated in IP proto 4 (IPIP) or in HEP procotol and store it into MySQL DB. To gain access to the capture DB, download a simple webinterface (PHP based). After it you are able to do search, generate PCAP file, callflow, look at the details of SIP messages etc. Homer works in our company and easy collect ~5 M packets per hour. The patches for kamailio and some other information can be found here: http://homer.googlecode.com Beta testers are welcome :-) I'll help everybody to setup/configure Homer :-) With best regards, Alexandr

3 3

$du gets set automatically before branch_route
by Alex Hermann 05 Aug '11

05 Aug '11

Hello, while trying to upgrade from 1.5 to 3.1 i noticed that between t_relay() and branch_route, $du gets automatically set to $ru if it has not been set before. This is a change from 1.5 i could not find documentation for, so it might have been unintened. Can someone enlighten me on the purpose of this change and if there is a way to determine if $du is (was) unset other than comparing $ru to $du, wich seems a bit inefficient? Is it even guaranteed that $du gets set to $ru in every possible scenario? -- Alex Hermann

3 8

udp vs. tcp receiver processes
by Juha Heinanen 05 Aug '11

05 Aug '11

i have 8 children specified in default file and in cfg file i have: listen=192.98.101.10:5060; listen=udp:127.0.0.1:5070; listen=udp:127.0.0.1:5070; listen=tcp:127.0.0.1:5070; listen=udp:127.0.0.1:5080; listen=tcp:127.0.0.1:5080; listen=tcp:127.0.0.1:6060; that results in 8 tcp receiver processes but 3 x 8 udp receiver processes: # sip-proxy_ctl ps 30720 attendant 30721 udp receiver child=0 sock=192.98.101.10:5060 30722 udp receiver child=1 sock=192.98.101.10:5060 30723 udp receiver child=2 sock=192.98.101.10:5060 30724 udp receiver child=3 sock=192.98.101.10:5060 30725 udp receiver child=4 sock=192.98.101.10:5060 30726 udp receiver child=5 sock=192.98.101.10:5060 30727 udp receiver child=6 sock=192.98.101.10:5060 30728 udp receiver child=7 sock=192.98.101.10:5060 30729 udp receiver child=0 sock=127.0.0.1:5070 30730 udp receiver child=1 sock=127.0.0.1:5070 30731 udp receiver child=2 sock=127.0.0.1:5070 30732 udp receiver child=3 sock=127.0.0.1:5070 30733 udp receiver child=4 sock=127.0.0.1:5070 30734 udp receiver child=5 sock=127.0.0.1:5070 30735 udp receiver child=6 sock=127.0.0.1:5070 30736 udp receiver child=7 sock=127.0.0.1:5070 30738 udp receiver child=0 sock=127.0.0.1:5080 30740 udp receiver child=1 sock=127.0.0.1:5080 30752 udp receiver child=2 sock=127.0.0.1:5080 30754 udp receiver child=3 sock=127.0.0.1:5080 30755 udp receiver child=4 sock=127.0.0.1:5080 30756 udp receiver child=5 sock=127.0.0.1:5080 30758 udp receiver child=6 sock=127.0.0.1:5080 30760 udp receiver child=7 sock=127.0.0.1:5080 30761 slow timer 30763 timer 30765 ctl handler 30803 TIMER NH 30805 TIMER RT 30806 tcp receiver child=0 30808 tcp receiver child=1 30810 tcp receiver child=2 30812 tcp receiver child=3 30813 tcp receiver child=4 30844 tcp receiver child=5 30846 tcp receiver child=6 30847 tcp receiver child=7 30849 tcp main process why cannot one udp receiver process handle all listening addresses like tcp processes seem to be able to do? the above seems like waste of resources. -- juha

3 4

Kamailio on a server with different network cards
by Rocío Ruz 05 Aug '11

05 Aug '11

Hello, My server has two different network cards with different IPs. I want kamailio to use both; the network card to be used depends on the destination. My problem is that I have to add an IP in the configuration file kamailio.cfg: listen=109.239.50.67 Will Kamailio listen only to packets sent to this IP? Can I add the two IPs corresponding to my two network cards here? Thank you RR

3 5

480 Temporarily Unavailable. in kamailio 3.1.4
by hala alramli 04 Aug '11

04 Aug '11

i installed kamailio and freeswitch according to http://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc i installed also rtpproxy . i managed to create 101,102,103 and all of them registered , i am not able to make calls between them, 480 temporarily unavailable always the error . i cant find in the configuration file what cause this error. this is the track. i will be thankful for any help . Hala interface: eth0 (192.168.1.0/255.255.255.0) filter: (ip or ip6) and ( port 5060 ) match: 101 U 2011/08/04 13:16:57.351872 192.168.1.175:5060 -> 192.168.1.233:5060 INVITE sip:102@192.168.1.233 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.175:5060;branch=z9hG4bK-6eabf9b3. From: <sip:101@192.168.1.233>;tag=43c22ff96586a71ao0. To: <sip:102@192.168.1.233>. Call-ID: 9acd1ca5-9162c25e(a)192.168.1.175. CSeq: 101 INVITE. Max-Forwards: 70. Contact: <sip:101@192.168.1.175:5060>. Expires: 240. User-Agent: Linksys/PAP2T-5.1.6(LS). Content-Length: 444. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura, replaces. Content-Type: application/sdp. . v=0. o=- 986176 986176 IN IP4 192.168.1.175. s=-. c=IN IP4 192.168.1.175. t=0 0. m=audio 16450 RTP/AVP 8 0 2 4 18 96 97 98 100 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:4 G723/8000. a=rtpmap:18 G729a/8000. a=rtpmap:96 G726-40/8000. a=rtpmap:97 G726-24/8000. a=rtpmap:98 G726-16/8000. a=rtpmap:100 NSE/8000. a=fmtp:100 192-193. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:30. a=sendrecv. U 2011/08/04 13:16:57.352037 192.168.1.233:5060 -> 192.168.1.175:5060 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 192.168.1.175:5060;branch=z9hG4bK-6eabf9b3. From: <sip:101@192.168.1.233>;tag=43c22ff96586a71ao0. To: <sip:102@192.168.1.233>;tag=b27e1a1d33761e85846fc98f5f3a7e58.7d11. Call-ID: 9acd1ca5-9162c25e(a)192.168.1.175. CSeq: 101 INVITE. Proxy-Authenticate: Digest realm="192.168.1.233", nonce="TjqA1U46f6liNM8a6LX5HBJ2wXEBtOIX". Server: kamailio (3.1.4 (x86_64/linux)). Content-Length: 0. . U 2011/08/04 13:16:57.370381 192.168.1.175:5060 -> 192.168.1.233:5060 ACK sip:102@192.168.1.233 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.175:5060;branch=z9hG4bK-6eabf9b3. From: <sip:101@192.168.1.233>;tag=43c22ff96586a71ao0. To: <sip:102@192.168.1.233>;tag=b27e1a1d33761e85846fc98f5f3a7e58.7d11. Call-ID: 9acd1ca5-9162c25e(a)192.168.1.175. CSeq: 101 ACK. Max-Forwards: 70. Contact: <sip:101@192.168.1.175:5060>. User-Agent: Linksys/PAP2T-5.1.6(LS). Content-Length: 0. . U 2011/08/04 13:16:57.380065 192.168.1.175:5060 -> 192.168.1.233:5060 INVITE sip:102@192.168.1.233 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.175:5060;branch=z9hG4bK-eab8c613. From: <sip:101@192.168.1.233>;tag=43c22ff96586a71ao0. To: <sip:102@192.168.1.233>. Call-ID: 9acd1ca5-9162c25e(a)192.168.1.175. CSeq: 102 INVITE. Max-Forwards: 70. Proxy-Authorization: Digest username="101",realm="192.168.1.233",nonce="TjqA1U46f6liNM8a6LX5HBJ2wXEBtOIX",uri="sip:102@192.168.1.233",algorithm=MD5,response="a019828b7a7334539ffa6b668dfcfada". Contact: <sip:101@192.168.1.175:5060>. Expires: 240. User-Agent: Linksys/PAP2T-5.1.6(LS). Content-Length: 444. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura, replaces. Content-Type: application/sdp. . v=0. o=- 986176 986176 IN IP4 192.168.1.175. s=-. c=IN IP4 192.168.1.175. t=0 0. m=audio 16450 RTP/AVP 8 0 2 4 18 96 97 98 100 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:4 G723/8000. a=rtpmap:18 G729a/8000. a=rtpmap:96 G726-40/8000. a=rtpmap:97 G726-24/8000. a=rtpmap:98 G726-16/8000. a=rtpmap:100 NSE/8000. a=fmtp:100 192-193. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:30. a=sendrecv. U 2011/08/04 13:16:57.380329 192.168.1.233:5060 -> 192.168.1.175:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 192.168.1.175:5060;branch=z9hG4bK-eab8c613. From: <sip:101@192.168.1.233>;tag=43c22ff96586a71ao0. To: <sip:102@192.168.1.233>. Call-ID: 9acd1ca5-9162c25e(a)192.168.1.175. CSeq: 102 INVITE. Server: kamailio (3.1.4 (x86_64/linux)). Content-Length: 0. . U 2011/08/04 13:16:57.488511 192.168.1.233:5060 -> 192.168.1.175:5060 SIP/2.0 480 Temporarily Unavailable. Via: SIP/2.0/UDP 192.168.1.175:5060;branch=z9hG4bK-eab8c613. From: <sip:101@192.168.1.233>;tag=43c22ff96586a71ao0. To: <sip:102@192.168.1.233>;tag=NDrNpDZ1g0jZe. Call-ID: 9acd1ca5-9162c25e(a)192.168.1.175. CSeq: 102 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-5d37bc9 2011-08-03 09-05-01 -0500. From: <sip:101@192.168.1.233>;tag=43c22ff96586a71ao0. To: <sip:102@192.168.1.233>. Call-ID: 9acd1ca5-9162c25e(a)192.168.1.175. CSeq: 102 INVITE. Server: kamailio (3.1.4 (x86_64/linux)). Content-Length: 0. . U 2011/08/04 13:16:57.488511 192.168.1.233:5060 -> 192.168.1.175:5060 SIP/2.0 480 Temporarily Unavailable. Via: SIP/2.0/UDP 192.168.1.175:5060;branch=z9hG4bK-eab8c613. From: <sip:101@192.168.1.233>;tag=43c22ff96586a71ao0. To: <sip:102@192.168.1.233>;tag=NDrNpDZ1g0jZe. Call-ID: 9acd1ca5-9162c25e(a)192.168.1.175. CSeq: 102 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-5d37bc9 2011-08-03 09-05-01 -0500. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Reason: Q.850;cause=16;text="NORMAL_CLEARING". Content-Length: 0. Remote-Party-ID: "kb-102" <sip:kb-102@192.168.1.233>;party=calling;privacy=off;screen=no. . U 2011/08/04 13:16:57.502085 192.168.1.175:5060 -> 192.168.1.233:5060 ACK sip:102@192.168.1.233 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.175:5060;branch=z9hG4bK-eab8c613. From: <sip:101@192.168.1.233>;tag=43c22ff96586a71ao0. To: <sip:102@192.168.1.233>;tag=NDrNpDZ1g0jZe. Call-ID: 9acd1ca5-9162c25e(a)192.168.1.175. CSeq: 102 ACK. Max-Forwards: 70. Proxy-Authorization: Digest username="101",realm="192.168.1.233",nonce="TjqA1U46f6liNM8a6LX5HBJ2wXEBtOIX",uri="sip:102@192.168.1.233",algorithm=MD5,response="a019828b7a7334539ffa6b668dfcfada". Contact: <sip:101@192.168.1.175:5060>. User-Agent: Linksys/PAP2T-5.1.6(LS). Content-Length: 0. .

2 1

How to reset t_on_failure/branch?
by Iñaki Baz Castillo 04 Aug '11

04 Aug '11

Hi, I remember that in Kamailio 1.5.X t_on_branch and t_on_failure still remain after rouning failure_route[XX], and I "think" there was a way to reset them. In 3.X it seems that t_on_branch and t_on_failure must be set again in failure_route to take effect in next t_relay() call, am I right? Thanks. -- Iñaki Baz Castillo <ibc(a)aliax.net>

2 2

AVPs missing in onreply_route
by Sebastian Damm 04 Aug '11

04 Aug '11

Hi, I'm still fighting problems in our Kamailio 1.5 to 3.1 migration. Right now I'm stuck with a strange behavior of AVPs in my dialplan. I set a lot of AVPs for accounting in the route directive, and I just inserted an avp_print() statement right before t_relay() is called, all values are set. Then I call avp_print() as the first statement of onreply_route(), and it doesn't print out anything. I call it again at the end of onreply_route, and the only AVP printed out is the one I set directly there. In 1.5 there was an option in the tm module to control AVP behavior in replies, but that doesn't exist anymore, I even found a mailing list post from Daniel where he explained that all AVPs I set in route are also available in onreply_route. Am I doing something wrong? Is there still a secret option that I accidentally turned on? What else could make kamailio forget all AVPs after t_relay? I tried both version 3.1.3 and 3.1.4. Any help appreciated. Best regards, Sebastian

3 9

ERROR : when create database in kamailio 3.1.0
by hala alramli 04 Aug '11

04 Aug '11

hello i a trying to instal kamailio and free switch according to the tutorial in http://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc but when i try to create the database and the extra tables this error apear could you help me? thanks Hala root@noga1:~# nano /usr/local/etc/kamailio/kamctlrc root@noga1:~# /usr/local/sbin/kamdbctl create MySQL password for root: database engine 'mysql' loaded INFO: test server charset INFO: creating database kamailio ... Creating core table: standard Creating core table: acc Creating core table: lcr Creating core table: domain Creating core table: group Creating core table: permissions Creating core table: registrar Creating core table: usrloc Creating core table: msilo Creating core table: alias_db Creating core table: uri_db Creating core table: speeddial Creating core table: avpops Creating core table: auth_db Creating core table: pdt Creating core table: dialog Creating core table: dispatcher Creating core table: dialplan INFO: Core Kamailio tables succesfully created. Install presence related tables? (y/n): y INFO: creating presence tables into kamailio ... INFO: Presence tables succesfully created. Install tables for imc cpl siptrace domainpolicy carrierroute userblacklist htable purple uac pipelimit? (y/n): y INFO: creating extra tables into kamailio ... Creating extra table: imc Creating extra table: cpl Creating extra table: siptrace Creating extra table: domainpolicy Creating extra table: carrierroute Creating extra table: userblacklist Creating extra table: htable Creating extra table: purple Creating extra table: uac Creating extra table: pipelimit /usr/local/lib64/kamailio//kamctl/kamdbctl.mysql: line 225: /usr/local/share/kamailio//mysql/pipelimit-create.sql: No such file or directory ERROR: Creating extra tables failed!

2 2

Re: [SR-Users] Chicago, ClueCon next week
by Daniel-Constantin Mierla 03 Aug '11

03 Aug '11

Hello, for questions about kamailio (openser), the best place to address them is still the mailing list for users. There are many people there that might be able to answer quickly. Messages sent personally to me for technical questions about kamailio (openser) are discarded, unless they were a result of a discussion on the mailing list where I asked for sensitive data. Cheers, Daniel On 8/3/11 1:47 AM, ifeanyi okoye wrote: > hi Daniel..i would like to discuss some of the issues am having with Openser/Kamailio > -----Original Message----- > Date: Tuesday, August 02, 2011 5:15:39 pm > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List"<sr-users(a)lists.sip-router.org> > Cc: ifeanyi okoye<ifeanyi_okoye(a)yahoo.com> > From: "Daniel-Constantin Mierla"<miconda(a)gmail.com> > Subject: Re: [SR-Users] Chicago, ClueCon next week > > Hello Ify, > > On 8/2/11 11:15 PM, ifeanyi okoye wrote: >> Hello Daniel, >> >> My name is Ify and I'm a research voip engineer.I would be attending >> the conference. Please can I reach you via telephone. >> > the phone might not be the most convenient at that time, so the best is > to ask the organizers to point you to me, unless you can spot me based > on the pictures you can find on the web. I will have the presentation on > the 2nd day, but I will be around on 1st day as well. > > Looking forward to meeting you, > Daniel > >> Regards, >> Ify >> >> --- On *Tue, 8/2/11, Daniel-Constantin Mierla /<miconda(a)gmail.com>/* >> wrote: >> >> >> From: Daniel-Constantin Mierla<miconda(a)gmail.com> >> Subject: [SR-Users] Chicago, ClueCon next week >> To: "kamailio users"<sr-users(a)lists.sip-router.org>, >> business(a)lists.kamailio.org >> Date: Tuesday, August 2, 2011, 1:59 AM >> >> Hello, >> >> I'll be in Chicago few days next week for ClueCon conference. If >> you are in the area, but no -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

Chicago, ClueCon next week
by Daniel-Constantin Mierla 03 Aug '11

03 Aug '11

Hello, I'll be in Chicago few days next week for ClueCon conference. If you are in the area, but not attending the ClueCon conference, drop me a message in case you want to meet. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

4 5

Kamailio 3.1.4 Bitwise Operators?
by Brandon Armstead 02 Aug '11

02 Aug '11

Hello, It seems as if the bitwise operations in Kamailio 3.1.4 are broken? Please correct me if I am doing something wrong - however simple evaluations such as: $avp(s:my-test-bit) = 10; $avp(s:my-test-bit) = (int)$avp(s:my-test-bit) & ~ 2; Seems to result in $avp(s:my-test-bit) being equal to "2" rather than "8" Any thoughts / Suggestions / Fixes? I also wonder if this has any relation to !~ not working? Thanks for all of your help in advance! P.S. I sent this both to user list and to devel, please forgive me if they do not belong in one of the lists. Sincerely, Brandon Armstead

2 3

Re: [SR-Users] 10 years of SER - statistics
by Javier Gallart 02 Aug '11

02 Aug '11

Hi Daniel a few numbers here...we use Kamailio in a carrier environment (no subscribers, no nat...), basically as proxy and redirect server. We use it in production and also as one of the foundations of other features still under development. As a redirect server, on average it handles around 600k-700k calls a day. Regards Javi > ------------------------------ > > Message: 4 > Date: Tue, 02 Aug 2011 11:23:26 +0200 > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > Subject: [SR-Users] 10 years of SER - statistics > To: kamailio users <sr-users(a)lists.sip-router.org>, sr-dev > <sr-dev(a)lists.sip-router.org>, business(a)lists.kamailio.org > Message-ID: <4E37C20E.3020902(a)gmail.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Hello, > > it is now one month till the 10 years SER event in Berlin. I am thinking > to collect few statistics about usage of SER-based code at this moment > (SER, OpenSER, Kamailio, OpenIMSCore ...) and present them during the > conference and publish on the web. > > Everything is going to be like a combined report, not individual > listing. Therefore, if you want to participate, your name or company > won't be mentioned. > > Here are the stats I thought of: > > - type of usage: production, evaluation (testing), research > - number of subscribers (phone lines) > - percentage of phones behind NAT > - number of calls per month > - number of call minutes per month > > I am not looking for exact numbers, but rough estimation (e.g., about 10 > 000 phone lines, ...). Of course, some of the metrics don't apply always > (e.g., if you do termination routed through SIP server, you don't have > subscribers, but just calls traffic). Those doing deployments, can make > one summary of statistics for all instances. You may send other > statistics you think they worth publishing. > > Again, this is voluntary, naming is not required. I hope only those > giving real number will write back -- I will try anyhow to figure out if > someone is just dumping fantasy numbers. If privacy is really a big > concern, from case to case, everything can be done under NDA. > > Thanks, > Daniel > > -- > Daniel-Constantin Mierla -- http://www.asipto.com > Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat > http://linkedin.com/in/miconda -- http://twitter.com/miconda > > > > > >

1 0

Carrierroute Compilation error Kamailio 3.1.4
by Omar 01 Aug '11

01 Aug '11

I am Using Solaris 10 and GCC 4.2, i do have libconfuse everything goes fine until this error comes up. carrierroute.c:54: error: 'MODULE_INTERFACE_VER' undeclared here (not in a function) carrierroute.c:87: error: expected '=', ',', ';', 'asm' or '__attribute__' before 'cmds' carrierroute.c:132: error: variable 'exports' has initializer but incomplete type carrierroute.c:133: warning: excess elements in struct initializer carrierroute.c:133: warning: (near initialization for 'exports') carrierroute.c:134: warning: excess elements in struct initializer carrierroute.c:134: warning: (near initialization for 'exports') carrierroute.c:135: error: 'cmds' undeclared here (not in a function) carrierroute.c:135: warning: excess elements in struct initializer carrierroute.c:135: warning: (near initialization for 'exports') carrierroute.c:136: warning: excess elements in struct initializer carrierroute.c:136: warning: (near initialization for 'exports') carrierroute.c:137: warning: excess elements in struct initializer carrierroute.c:137: warning: (near initialization for 'exports') carrierroute.c:138: warning: excess elements in struct initializer carrierroute.c:138: warning: (near initialization for 'exports') carrierroute.c:139: warning: excess elements in struct initializer carrierroute.c:139: warning: (near initialization for 'exports') carrierroute.c:140: warning: excess elements in struct initializer carrierroute.c:140: warning: (near initialization for 'exports') carrierroute.c:141: warning: excess elements in struct initializer carrierroute.c:141: warning: (near initialization for 'exports') carrierroute.c:142: warning: excess elements in struct initializer carrierroute.c:142: warning: (near initialization for 'exports') carrierroute.c:143: warning: excess elements in struct initializer carrierroute.c:143: warning: (near initialization for 'exports') carrierroute.c:145: warning: excess elements in struct initializer carrierroute.c:145: warning: (near initialization for 'exports') carrierroute.c: In function 'mod_init': carrierroute.c:161: error: invalid use of undefined type 'struct module_exports' make[1]: *** [carrierroute.o] Error 1 make: *** [modules] Error 1

3 8

Detect phone and fax
by Linux Guy 31 Jul '11

31 Jul '11

Hi, Is there any way to separate detect phone and fax calls in kamailio ? I understand that kamailio is a signaling proxy only but is there any way we can detect if a call is a fax or phone ? Thanks Linux Guy..

2 1

what is tcp_accept_no_cl core param?
by Juha Heinanen 31 Jul '11

31 Jul '11

xcap_server module requires tcp_accept_no_cl=yes i didn't find that param defined in core cookbook. what does it do? -- juha

2 1

Set CLI.
by Henrik Aagaard Sørensen 29 Jul '11

29 Jul '11

Is it possible to set CLI via the usr_preferences table in Kamailio? Or any other way? So that certain subscribers get there from-number overwritten with a specific one.

3 2

[Urgent] Double REGISTER and MESSAGE issue in Kamailio3.1
by Coca 29 Jul '11

29 Jul '11

Dear List, I have Kamailio 3.1 installed in Amazon EC 2 server. Everything is working fine except that every REGISTER / MESSAGE was looped for twice or 3 times. And I am sure this is not caused by UA. I have attached the ngrep log in this message. And also, my main modification to the kamailio.cfg is like below: Your help will be great great appreciated. Thank you!! Coca ---------------------------------- #!define WITH_MYSQL #!define WITH_NAT alias="mydomain.org" alias="xx.xx.xx.xx" advertised_address="xx.xx.xx.xx" ... ... 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(); /* for NAT support */ record_route_preset("xx.xx.xx.xx"); } # 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); } # 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; } }

1 0

Problem using textops API
by Timo Klecker 29 Jul '11

29 Jul '11

Hi, I'm trying to extend a Kamailio Module and need to add and remove headers in a SIP Message. To do this I want to use the textops module and I am importing the api and trying to bind the textops module. #include "../textops/api.h" [..] struct textops_binds tob; [..] /* register callbacks to TEXTOPS */ if (load_textops_api(&tob)!=0) { LM_ERR("can't load textops api. Is module textops loaded?\n"); return -1; } Later I use tob.append_hf to add a header field. Everything compiles fine but on Kamailio startup there is a Error Message: [..] 0(21782) DEBUG: <core> [route_struct.c:166]: ACTION_#25 #0/2: 9(9)/ (nil) 0(21782) DEBUG: <core> [route_struct.c:166]: ACTION_#25 #1/2: 3(3)/ (nil) 0(21782) DEBUG: <core> [sr_module.c:625]: find_export_record: found <is_method> in module textops [/usr/lib/kamailio/modules_k/textops.so] 0(21782) DEBUG: <core> [route_struct.c:166]: ACTION_#25 #0/2: 9(9)/ (nil) 0(21782) DEBUG: <core> [route_struct.c:166]: ACTION_#25 #1/2: 3(3)/ (nil) 0(21782) DEBUG: <core> [sr_module.c:625]: find_export_record: found <is_method> in module textops [/usr/lib/kamailio/modules_k/textops.so] 0(21782) DEBUG: <core> [route_struct.c:166]: ACTION_#25 #0/2: 9(9)/ (nil) 0(21782) DEBUG: <core> [route_struct.c:166]: ACTION_#25 #1/2: 3(3)/ (nil) [..] 0(21784) DEBUG: <core> [sr_module.c:625]: find_export_record: found <t_release> in module tm [/usr/lib/kamailio/modules/tm.so] 0(21784) DEBUG: <core> [sr_module.c:631]: find_export_record: <load_textops> not found 0(21784) ERROR: siptrace [../../modules_k/textops/api.h:63]: Failed to import load_textops 0(21784) ERROR: siptrace [siptrace.c:314]: can't load textops api. Is module textops loaded? 0(21784) ERROR: <core> [sr_module.c:875]: init_mod(): Error while initializing module siptrace (/usr/lib/kamailio/modules_k/siptrace.so) ERROR: error while initializing modules 0(21784) DEBUG: tm [t_funcs.c:122]: DEBUG: tm_shutdown : start 0(21784) DEBUG: tm [t_funcs.c:125]: DEBUG: tm_shutdown : emptying hash table [..] I am running Kamailio 3.1.4. Is this a bug in Kamailio or am I doing anything wrong in my approach to use textops? Kind regards, Timo Klecker

2 3

problem with siptrace outgoing SIP message logging
by Peter Kaszas 29 Jul '11

29 Jul '11

Dear List members, Im quite new to kamailio, and i got a problem with the siptrace module. I read that siptrace module can capture and log all incoming and outgoing sip messages. So i put to the config file the following lines: loadmodule "siptrace.so" . . . #-------- siptrace paramas ----- modparam("siptrace", "db_url", DBURL) modparam("siptrace", "table", "sip_trace") modparam("siptrace", "trace_flag", 18) modparam("siptrace", "trace_on", 1) modparam("siptrace", "trace_sl_acks", 1) . . . route { sip_trace(); ....... } Now I can capture all incoming SIP messages, but I none of the outgoing. Where can I put the sip_trace(); command to capture also the outgoing messages? Thank you, Peter

2 3

Disk space went full in 3 days. Kamailio 3.1 @debian64
by Coca 29 Jul '11

29 Jul '11

Hi List , I have newly installed Kamailio 3.1 on a debian 64 server , replacing the OpenSIPs server And the subscribers are about 1000 people. After running 3 days, the disk with 8GB became full and caused Mysql stopped. I have checked /etc/my.cnf , and the setting about binary log is like: #log_bin = /var/log/mysql/mysql-bin.log expire_logs_days = 10 max_binlog_size = 100M It seems that binaly log for mysql is disabled. What else possible reason could cause this problem?? Any hint will be helpful. Thank you all, Coca

2 2

Kamctl and Postgresql
by Nathaniel L Keeling 29 Jul '11

29 Jul '11

I was using kamctl on Solaris to add a table entry into a postgresql database table and received this error: # ./kamctl domain showdb psql: warning: extra command-line argument "-c" ignored psql: warning: extra command-line argument "select * FROM domain ; " ignored After some research, I found in kamctl.pgsql the functions: # input: sql query, optional pgsql command-line params pgsql_query() { # if password not yet queried, query it now prompt_pw "PgSQL password for user '$DBRWUSER@$DBHOST'" mecho "pgsql_query: $PGSQL $2 -A -q -t -P fieldsep=' ' -h $DBHOST -U $DBRWUSER $DBNAME -c '$1'" PGPASSWORD="$DBRWPW" $PGSQL $2 \ -A -q -t \ -P fieldsep=" " \ -h $DBHOST \ -U $DBRWUSER \ $DBNAME \ -c "$1" } # input: sql query, optional pgsql command-line params pgsql_ro_query() { mdbg "pgsql_ro_query: $PGSQL $2 -h $DBHOST -U $DBROUSER $DBNAME -c '$1'" PGPASSWORD="$DBROPW" $PGSQL $2 \ -h $DBHOST \ -U $DBROUSER \ $DBNAME \ -c "$1" } I noticed that the database name parameter ($DBNAME) did not have a flag. When I added "-d" flag, kamctl worked. # input: sql query, optional pgsql command-line params pgsql_query() { # if password not yet queried, query it now prompt_pw "PgSQL password for user '$DBRWUSER@$DBHOST'" mecho "pgsql_query: $PGSQL $2 -A -q -t -P fieldsep=' ' -h $DBHOST -U $DBRWUSER $DBNAME -c '$1'" PGPASSWORD="$DBRWPW" $PGSQL $2 \ -A -q -t \ -P fieldsep=" " \ -h $DBHOST \ -U $DBRWUSER \ -d $DBNAME \ -c "$1" } # input: sql query, optional pgsql command-line params pgsql_ro_query() { mdbg "pgsql_ro_query: $PGSQL $2 -h $DBHOST -U $DBROUSER $DBNAME -c '$1'" PGPASSWORD="$DBROPW" $PGSQL $2 \ -h $DBHOST \ -U $DBROUSER \ -d $DBNAME \ -c "$1" } Was this the correct fix or am I missing something else. Thanks Nathaniel

2 1

About RTimer module
by Iñaki Baz Castillo 28 Jul '11

28 Jul '11

Hi RTimer module doc says: ---------- The module executes route blocks on a timer base. It can create new timer processes and execute many route blocks on same timer. ---------- What does it mean "It can create new timer processes"? Let's suppose I run a blocking call (i.e. sending an HTTP request which takes too long) from a route block called by a timer of RTimer. What would it occur? Thanks. -- Iñaki Baz Castillo <ibc(a)aliax.net>

3 5

Kamailio - ISC interface
by Rocío Ruz 28 Jul '11

28 Jul '11

Hello, I was wondering if it would be possible to plug an Application Server on Kamilio solution using the ISC interface (other than the AS module provided with Kamailio solution). Would it be mandatory to use IMS extensions to do so? Thank you. RR

3 4

Kamctl and Postgresql
by Nathaniel L Keeling 28 Jul '11

28 Jul '11

I was using kamctl on Solaris to add a table entry into a postgresql v9.0 database table and received this error: # ./kamctl domain showdb psql: warning: extra command-line argument "-c" ignored psql: warning: extra command-line argument "select * FROM domain ; " ignored After some research, I found in kamctl.pgsql the functions: # input: sql query, optional pgsql command-line params pgsql_query() { # if password not yet queried, query it now prompt_pw "PgSQL password for user '$DBRWUSER@$DBHOST'" mecho "pgsql_query: $PGSQL $2 -A -q -t -P fieldsep=' ' -h $DBHOST -U $DBRWUSER $DBNAME -c '$1'" PGPASSWORD="$DBRWPW" $PGSQL $2 \ -A -q -t \ -P fieldsep=" " \ -h $DBHOST \ -U $DBRWUSER \ $DBNAME \ -c "$1" } # input: sql query, optional pgsql command-line params pgsql_ro_query() { mdbg "pgsql_ro_query: $PGSQL $2 -h $DBHOST -U $DBROUSER $DBNAME -c '$1'" PGPASSWORD="$DBROPW" $PGSQL $2 \ -h $DBHOST \ -U $DBROUSER \ $DBNAME \ -c "$1" } I noticed that the database name parameter ($DBNAME) did not have a flag. When I added "-d" flag, kamctl worked. # input: sql query, optional pgsql command-line params pgsql_query() { # if password not yet queried, query it now prompt_pw "PgSQL password for user '$DBRWUSER@$DBHOST'" mecho "pgsql_query: $PGSQL $2 -A -q -t -P fieldsep=' ' -h $DBHOST -U $DBRWUSER $DBNAME -c '$1'" PGPASSWORD="$DBRWPW" $PGSQL $2 \ -A -q -t \ -P fieldsep=" " \ -h $DBHOST \ -U $DBRWUSER \ -d $DBNAME \ -c "$1" } # input: sql query, optional pgsql command-line params pgsql_ro_query() { mdbg "pgsql_ro_query: $PGSQL $2 -h $DBHOST -U $DBROUSER $DBNAME -c '$1'" PGPASSWORD="$DBROPW" $PGSQL $2 \ -h $DBHOST \ -U $DBROUSER \ -d $DBNAME \ -c "$1" } Was this the correct fix or am I missing something else. Thanks Nathaniel

1 0

Filtering unregister messges
by Lucas Alvarez 28 Jul '11

28 Jul '11

Hi, how can I do to filter the unregister messages? method == "REGISTER" && ??? Thanks ins advance. Lucas

2 1

Kamailio - IMS extensions
by Rocío Ruz 27 Jul '11

27 Jul '11

Hello, I would like to use kamailio and its IMS extensions. Is it mandatory to install and run each module (s-cscf, i-cscf, p-cscf, Hss) on different machines? Could I just install/run all the modules on the same VM? Thank you. RR

2 1

issue tracking system
by Daniel-Constantin Mierla 27 Jul '11

27 Jul '11

Hello, sightly off-topic and cc-ed to users mailing lists, but hopefully there will be people giving valuable feedback based on their experience. When we started the integration of source code for kamailio and ser, we went for FlySpray as issue tracking system (aka bug tracker). The project seems to be dead, nothing happened to it for long time, no updates and there are warnings using latest php versions, that may result in malfunctionality or no-functionality with future versions. Therefore maybe it is the time to make a decision, rather soon than later, but this time we should look a bit more carefully to available options, to avoid ending up (in short time) to the same issue. I see two major options: A) going to use a source code hosting portal for bug traker (like sourceforge, berlios, github, a.s.o.) B) going to use an existing software to be installed on our server and managed by us I tend to go for A) as the maintenance of the software will not be done by us. I do not know which would be the best choice to go for. Of course, we should consider possible down times (like it happened in the past) or slow connection to their servers. B) will give us full control, but then it is about the time to maintain it (upgrades, security patches, a.s.o.). More than that, it is about selecting the right software to go with. Since SEMS code is now hosted in GIT repo of sip-router.org, maybe the tracker can be used by them as well, eventually by future related projects hosted in our git. Opinions? Thanks, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

12 15

Content of $sndto(ip) is "ip:port"
by Andreas Granig 27 Jul '11

27 Jul '11

Hi, Looks like in kam-3.1 the value of $sndto(ip) in onsend_route is "ip:port", e.g. "1.2.3.4:5060" instead of just "1.2.3.4". Am I the only one experiencing this issue? Andreas

2 2

Understanding alias_db correctly.
by Henrik Aagaard Sørensen 27 Jul '11

27 Jul '11

Can someone help me by explaining what the alias_db exactly does: http://www.kamailio.org/docs/modules/stable/modules_k/alias_db.html As I've understood it, I can add several aliases to my subscribers, for example: 100(a)my-domain.com can have the alias 600(a)my-domain.com and therefore a client can login with the username 600(a)my-domain.com, but really have the preferences of 100(a)my-domain.com as that is the real subscriber. But it doesn't seems to work.

3 3

load_credentials into non-AVP PVs
by Alex Balashov 27 Jul '11

27 Jul '11

When I try this: modparam("auth_db", "load_credentials", "$var(x)=some_col"); I get a parse error: malformed or non AVP $var(customer_bg_id) AVP definition Is there any particular reason these days why auth_db columns have to be loaded to AVPs only? -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

2 1

Kamailio Advanced Training
by Roman Dmytriv 27 Jul '11

27 Jul '11

Hello Daniel, Are you planning to organize Kamailio Advanced Training on American soil in the near future? Regards Roman

1 1

kamailio password length
by Jon Bonilla 27 Jul '11

27 Jul '11

I have a running kamailio with db backend and I'd like to use 40 character strings as subscriber passwords. Would be enought to increase the db field (actually varchar25) or is there any other lenght limit in kamailio's code? thanks Jon

2 1

Kamailio-IMS-Packages
by Rocío Ruz 27 Jul '11

27 Jul '11

Hello, I would like to know if Kamailio IMS extension RPM packages are available for Red Hat distributions. If they are available, I would like to know the repository. For the moment I have only found Debian packages: http://www.kamailio.org/w/2011/01/ims-extensions-available-for-testing/ but I am having problems with the repository (http://repository.ng-voice.comseems to be not available). Could anybody please help me? Thank you RR

2 1

Kamailio Multiple Source IP
by Linux Guy 27 Jul '11

27 Jul '11

Hi, How can I set a specific source ip for outbound calls when I have multiple Ips that kamailio is listening on. For Eg. I have a table with Caller ID ( Phone Numbers ) Call Enters Kamailio > Check the table >> If the phone number is found >> SOURCE IP : 1.1.1.1 If the phone number is not found >> Source IP : 2.2.2.2 Kamailio is listening on both 1.1.1.1 and 2.2.2.2 . Thanks. Linux Guy..

2 1

bypass rtp traffic.
by MingHon 26 Jul '11

26 Jul '11

Hi List, i would like to know is it possible to bypass the rtp traffic forwarding to asterisk server? my kamailio and rtpproxy is on the same box and asterisk is on the other box. can kamailio/rtpproxy handle the rtp traffic without forwarding to asterisk box? thanks in advance. -- Regards, MingHon

5 14

Kamailio without sound, what to do?
by Henrik Aagaard Sørensen 26 Jul '11

26 Jul '11

I'm a newbee in the world of Kamailio. I've managed to setup a fresh installation of Kamailio, with authentication etc. (with help from some great guys on this mailing-list). Everything seems to work with registers, calls etc. except that there is no sound on my calls. How do I figure out what the problem is?

2 3

Re: [SR-Users] Avoiding to send 603 DECLINED to un-trusted IPs - Is that possible with Kamilio?
by Andrade Ricardo (CI/AFU1) 25 Jul '11

25 Jul '11

Hello Bruce, Strangely, I've never gotten an Anonymous INVITE rejected by Asterisk with 603, but that was the version 1.4. Let me try to help with your request for a solution with Kamailio. These few lines could be added to your main route, after the place where your handle the registration. It will block any request from unregistered users: Route { ... if (!registered("location")) { { exit; } ... } I am assuming that you refer to the subscribers registered in the location table as "peers", I mean, they should be registered in Kamailio, not in asterisk. If not, you can do something similar with some database query. But let us know if that is the case. Perhaps you'd be interested in these next ones as well, they helped me a short while ago to workaround on some issues with the code 603: Module configurations: modparam("tm", "disable_6xx_block", 1) onreply_route { if ($T_reply_code == "603") { drop; } } Best regards, Ricardo.

2 1

How to get mem_summary (SIGUSR1) on a worker process?
by Andrade Ricardo (CI/AFU1) 25 Jul '11

25 Jul '11

Hi list, I'm running out of private memory in Kamailio 3.1.0. However, before simply increasing it from the current 4 MB to some higher value, I'd like to know how much is really in use for every process and possibly monitor it. My expectation is that I should be able to send the SIGUSR1 to any of the processes shown by the "sercmd ps" command, and get the trace. But it only succeeds if I send the signal to the "attendant" process. Some more info: # sercmd core.info { version: kamailio 3.1.0 id: 21a375 compiler: gcc 4.1.2 compiled: 16:28:54 Dec 14 2010 flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES } kamailio.cfg: debug=2 memlog=1 memdbg=4 mem_summary=15 Any ideas? Best regards, Ricardo Andrade.

1 0

Fresh Kamailio installation doesn't require authorization.
by Henrik Aagaard Sørensen 25 Jul '11

25 Jul '11

I've installed Kamailio on Ubuntu 10.04 LTS via: http://www.kamailio.org/dokuwiki/doku.php/packages:debs Updated /etc/kamailio/kamctlrc with: SIP_DOMAIN=MY-IP DBENGINE=MYSQL I've also installed the kamailio-mysql-modules via apt-get. Then I've run kamdbctl create and created the MySQL database. After that I've started Kamailio via kamctl start. Success. At last I've added two users: kamctl add 100 PASS1 kamctl add 101 PASS2 Everything seems to work, but I've found out that everyone can register and call via the server. Not only the two users/subscribers I've added. How come authorization isn't working?

4 6

Kamailio 1.5: replying a 200 (OPTIONS) with SDP
by Iñaki Baz Castillo 24 Jul '11

24 Jul '11

Hi, I've a stupid SIP PBX (Alcatel) which after registering sends a periodic OPTIONS with no RURI username and with "Accept: application/sdp". My proxy replies 200 in this case, but without SDP body (how could I know which codecs to show???) and unfortunately the Alcatel does not like it and send retransmissions of the requests (sad but true). After a while Alcatel determines that the proxy is down (pathetic but true). So I must do a hack and reply 200 with SDP from Kamailio when the request is an OPTIONS and contains "Accept: application/sdp". I don't want to forward the OPTIONS to PSTN gateways behind Kamailio for different reasons. Which would be the best way to do it? is it possible to add a spoofed body to a response replied stateless from Kamailio? Note: options_reply() in siputils module is not valid as AFAIK it does not show a SDP. Thanks a lot. -- Iñaki Baz Castillo <ibc(a)aliax.net>

2 6

Avoiding to send 603 DECLINED to un-trusted IPs - Is that possible with Kamilio?
by Bruce B 23 Jul '11

23 Jul '11

Hi everyone, Brand new to kamailio and was told that Asterisk is not able to NOT send 603 DECLINED for anonymous SIP Invites. Is it possible to set Kamilio to absorb SIP invites from all over the world and only respond to those requests whose IPs are registered as PEERS and to not respond back AT ALL to those IPs that are not registered peers? Thanks, Bruce

1 0

Siremis & Asterisk
by Roman Dmytriv 22 Jul '11

22 Jul '11

Hello, I have my Kamailio super node up and running on Debian 6.0 machine and have few questions: I'm planning to install Asterisk and Siremis on top of it. Which I should do first if it matters? Do you have any guides documentations on how to do it? I was testing Jitsi and found out that I can only use IP for registrar not FQDN. For proxy I can use FQDN no problem. What could possibly cause that? Regards Roman

2 2

dispatcher - call load distribution algorithm
by Giuseppe Carella 21 Jul '11

21 Jul '11

Hi all, I'm using kamailio like dispatcher, for INVITE requests. The client is the IMS bench as UAC, and the backend node are sip servlet container, in which is running an UAS servlet. Between the client and kamailio there is the OpenIMSCore. I was using the hash over callid like algorithm, and all was going well. Now I'm trying to use the last algorithm, based on the call load of the destination backend, and the ACK message of the same dialog is sent to another instance. I read on the documentation that there is no SIP dialog tracking depending on other modules. This is my configuration: # main request routing block route { if ( !mf_process_maxfwd_header("10") ) { sl_send_reply("483","To Many Hops"); exit; } # select from first dst group by round-robin if(!ds_select_dst("1", "10")) { sl_send_reply("500", "No destination available"); exit; } t_on_failure("RTF_DISPATCH"); if(!t_relay()) { sl_reply_error(); exit; } } # dispatcher failure routing block failure_route[RTF_DISPATCH] { if (t_is_canceled()) { exit; } # select next destination only for local timeout if (t_branch_timeout() & !t_branch_replied()) { if(ds_next_dst()) { t_on_failure("RTF_DISPATCH"); t_relay(); exit; } } } Thanks and best regards, Giuseppe

1 0

Kamailio presence server problem with PostgreSQL
by Roberto Fichera 21 Jul '11

21 Jul '11

Hi all, I've setup for testing purpose an xcap server with Kamailio v3.1.4 backed with a postgresql v8.4.8 all in 64bits versions. On that configuration, I'm getting the errors below and consequently the presence doesn't correctly reports the status. Does anyone has the same problem? Best regards, Roberto Fichera. Jul 19 10:31:52 roberto /usr/sbin/kamailio[2005]: INFO: pua_usrloc [ul_publish.c:211]: should not send ul publish Jul 19 10:31:52 roberto /usr/sbin/kamailio[2015]: ERROR: <script>: ===== xhttp: xcap-caps : <null> Jul 19 10:31:52 roberto /usr/sbin/kamailio[2017]: ERROR: <script>: ===== xhttp: resource-lists : sip:roberto@192.168.2.92 Jul 19 10:31:52 roberto /usr/sbin/kamailio[2017]: ERROR: <script>: ===== xhttp: get sip:roberto@192.168.2.92 => /xcap-root/resource-lists/users/sip:roberto@192.168.2.92/index Jul 19 10:31:52 roberto /usr/sbin/kamailio[2021]: ERROR: <script>: ===== xhttp: pres-rules : sip:roberto@192.168.2.92 Jul 19 10:31:52 roberto /usr/sbin/kamailio[2021]: ERROR: <script>: ===== xhttp: get sip:roberto@192.168.2.92 => /xcap-root/pres-rules/users/sip:roberto@192.168.2.92/presrules Jul 19 10:31:52 roberto /usr/sbin/kamailio[1999]: ERROR: presence [subscribe.c:711]: Missing or unsupported event header field value Jul 19 10:31:52 roberto /usr/sbin/kamailio[1999]: ERROR: presence [subscribe.c:714]: #011event= message-summary Jul 19 10:31:52 roberto /usr/sbin/kamailio[2004]: INFO: presence [notify.c:1588]: NOTIFY sip:roberto@192.168.2.92 via sip:roberto@192.168.2.92:5090;transport=udp;registering_acc=192_168_2_92 on behalf of sip:roberto@192.168.2.92 for event presence.winfo Jul 19 10:31:52 roberto /usr/sbin/kamailio[2005]: INFO: presence [notify.c:1588]: NOTIFY sip:roberto@192.168.2.92 via sip:roberto@192.168.2.92:5090;transport=udp;registering_acc=192_168_2_92 on behalf of sip:mattia@192.168.2.92 for event presence *Jul 19 10:31:52 roberto /usr/sbin/kamailio[1999]: ERROR: db_postgres [km_dbase.c:427]: invalid query, execution aborted Jul 19 10:31:52 roberto /usr/sbin/kamailio[1999]: ERROR: db_postgres [km_dbase.c:428]: driver error: PGRES_FATAL_ERROR, ERROR: null value in column "sender" violates not-null constraint#012* *Jul 19 10:31:52 roberto /usr/sbin/kamailio[1999]: WARNING: db_postgres [km_dbase.c:473]: unexpected result returned*

5 17

xcap server - http load–balancer: xcap authentication problem
by laura testi 21 Jul '11

21 Jul '11

Hello all, I’m using two Kamailio Presence Servers that are also XCAP Servers for the presence and resources management and the Kamailio Dispatcher in order to balance the SIP requestes coming from SIP Client. Then I’m trying to configure a HTTP load–balancer for the xcap/http traffic. I’m testing both Apache and INginx HTTP proxy but I’m facing with a problem related with the Xcap authentication: when a client sends the first xcap request to the http load-balancer, it forwards the request to one of the configured Xcap Server that replies with the 401 response in order to let the client authenticates. So, the client sends the second request with the Digest authentication to the http load-balancer. But, sometimes, the http load-balancer forwards the authenticate request to the second Xcap Server that is not able to authenticate the request. Then the client receive the “Unauthorized” response… Could you please give us some hints about this problem? Thanks in advance, laura

4 11

dns_srv_lb
by Jon Bonilla 21 Jul '11

21 Jul '11

Hi all I'm running a kamailio 3.1 based proxy which actually relays REGISTER requests to a domain "example.com" I've setup a DNS SRV record for that domain which returns two hosts with same prio and weight. I want kamailio to balance the requests to both servers 50%. Taking a look to ./doc/dns.txt I've setup these configuration options: dns_try_ipv6=no dns_retr_time=1 dns_retr_no=1 dns_use_search_list=no use_dns_cache=on use_dns_failover=on dns_srv_lb=on If I understood right the last setting would have to make kamailio work as I described, balancing the requests between both servers. But my logs say that kamailio is actually using only one of the ip addresses (host2.example.com) to send the REGISTER requests. Here's the output of sercmd dns.view: sercmd> dns.view { name: _sip._udp.example.com type: SRV size (bytes): 232 reference counter: 1 permanent: no expires in (s): 31364 last used (s): 446 negative entry: no rr name: host1.example.com rr port: 5060 rr priority: 0 rr weight: 0 rr expires in (s): 31364 rr name: host2.example.com rr port: 5060 rr priority: 0 rr weight: 0 rr expires in (s): 31364 } { name: host2.example.com type: A size (bytes): 128 reference counter: 1 permanent: no expires in (s): 592964 last used (s): 446 negative entry: no rr ip: 192.168.0.103 rr expires in (s): 592964 } { name: host1.example.com type: A size (bytes): 128 reference counter: 1 permanent: no expires in (s): 592964 last used (s): 11835 negative entry: no rr ip: 192.168.0.102 rr expires in (s): 592964 } Any suggestions? What am I doing wrong? Thanks, Jon

2 2

Call is terminated in callee side after 36s @Kamailio 3.1 @debian sqeeze64
by Coca 21 Jul '11

21 Jul '11

Hi List, I have a Kamailio3.1 server and RTPProxy running in WAN. The calls between UA will automatically terminated in Callee UA side 36s after connected, while no one sends a BYE. While Kamailio and UA are in LAN at all , everything is just working well. Is it my rtpproxy doesn't working correctly or something else is wrong? What can I do to fix it. Any hint?? BTW, Kamailio is installed following official guide http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.1.x-from-git The kamailio.cfg wasn't changed at all except for below: ----------------------------------------------------------------------------------------- 1) adding the following lines: #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_NAT 2)uncommenting the line below in route[REGISTRAR], setbflag(FLB_NATSIPPING); 3)change rtpproxy port corresponding modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:22222") ----------------------------------------------------------------------------------------- and my rtpproxy1.2.1 was installed by apt-get install rtpproxy, with 1) /etc/default/rtpproxy changed into: # Defaults for rtpproxy # The control socket. #CONTROL_SOCK="unix:/var/run/rtpproxy/rtpproxy.sock" # To listen on an UDP socket, uncomment this line: CONTROL_SOCK=udp:127.0.0.1:22222 LISTEN_ADDR=xx.xx.xx.xx # Additional options that are passed to the daemon. EXTRA_OPTS="-l ${LISTEN_ADDR}" 2) and started by rtpproxy -l xx.xx.xx.xx -s udp:localhost:22222 -u kamailio Your help will be great appreciated. Coca

1 0

Kamailio: 500 Server error occurred (19/SL).
by Max Doronin 21 Jul '11

21 Jul '11

Hello guys, I have strange behavior on my Kamailio-1.5.1-notls instance. I have LCR and TM modules enabled.For a particular destination I have 4 routes to try. When I call to non-existing number, it tries - route 1 (404 Not Found) - route 2 (404 Not Found) - route 3 (503 Service unavailable) And the problem is with the last 503. Kamailio ACKs it, immediately drops these 2 lines into syslog Jul 13 04:14:47 /usr/local/sbin/kamailio[23572]: ERROR:tm:t_forward_nonack: no branch for forwarding Jul 13 04:14:47 /usr/local/sbin/kamailio[23572]: ERROR:tm:w_t_relay: t_forward_nonack failed And sends SIP/2.0 500 Server error occurred (19/SL). To the call originator. 30 microseconds later it sends SIP/2.0 500 Service Unavailable. to the originator again Latter 500 looks like the relayed original "503 Service unavailable". I think so because the Reason header is the same: Reason: SIP;cause=503;text="Service Unavailable";icodetext="NoCircuitAvailable";iintcode=10034;isubsystem=3. Obviously call is terminated. I noticed that that last peer rewrites my Via headers like this: Original: == Via: SIP/2.0/UDP 187.45.214.132;branch=z9hG4bK3dac.fae9b793.2. Via: SIP/2.0/UDP 192.168.2.33:5622;received=49.49.59.23;branch=z9hG4bK-d8754z-a6ba40014f97fb7c-1---d8754z-;rport=23832. == Rewritten == Via: SIP/2.0/UDP 187.45.214.132;received=187.45.214.132;branch=z9hG4bK3dac.fae9b793.2,SIP/2.0/UDP 192.168.2.33:5622;received=49.49.59.23;branch=z9hG4b == I tried to modify via1_matching parameter (1 -> 0) but no changes. My questions are: - What can be the reason of that 500 with 19/SL - Can kamailio properly handle this 1 line Via header? - What can be the reason of that "t_forward_nonack: no branch for forwarding"? Regards, Max

2 4

(no subject)
by Eliezer Croitoru 20 Jul '11

20 Jul '11

1 0

Re: [SR-Users] IMC module unresolved issues
by Ozren Lapcevic 20 Jul '11

20 Jul '11

Hi, did anyone run into similar problems or have successfully set up IMC module with mysql? Can you please share your know-how on the subject? Besides issue described below, I also had several other issues, but I think they all might be related to the fact that my rooms and members are only stored in cache, but not in the database. To repeat, this is really strange behavior because all other modules that I'm using write to the database without a problem. Best regards Ozren On Mon, May 30, 2011 at 6:28 PM, Ozren Lapcevic <ozren.lapcevic(a)gmail.com>wrote: > Hi all, > > I'm trying to configure IMC module and get it to work properly but I've run > into problems with saving IMC module data to mysql database. > > I'm using following code provided in IMC module documentation: > > modparam("imc", "db_url", DBURL) > > ## route block ### > if(is_method("MESSAGE") && (uri=~ "sip:chat-[0-9]+@") || (uri=~ > "sip:chat-manager@")) { > xlog("L_INFO", "going to imc: F [$fu] T [$tu] B [$rb]\n"); > > if(imc_manager()) > sl_send_reply("200", "ok - imc_manager()"); > else > sl_send_reply("500", "command error"); > exit; > } > > I am creating chat room with following message sent to chat-manager(a)abc.hr: > "#create chat-111" > > DEBUG log returns following: > May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]: > INFO: <script>: going to imc: F [sip:oz@abc.hr] T [sip:chat-manager@abc.hr] > B [#create chat-111] > May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]: > DEBUG: imc [imc.c:525]: found command > May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]: > DEBUG: imc [imc_cmd.c:146]: command: [create] > May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]: > DEBUG: imc [imc_cmd.c:151]: parameter 0=[chat-111] > May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]: > DEBUG: imc [imc_cmd.c:171]: new room [chat-111] > May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]: > DEBUG: imc [imc_cmd.c:186]: added room uri= sip:chat-111@abc.hr > May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]: > DEBUG: imc [imc_mng.c:322]: [uri]= sip:oz@abc.hr > May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]: > DEBUG: imc [imc_mng.c:326]: [user]= oz > May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]: > DEBUG: imc [imc_cmd.c:196]: added the owner as the first member [ > sip:oz@abc.hr] > > "./kamctl fifo imc_list_rooms" returns following: > ROOM:: URI=sip:chat-111@abc.hr MEMBERS=0 OWNER=sip:oz@abc.hr > > "./kamctl fifo imc_list_members sip:chat-111@abc.hr" returns following: > ROOM:: sip:chat-111@abc.hr NR_OF_MEMBERS=1 > MEMBER:: sip:oz@abc.hr > > However, that data is not persisted/saved in the imc_rooms and imc_members > tables in the database. Any clues what might be the problem here? > > Any help is greatly appreciated. > > Best regards, > Ozren Lapcevic > > PS > I'm runing Kamailio 3.1.4 with default cfg file with enabled mysql, > multidomain, usrlocdb, auth and NAT. I can register clients, establish > calls, send IMs, etc. imc_rooms and imc_members tables were created when > kamailio was installed. I've also checked imc-create.sql script and tables > are correctly created. > > The same DBURL used for imc is also used for auth_db, domain and usrloc > modules and I haven't experienced any problems with those modules and > database connection/persistence. domain, location, version and subscriber > tables are correctly populated. > > I'm not getting any errors from imc_manager(), it returns 200 OK. I can't > trace any errors in DEBUG related to problems with mysql database. >

2 7

how do i set up outbound on iptel
by Bill Peebles 20 Jul '11

20 Jul '11

i really want to find out how. I dont have a home phone.

2 1

Project conference in Berlin, Sep 02, 2011
by Daniel-Constantin Mierla 20 Jul '11

20 Jul '11

Hello, the details for organizing the 10 years SER conference in Berlin were settled. After the initial announcement the interest was quite high, by now we have a good number of attendees and speakers. We are still open for new speakers, with the target of showing use cases of Kamailio or SER. The event will be hosted by FhG Fokus institute, where SER project was started. It will be a day full of presentations, followed by an evening social networking event. We hope to be able to provide the evening event for free to everyone attending, for that reason we look for sponsorships - if you are interested to sponsor, contact us and we will provide more details about available options. Registration is now open for attendees. Note that the number of seats is limited, don't wait for the last minute to register. More details, including the registration form can be accessed on the event's web page: * http://sip-router.org/10-years-ser/ Looking forward to meeting you in Berlin, enjoy a great summer till then, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

Pipelimit Table
by Nathaniel L Keeling 20 Jul '11

20 Jul '11

I have installed Kamailio 3.1.0 and I am trying to test the pipelimit module with postgresql. I ran the kamdbctl to create the tables but it did not create the default table for pipelimit. Is there a script someplace that I am missing or do I need to create my own script to create the tables? I have search the directories and did not find a script for the tables. Thanks Nathaniel L Keeling

2 1

Change aliases table without a restart of Kamailio
by Morten Isaksen 19 Jul '11

19 Jul '11

Hi, Is there any way to reload the content of the aliases table without restarting Kamailio? I have this in my cfg file if (lookup("location")) { xlog("L_WARN", "Callee is online - - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(TO_USER_SIPCHECKS); route(RELAY); } else if (lookup("aliases")) { xlog("L_WARN", "Callee is in aliases - - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(TO_USER_SIPCHECKS); route(RELAY); } else { xlog("L_WARN", "Callee is offline - - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); t_newtran(); t_reply("486", "Busy here"); exit; } And changes in the aliases table does is ignored until I restart Kamailio. -- Morten Isaksen

2 5

Syntax error - custom parameter
by Skyler 19 Jul '11

19 Jul '11

Hi all, I'm getting this error when using custom parameters and I don't why. when I use #!define WITH_ASTERISK the script throws the below error and when I remove the #!define WITH_ASTERISK the script loads without errors. I'm running kamailio-3.1.4-2.1.i386.rpm on Centos 5.6 Does anyone know why this might be happening? parse error in config file //etc/kamailio/kamailio.cfg, line 123, column 16: syntax error 0(2907) : <core> [cfg.y:3412]: parse error in config file //etc/kamailio/kamailio.cfg, line 123, column 16: unknown config variable 0(2907) : <core> [cfg.y:3409]: parse error in config file //etc/kamailio/kamailio.cfg, line 123, column 17-19: ERROR: bad config file (3 errors) kamailio.cfg::: lines 122-127 #!ifdef WITH_ASTERISK asterisk.bindip = "127.0.0.1" desc "Asterisk IP Address" asterisk.bindport = "5070" desc "Asterisk Port" kamailio.bindip = "209.15.226.21" desc "Kamailio IP Address" kamailio.bindport = "5060" desc "Kamailio Port" #!endif # This follows as described here: http://by-miconda.blogspot.com/2010/04/kamailio-30x-dyk-07-custom-cfg.html Thanks in advance

2 2

Kamailio as proxy or b2bua?
by Eric Chamberlain 19 Jul '11

19 Jul '11

Hello, My experience has been with Asterisk, but for what I need to do, I think Kamailio may be a better solution. We currently log inbound caller ID information for voip calls (voip app posts call info to a URL). Our users have to fork calls to a custom SIPURI and we log the call information for them. We would like to be able to accommodate users that can't fork calls, and have to register with a provider (PBX or VoIP provider). We see two ways to accomplish this: 1) Act as a b2bua, the user would register with our server and our server would register with their upstream VoIP service (PBX or provider). or 2)Act as a proxy and modify the REGISTER request, such that incoming call signaling traffic goes through our proxy and we forward the signaling traffic on to the user's device. Can Kamailio easily do either of these options? Asterisk realtime doesn't support realtime changes of remote registration information, can Kamailio? Would proxying in such a manner be supported my most voip devices and service providers? -- Eric Chamberlain

2 1

siremis 2.0: generates CDRs
by Klaus Darilion 19 Jul '11

19 Jul '11

Hi! Seems like I am blind. There is a stored procedure which generates CDRs from acc table entries - but how/when does this stored procedure gets triggered? Thanks Klaus

2 5

MYSQL Compilation Kamailio 3.1.4
by Omar 16 Jul '11

16 Jul '11

bash-3.00# make clean bash-3.00# make quiet=0 /opt/SUNWgccfss/gcc/bin/gcc -m64 -fPIC -DPIC -g -O9 -funroll-loops -m64 -mcpu=ultrasparc -mtune=ultrasparc -fno-strict-overflow -ftree-vectorize -Wall -DNAME='"kamailio"' -DVERSION='"3.1.4"' -DARCH='"sparc64"' -DOS='solaris_' -DOS_QUOTED='"solaris"' -DCOMPILER='"/opt/SUNWgccfss/gcc/bin/gcc 4.2.0"' -D__CPU_sparc64 -D__OS_solaris -DSER_VER=3001004 -DCFG_DIR='"/opt/kamailio3/etc/kamailio/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST -DUSE_NAPTR -DDBG_QM_MALLOC -DUSE_TLS -DTLS_HOOKS -DUSE_CORE_STATS -DSTATISTICS -DMALLOC_STATS -DWITH_AS_SUPPORT -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM -DSPARC64_MODE -DHAVE_GETIPNODEBYNAME -DHAVE_SYS_SOCKIO_H -DHAVE_SCHED_YIELD -DHAVE_ALLOCA_H -DUSE_SIGACTION -DHAVE_DEVPOLL -DHAVE_SELECT -DHAVE_FILIO_H -DSER_MOD_INTERFACE -DSER_MOD_INTERFACE -I/opt/openser/include -I/opt/openser/include/mysql -I/opt/openser/mysql/include -I/usr/include/mysql -DMOD_NAME='"db_mysql"' -c km_db_mysql.c -o km_db_mysql.o -MMD -MP /opt/SUNWgccfss/gcc/bin/gcc -m64 -fPIC -DPIC -g -O9 -funroll-loops -m64 -mcpu=ultrasparc -mtune=ultrasparc -fno-strict-overflow -ftree-vectorize -Wall -DNAME='"kamailio"' -DVERSION='"3.1.4"' -DARCH='"sparc64"' -DOS='solaris_' -DOS_QUOTED='"solaris"' -DCOMPILER='"/opt/SUNWgccfss/gcc/bin/gcc 4.2.0"' -D__CPU_sparc64 -D__OS_solaris -DSER_VER=3001004 -DCFG_DIR='"/opt/kamailio3/etc/kamailio/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST -DUSE_NAPTR -DDBG_QM_MALLOC -DUSE_TLS -DTLS_HOOKS -DUSE_CORE_STATS -DSTATISTICS -DMALLOC_STATS -DWITH_AS_SUPPORT -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM -DSPARC64_MODE -DHAVE_GETIPNODEBYNAME -DHAVE_SYS_SOCKIO_H -DHAVE_SCHED_YIELD -DHAVE_ALLOCA_H -DUSE_SIGACTION -DHAVE_DEVPOLL -DHAVE_SELECT -DHAVE_FILIO_H -DSER_MOD_INTERFACE -DSER_MOD_INTERFACE -I/opt/openser/include -I/opt/openser/include/mysql -I/opt/openser/mysql/include -I/usr/include/mysql -DMOD_NAME='"db_mysql"' -c my_uri.c -o my_uri.o -MMD -MP /opt/SUNWgccfss/gcc/bin/gcc -m64 -fPIC -DPIC -g -O9 -funroll-loops -m64 -mcpu=ultrasparc -mtune=ultrasparc -fno-strict-overflow -ftree-vectorize -Wall -DNAME='"kamailio"' -DVERSION='"3.1.4"' -DARCH='"sparc64"' -DOS='solaris_' -DOS_QUOTED='"solaris"' -DCOMPILER='"/opt/SUNWgccfss/gcc/bin/gcc 4.2.0"' -D__CPU_sparc64 -D__OS_solaris -DSER_VER=3001004 -DCFG_DIR='"/opt/kamailio3/etc/kamailio/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST -DUSE_NAPTR -DDBG_QM_MALLOC -DUSE_TLS -DTLS_HOOKS -DUSE_CORE_STATS -DSTATISTICS -DMALLOC_STATS -DWITH_AS_SUPPORT -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM -DSPARC64_MODE -DHAVE_GETIPNODEBYNAME -DHAVE_SYS_SOCKIO_H -DHAVE_SCHED_YIELD -DHAVE_ALLOCA_H -DUSE_SIGACTION -DHAVE_DEVPOLL -DHAVE_SELECT -DHAVE_FILIO_H -DSER_MOD_INTERFACE -DSER_MOD_INTERFACE -I/opt/openser/include -I/opt/openser/include/mysql -I/opt/openser/mysql/include -I/usr/include/mysql -DMOD_NAME='"db_mysql"' -c my_res.c -o my_res.o -MMD -MP /opt/SUNWgccfss/gcc/bin/gcc -m64 -fPIC -DPIC -g -O9 -funroll-loops -m64 -mcpu=ultrasparc -mtune=ultrasparc -fno-strict-overflow -ftree-vectorize -Wall -DNAME='"kamailio"' -DVERSION='"3.1.4"' -DARCH='"sparc64"' -DOS='solaris_' -DOS_QUOTED='"solaris"' -DCOMPILER='"/opt/SUNWgccfss/gcc/bin/gcc 4.2.0"' -D__CPU_sparc64 -D__OS_solaris -DSER_VER=3001004 -DCFG_DIR='"/opt/kamailio3/etc/kamailio/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST -DUSE_NAPTR -DDBG_QM_MALLOC -DUSE_TLS -DTLS_HOOKS -DUSE_CORE_STATS -DSTATISTICS -DMALLOC_STATS -DWITH_AS_SUPPORT -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM -DSPARC64_MODE -DHAVE_GETIPNODEBYNAME -DHAVE_SYS_SOCKIO_H -DHAVE_SCHED_YIELD -DHAVE_ALLOCA_H -DUSE_SIGACTION -DHAVE_DEVPOLL -DHAVE_SELECT -DHAVE_FILIO_H -DSER_MOD_INTERFACE -DSER_MOD_INTERFACE -I/opt/openser/include -I/opt/openser/include/mysql -I/opt/openser/mysql/include -I/usr/include/mysql -DMOD_NAME='"db_mysql"' -c my_cmd.c -o my_cmd.o -MMD -MP In file included from ../../globals.h:37, from ../../pt.h:42, from ../../counters.h:43, from mysql_mod.h:39, from my_cmd.c:40: ../../ip_addr.h:86: error: field 'sin6' has incomplete type ../../ip_addr.h: In function 'ip_addr_loopback': ../../ip_addr.h:283: warning: implicit declaration of function 'htonl' ../../ip_addr.h:286: warning: implicit declaration of function 'IN6_IS_ADDR_LOOPBACK' ../../ip_addr.h: In function 'sockaddr2ip_addr': ../../ip_addr.h:353: error: dereferencing pointer to incomplete type ../../ip_addr.h: In function 'su_getport': ../../ip_addr.h:399: warning: implicit declaration of function 'ntohs' ../../ip_addr.h: In function 'su_setport': ../../ip_addr.h:418: warning: implicit declaration of function 'htons' make: *** [my_cmd.o] Error 1 the onyl thing i did is uncomment the CROSSCOMPILE line regards O

4 10

Re: [SR-Users] sr-users Digest, Vol 74, Issue 41
by Henrik Aagaard Sørensen 15 Jul '11

15 Jul '11

On Fri, Jul 15, 2011 at 12:00 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. dbaliases for ring groups, possible? (Henrik Aagaard S?rensen) > 2. Re: dbaliases for ring groups, possible? (Alex Balashov) > 3. Re: dbaliases for ring groups, possible? (Morten Isaksen) > 4. Re: xcap server - http load?balancer: xcap authentication > problem (laura testi) > 5. Re: bypass rtp traffic. (MingHon) > 6. Re: AVPs missing in onreply_route (Sebastian Damm) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Fri, 15 Jul 2011 09:41:16 +0200 > From: Henrik Aagaard S?rensen <henrikaagaardsorensen(a)gmail.com> > Subject: [SR-Users] dbaliases for ring groups, possible? > To: sr-users(a)lists.sip-router.org > Message-ID: > <CAGH8SebPV=xz1n6Jp=M5fjnUhK0t=Za2aWd_jrpzbjdpdejatw(a)mail.gmail.com > > > Content-Type: text/plain; charset="iso-8859-1" > > I've read somewhere about using dbaliases for ring groups. So I can call > one > number and several different numbers will then be called. The first to pick > up the call gets it. > > I've tried to an alias in my dbaliases table in MySQL, but I can only make > it possible to call different numbers for the same user. In a way the other > way around then what I thought. > > Is it somehow possible to call one number and then having Kamailio call > several different numbers /users? > > As I'm very newbee to Kamailio it would be nice with a more detailed > answer, > then perhaps just "Yes, use the module ...". >

2 1

Dispatcher OPTIONS ping not seen in onsend_route
by Olle E. Johansson 15 Jul '11

15 Jul '11

Hello! I noticed that the dispatcher's OPTIONs messages are not seen in onsend_route, even though I see responses in the onreply_route. Any reason why I can't see these in onsend_route? (Kamailio 3.1) /O

2 1

dbaliases for ring groups, possible?
by Henrik Aagaard Sørensen 15 Jul '11

15 Jul '11

I've read somewhere about using dbaliases for ring groups. So I can call one number and several different numbers will then be called. The first to pick up the call gets it. I've tried to an alias in my dbaliases table in MySQL, but I can only make it possible to call different numbers for the same user. In a way the other way around then what I thought. Is it somehow possible to call one number and then having Kamailio call several different numbers /users? As I'm very newbee to Kamailio it would be nice with a more detailed answer, then perhaps just "Yes, use the module ...".

3 2

Re: [SR-Users] sr-users Digest, Vol 74, Issue 38
by Henrik Aagaard Sørensen 14 Jul '11

14 Jul '11

On Thu, Jul 14, 2011 at 9:51 AM, <sr-users-request(a)lists.sip-router.org>wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > > You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. Re: memory requirements for a registrar server > (Henning Westerholt) > 2. Re: kamailio - proxy/relay (Sunny) > 3. Syntax error - custom parameter (Skyler) > 4. Syntax error - custom parameter (Skyler) > 5. Re: kamailio - proxy/relay (I?aki Baz Castillo) > 6. Re: topoh module - query about operation (Pete Kelly) > 7. Re: Call several different subscribers, first to pickup gets > the call. (Klaus Darilion) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Thu, 14 Jul 2011 00:27:18 +0200 > From: Henning Westerholt <Henning.Westerholt(a)1und1.de> > Subject: Re: [SR-Users] memory requirements for a registrar server > To: Fabian Borot <fborot(a)hotmail.com> > Cc: "sr-users(a)lists.sip-router.org" <sr-users(a)lists.sip-router.org> > Message-ID: > <D3B36E097FE87E44A5CDE7C1AB61682C3CED2EC9(a)EXCHANGE03.webde.local> > Content-Type: text/plain; charset="us-ascii" > > Fabian wrote: > > ok, great...this is, I think, the relevant section on config.h > > Hi Fabian, > > > #define PKG_MEM_SIZE 4 > > #define PKG_MEM_POOL_SIZE PKG_MEM_SIZE*1024*1024 /*!< used only if > PKG_MALLOC is defined*/ > > #define SHM_MEM_SIZE 32 /*!< used if SH_MEM is > defined*/ > > > > > > SHM_MEM_SIZE 32 --> this looks like 32 MB > > and PKG_MEM_SIZE 4 looks like Bytes, that is why the PKG_MEM_POOL_SIZE is > multiplied by 1024*1024 to get MBs I assume. > > Correct, 4MB and 32MB. I'll have a look to to the file to document it a bit > better in the next days. > > You don't need to change config.h to change the shared pool. There exists a > cmd line parameter, > have a look to 'man kamailio' or run 'kamailio -h' for more informations. > The specified values are > just used in the case nothing else is configured. They are also some cfg > script parameters, i think. > > Regards, > > Henning > > > ------------------------------ > > Message: 2 > Date: Wed, 13 Jul 2011 23:39:57 +0100 > From: Sunny <no7find(a)gmail.com> > Subject: Re: [SR-Users] kamailio - proxy/relay > To: miconda(a)gmail.com > Cc: "SIP Router - Kamailio $OpenSER$ and SIP Express Router $SER$ > - Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: > <CAE_sWtEE4qv5u7bBWyhMSAjMWYB6hKpO6G=_DGyoCK7Jt0OKEA(a)mail.gmail.com > > > Content-Type: text/plain; charset="iso-8859-1" > > Hi Daniel, > > I was using: > $ru = $(ru{re.subst,/[KAMAILIO]/[IP-PBX ]/}); > t_relay_to_udp("[IP-PBX ]", "5060"); > but your way looks better. > > This works fine in this direction: > [UA] --> [KAMAILIO] --> [IP-PBX ] > > but not in this: > [UA] <-- [KAMAILIO] <-- [IP-PBX ] > because Kamailio is not keeping the location of the UA. > > I want to register UA's on Kamailio, but the authentication is done by > [IP-PBX]... > So Kamailio as Outbound Proxy + Location Server ... any idea on how I can > achieve this configuration? > > Thanks, > Sunny > > > > On 8 July 2011 09:38, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: > > > Hello, > > > > > > On 7/5/11 8:46 PM, Sunny wrote: > > > >> Hi list, > >> > >> I'm trying to configure kamailio to work as a OBP (Outbound Proxy). > >> > >> Basically KAMAILIO receive requests from UAs, after that it acts like a > >> client and send those requests to the IP-PBX. > >> On the same way the responses are forwarded from IP-PBX to the UAs. > >> > >> > >> [UA] --> [KAMAILIO] --> [IP-PBX ] > >> [UA] <-- [KAMAILIO] <-- [IP-PBX ] > >> > >> Following possible call flow: > >> > >> [UA ] -> [KAMAILIO] SIP/SDP Request: INVITE sip:12345678909 > @[KAMAILIO], > >> with session description > >> [KAMAILIO] -> [UA ] SIP Status: 100 Trying > >> [KAMAILIO] -> [IP-PBX ] SIP/SDP Request: INVITE sip:12345678909 > @[IP-PBX > >> ], with session description > >> [IP-PBX ] -> [KAMAILIO] SIP Status: 100 Trying > >> [IP-PBX ] -> [KAMAILIO] SIP Status: 407 Proxy Authentication Required > >> [KAMAILIO] -> [IP-PBX ] SIP Request: ACK sip:12345678909@[IP-PBX ] > >> [KAMAILIO] -> [UA ] SIP Status: 407 Proxy Authentication Required > >> [UA ] -> [KAMAILIO] SIP Request: ACK sip:12345678909@[KAMAILIO] > >> [UA ] -> [KAMAILIO] SIP/SDP Request: INVITE sip:12345678909 > @[KAMAILIO], > >> with session description > >> [KAMAILIO] -> [UA ] SIP Status: 100 Trying > >> [KAMAILIO] -> [IP-PBX ] SIP/SDP Request: INVITE sip:12345678909 > @[IP-PBX > >> ], with session description > >> [IP-PBX ] -> [KAMAILIO] SIP Status: 100 Trying > >> [IP-PBX ] -> [KAMAILIO] SIP/SDP Status: 183 Session Progress, with > >> session description > >> [KAMAILIO] -> [UA ] SIP/SDP Status: 183 Session Progress, with > >> session description > >> [IP-PBX ] -> [KAMAILIO] SIP/SDP Status: 180 Ringing, with session > >> description > >> [KAMAILIO] -> [UA ] SIP/SDP Status: 180 Ringing, with session > >> description > >> (...) > >> [UA ] -> [KAMAILIO] SIP Request: CANCEL sip:12345678909@[KAMAILIO] > >> [KAMAILIO] -> [UA ] SIP Status: 200 OK > >> [KAMAILIO] -> [IP-PBX ] SIP Request: CANCEL sip:12345678909@[IP-PBX ] > >> [IP-PBX ] -> [KAMAILIO] SIP Status: 200 OK > >> [IP-PBX ] -> [KAMAILIO] SIP Status: 487 Request Terminated > >> [KAMAILIO] -> [IP-PBX ] SIP Request: ACK sip:12345678909@[IP-PBX ] > >> [KAMAILIO] -> [UA ] SIP Status: 487 Request Terminated > >> [UA ] -> [KAMAILIO] SIP Request: ACK sip:12345678909@[KAMAILIO] > >> > >> Wondering if anyone could help to make this configuration work? > >> > > > > what I understood from above is that you need to update the domain in > R-URI > > from [kamailio] to [ip-pbx]. This is easy to do: > > > > $rd = [ip-pbx]; > > > > You have to do this for initial requests (register, invite). > > > > If you need to update From header URI, look at uac module, > > uac_replace_from(). > > > > Cheers, > > Daniel > > > > -- > > Daniel-Constantin Mierla -- http://www.asipto.com > > Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat > > http://linkedin.com/in/miconda -- http://twitter.com/miconda > > > > >

1 0

Call several different subscribers, first to pickup gets the call.
by Henrik Aagaard Sørensen 14 Jul '11

14 Jul '11

I'm new to Kamailio. Is it possible to have a group of subscribers (with different extensions) that you can call? And the first to pickup the call gets it.

2 1

topoh module - query about operation
by Pete Kelly 14 Jul '11

14 Jul '11

Hi I have been researching the topoh module which I understand is stateless and can persist across a server restart. >From what I can gather it does this by adding some extra, encrypted headers to the INVITE as it passes through the proxy- is this correct? If so, can I ask how it handles situations where the UAS does not resend these headers in subsequent requests? Pete

2 2

Syntax error - custom parameter
by Skyler 13 Jul '11

13 Jul '11

Hi all, I'm getting this error when using custom parameters and I don't why. when I use #!define WITH_ASTERISK the script throws the below error and when I remove the #!define WITH_ASTERISK the script loads without errors. I'm running kamailio-3.1.4-2.1.i386.rpm on Centos 5.6 Does anyone know why this might be happening? parse error in config file //etc/kamailio/kamailio.cfg, line 123, column 16: syntax error 0(2907) : <core> [cfg.y:3412]: parse error in config file //etc/kamailio/kamailio.cfg, line 123, column 16: unknown config variable 0(2907) : <core> [cfg.y:3409]: parse error in config file //etc/kamailio/kamailio.cfg, line 123, column 17-19: ERROR: bad config file (3 errors) kamailio.cfg::: lines 122-127 #!ifdef WITH_ASTERISK asterisk.bindip = "127.0.0.1" desc "Asterisk IP Address" asterisk.bindport = "5070" desc "Asterisk Port" kamailio.bindip = "209.15.226.21" desc "Kamailio IP Address" kamailio.bindport = "5060" desc "Kamailio Port" #!endif # This follows as described here: http://by-miconda.blogspot.com/2010/04/kamailio-30x-dyk-07-custom-cfg.html Thanks in advance

1 0

memory requirements for a registrar server
by Fabian Borot 13 Jul '11

13 Jul '11

Hello We are going to use kamailio 3.1.4 for a registrar server and need to know if the default parameters [shared memory and the like] is fine or if we need to plan in advance and increase it at compile time. If so, how should we proceed about it? For subscribers and location type of information, which memory type should we be more concerned about? private memory, shared memory, memory per process etc because it looks like some modules use shared memory others private memory etc. Also , I see that the command "kamctl fifo get_statistics all" gives info about memory used etc. How should we interpret the output like this: shmem:fragments = 24 shmem:free_size = 9665456 shmem:max_used_size = 23915072 shmem:real_used_size = 23888976 shmem:total_size = 33554432 shmem:used_size = 23648072 is this fifo function exported by a specific module? thank you in advance fborot

2 5

kamailio - mssql
by Borin 13 Jul '11

13 Jul '11

Hi, pls do not ask me why I need this..but did anyone try to use Kamailio with mssql or is it at least possible? Best Regards, Kate

3 3

Header field manipulation
by Spencer Thomason 12 Jul '11

12 Jul '11

Hello all, Is it preferable to to a check before removing a header field or to simply call remove_hf()? Are there any performance considerations? i.e. if is_present_hf("X-Account-Code") remove_hf("X-Account-Code"); vs remove_hf("X-Account-Code"); Thanks, Spencer

2 2

siremis 2.0: disable captcha
by Klaus Darilion 12 Jul '11

12 Jul '11

Hi! Is there an easy way to disable the captcha? As siremis is an admin tool, not being public accessible (at least it shouldn't be) I want to get rid of it. Thanks Klaus

2 1

siremis 2.0 statistics problem
by Klaus Darilion 12 Jul '11

12 Jul '11

Hi! I just installed siremis 2.0. I did not let siremis update the SIP DB but did it manually (to sensitive) according to: http://siremis.git.sourceforge.net/git/gitweb.cgi?p=siremis/siremis;a=blob_… But when accessing some statistics, e.g. http://mydomain/siremis/ser/charts_tm/cg=tm I get an error: SELECT time_stamp,tm_active FROM statistics ORDER BY id DESC LIM... --> Unknown column 'tm_active' in 'field list' as statistics table does not have column "tm_active". Am I doing something wrong? Thanks Klaus

2 1

siremis 2.0 documentation
by Klaus Darilion 12 Jul '11

12 Jul '11

Hi! IMO the description "Import Default Data" (Import Default Data - this imports initial data into database 'siremis') is too ambiguous (at least it successfully confused me and I had to dig into the code to find out what it really does). Suggestion: Create Siremis Tables (Create Siremis Tables - this creates the tables in the siremis DB and fills it with inital data). regards Klaus

2 1

app_lua problem setting module parameter
by Bret McDanel 12 Jul '11

12 Jul '11

In 3.1.1 this worked just fine. In 3.1.3 it is giving me an error. I am unsure what to check at this point for why it would be failing. I am doing a kamailio -c to validate the config and not actually launching the program but I do not think that should cause this. modparam("app_lua", "register", "sqlops") sqlops.so is loaded. if I remove that modparam line the config validates, however the lua script would fail because it uses sqlops. modules.lst is the same for building. While I do not think that it is lua specific, both are using liblua5.1.0, one is Ubuntu 11.04 natty (where it fails) and the other is Debian wheezy (where its working). 0(18226) WARNING: <core> [sr_module.c:578]: /usr/local/kamailio/lib64/kamailio/modules/app_lua.so: exports dlflags interface is deprecated and it will notbe supported in newer versions; consider using mod_register() instead 0(18226) : <core> [cfg.y:3412]: parse error in config file kamailio.cfg, line 262, column 41: Can't set module parameter As a side note the "notbe supported ..." line is missing a space between not and be and a newline at the end. -- Trixter aka Bret McDanel website: http://www.0xdecafbad.com pgp key: http://bit.ly/9XYK4b

4 3

rtpproxy_stream2uac
by s.maertens＠telenet.be 12 Jul '11

12 Jul '11

Hi, I am trying to get kamailio (or more specific rtpproxy) to play a wav file when a call is setup In kamailio.cfg I have added the command in the onreply_route rtpproxy_startrecording(); rtpproxy_stream2uas("/var/log/rtpproxy/message","-1"); I am sure the command is executed and given from kamailio to rtpproxy because in syslog I see following output (using dbug output of rtpproxy) rtpproxy[2648]: DBUG:handle_command: received command "P-1 024a73d95bbe016014de647e700 /var/log/rtpproxy/behappy session 487095817;1 024a73d95bbe015f14de647e700;1" rtpproxy[2648]: DBUG:doreply: sending reply "E6#012" rtpproxy[2648]: DBUG:handle_command: received command "R 024a73d95bbe016014de647e700 487095817 024a73d95bbe015f14de647e700" rtpproxy[2648]: DBUG:doreply: sending reply "0#012" The "R" command is the startrecording command and that is indeed working perfect. (recording to /var/log/rtpproxy) The "P" command is the "Playing or stream2uas" command. Seems that the message rtpproxy gets differs a bit from the one I found in the rtpproxy manual from the manual : P[args] callid play_name codecs from_tag to_tag Direction of the playback is defined by the order of the from_tag and to_tag parameters. R callid from_tag to_tag I have used the command " makeann behappy.wav behappy" to convert a wavfile. It gave me a .0 and .a file which I have placed in the /var/log/rtpproxy directory (makeann comes with the rtpproxy source and is rather a bit shy with information to say the least ;) ) There is not too much information or examples found about this so I was hoping maybe somebody on this list could help me in finding out why i'm not getting more information in syslog and why i'm not hearing the audiofile. Best regards

2 3

transformation and PVAR
by Vitaliy Aleksandrov 11 Jul '11

11 Jul '11

>From the transformations documentation: >>{s.strip,len} >>Return string after removing starting 'len' characters. Parameter 'len' can be positive integer or pseudo-variable holding a positive integer. I tried to use pvar holding len but got an error: Jul 11 14:51:37 localhost /usr/sbin/kamailio[16905]: ERROR: pv [pv_trans.c:434]: select cannot get p1 Jul 11 14:51:37 localhost /usr/sbin/kamailio[16905]: ERROR: <core> [lvalue.c:351]: non existing right pvar This is part of my config file: $avp(varname) = '1'; <---- pseudo variable holding a positive integer $avp(result) = $(fU{s.strip,$avp(varname)}); <---- s.strip How can i use variables with transformation ?

2 1

transformation and pvar
by Vitaliy Aleksandrov 11 Jul '11

11 Jul '11

>From the transformations documentation: > {s.strip,len} > Return string after removing starting 'len' characters. Parameter 'len' can > be positive integer or pseudo-variable holding a positive integer. I tried to use pvar holding len but got an error. Jul 11 14:51:37 localhost /usr/sbin/kamailio[16905]: ERROR: pv [pv_trans.c:434]: select cannot get p1 Jul 11 14:51:37 localhost /usr/sbin/kamailio[16905]: ERROR: <core> [lvalue.c:351]: non existing right pvar This is part of my config file: $avp(varname) = '1'; <---- pseudo variable holding a positive integer $avp(result) = $(fU{s.strip,$avp(varname)}); <---- s.strip How can i use variables with transformation ?

1 0

Memcached Module
by Spencer Thomason 11 Jul '11

11 Jul '11

Hello all, I'm interested in using the memcached module however it requires libmemcache instead of libmemcached. As libmemcache is no longer supported I was curious if anyone had successfully patched the module to use libmemcached instead. Also as a side note, the provided rpm spec files in the git repo erroneously contains a reference to libmemcached. Thanks, Spencer

2 1

Employment opportunity
by miklos＠iptel.org 10 Jul '11

10 Jul '11

Spam detection software, running on the system "mail.iptel.org", has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: Salute! I take up a position of HR manager in a large multinational company. Our enterprise is inseparably connected with a great number of various activities, like: - estate property - founding and abatement of enterprises - supporting in opening of banking accounts - etc. [...] Content analysis details: (23.1 points, 5.0 required) pts rule name description ---- ---------------------- -------------------------------------------------- 1.7 URIBL_BLACK Contains an URL listed in the URIBL blacklist [URIs: totaljoblists.net] 1.6 URIBL_WS_SURBL Contains an URL listed in the WS SURBL blocklist [URIs: totaljoblists.net] 1.7 URIBL_DBL_SPAM Contains an URL listed in the DBL blocklist [URIs: totaljoblists.net] 3.5 BAYES_99 BODY: Bayes spam probability is 99 to 100% [score: 0.9981] 3.2 FH_HELO_EQ_D_D_D_D Helo is d-d-d-d 3.3 RCVD_IN_PBL RBL: Received via a relay in Spamhaus PBL [2.60.89.135 listed in zen.spamhaus.org] 0.4 RCVD_IN_XBL RBL: Received via a relay in Spamhaus XBL 1.3 RCVD_IN_BL_SPAMCOP_NET RBL: Received via a relay in bl.spamcop.net [Blocked - see <http://www.spamcop.net/bl.shtml?2.60.89.135>] 1.3 RCVD_IN_RP_RNBL RBL: Relay in RNBL, https://senderscore.org/blacklistlookup/ [2.60.89.135 listed in bl.score.senderscore.com] 1.4 RCVD_IN_BRBL_LASTEXT RBL: RCVD_IN_BRBL_LASTEXT [2.60.89.135 listed in bb.barracudacentral.org] 0.7 SPF_SOFTFAIL SPF: sender does not match SPF record (softfail) 0.0 UNPARSEABLE_RELAY Informational: message has unparseable relay lines 1.0 RDNS_DYNAMIC Delivered to internal network by host with dynamic-looking rDNS 0.0 T_URIBL_BLACK_OVERLAP T_URIBL_BLACK_OVERLAP 2.0 HELO_DYNAMIC_IPADDR Relay HELO'd using suspicious hostname (IP addr 1)

1 0

Job offer match, respond to apply
by miklos＠iptel.org 09 Jul '11

09 Jul '11

Spam detection software, running on the system "mail.iptel.org", has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: Hello! I am a superintendent of Human Resources department of a large international company. Our company takes an active part in the life of its subsidiaries, for instance: \ real estate \ logistics \ private undertaking service \ etc. [...] Content analysis details: (12.1 points, 5.0 required) pts rule name description ---- ---------------------- -------------------------------------------------- 3.3 RCVD_IN_PBL RBL: Received via a relay in Spamhaus PBL [212.164.140.23 listed in zen.spamhaus.org] 1.7 URIBL_BLACK Contains an URL listed in the URIBL blacklist [URIs: totaljoblists.net] 1.6 URIBL_WS_SURBL Contains an URL listed in the WS SURBL blocklist [URIs: totaljoblists.net] 1.7 URIBL_DBL_SPAM Contains an URL listed in the DBL blocklist [URIs: totaljoblists.net] 1.4 RCVD_IN_BRBL_LASTEXT RBL: RCVD_IN_BRBL_LASTEXT [212.164.140.23 listed in bb.barracudacentral.org] 0.0 UNPARSEABLE_RELAY Informational: message has unparseable relay lines 1.5 BAYES_60 BODY: Bayes spam probability is 60 to 80% [score: 0.6047] 0.0 T_URIBL_BLACK_OVERLAP T_URIBL_BLACK_OVERLAP 0.8 RDNS_NONE Delivered to internal network by a host with no rDNS

1 0

on_reply and failure route names
by Fabian Borot 08 Jul '11

08 Jul '11

Hi Is it possible to use string names also for on_reply and failure_route blocks? I guess these errors when starting kamailio Jul 8 10:00:25 registrar kamailio: : <core> [cfg.y:3412]: parse error in config file /usr/local/kamailio-3.1//etc/kamailio//kamailio.cfg, line 432, column 41: syntax error Jul 8 10:00:25 registrar kamailio: : <core> [cfg.y:3412]: parse error in config file /usr/local/kamailio-3.1//etc/kamailio//kamailio.cfg, line 432, column 41: '('')' expected (function call) Jul 8 10:00:25 registrar kamailio: : <core> [cfg.y:3412]: parse error in config file /usr/local/kamailio-3.1//etc/kamailio//kamailio.cfg, line 432, column 41: function t_on_reply: parameter 1 is not constant Jul 8 10:00:25 registrar kamailio: : <core> [cfg.y:3412]: parse error in config file /usr/local/kamailio-3.1//etc/kamailio//kamailio.cfg, line 432, column 41: bad command thank you fborot

3 5

following the path a request take in the config file
by Fabian Borot 08 Jul '11

08 Jul '11

Hi Is there a way of tracing the path a request takes as it is being processed by the config file? Aside from logging using xlog(). I am not sure whether I read about this some where on the web , but I am under the impression that starting on version 3.x there was going to be a similar feature. please advise fborot

2 2

Loading multiple htables from db on startup? (Kamailio 1.5)
by Robin Rodriguez 08 Jul '11

08 Jul '11

Hi all, Currently running on 1.5.4-notls and using a single htable that gets loaded at startup ala modparam("htable", "htable", "local_routes=>size=4;autoexpire=0;dbtable=TABLE_A;") modparam("htable", "key_name_column", "NAME_1") modparam("htable", "key_value_column", "VALUE_1") modparam("htable", "fetch_rows", 512) This works great, the problem I've run into is attempting to load a second htable on startup whose data comes from a different table/scheme ala modparam("htable", "htable", "inbound_acl=>size=4;autoexpire=0;dbtable=TABLE_B;") modparam("htable", "key_name_column", "NAME_2") modparam("htable", "key_value_column", "VALUE_2") modparam("htable", "fetch_rows", 512) With both of them in the config file Kamailio fails to start with the error INFO:core:init_mod: initializing module htable ERROR:db_mysql:db_mysql_submit_query: driver error on query: Unknown column 'NAME_2' in 'field list' 'NAME_2' does not exist in 'TABLE_1' so that seems to suggest that only the first dbtable value is used, the SQL side supports this: select NAME_2,key_type,value_type,VALUE_2 from TABLE_A order by key_type So it uses the first defined database table, but then the last defined key_name and key_value. Is there any way that I've overlooked in the docs or elsewhere about about loading multiple htables from different database tables? I'd like to use this htable in an ACL and would like to avoid database hits though can make it work with avpops if I have to. Any suggestions are greatly appreciated. Thanks, Robin

3 3

Dialplan question
by alex pappas 08 Jul '11

08 Jul '11

Dear all, I have these 3 entries in the dialplan: dpid pr match_exp attr 2 50 ^38.+ porvider_A 2 50 ^3802.+ porvider_B 2 50 ^3806.+ porvider_C with the fifo command : *kamctl fifo dp_translate 2 3806328881* Output:: ATTRIBUTES::* porvider_A* I would expect that the dialplan module would choose the third choice which is the correct one for this call. >From the dialplan documentation i understand that the priority is needed when the prefixes are overlaping in terms of they are the same but with different attrs like : 2 50 ^38.+ porvider_A 2 50 ^3802.+ porvider_B *2 50 ^3806.+ porvider_C* *2 50 ^3806.+ porvider_D* * * here i understand that I should use priorities. Does anyone knnows how this works ?? Thank you in advance Alex * *

2 1

acc table versions
by Klaus Darilion 08 Jul '11

08 Jul '11

Hi! Just as a note: I find it a bit strange that acc (4) and missed_calls (3) have different version numbers although they are identical. Anyway, it seems that acc module does not make version checks at all. regards Klaus

2 1

Celebrating 10 years of SER
by Daniel-Constantin Mierla 07 Jul '11

07 Jul '11

Hello, the 3rd of September will mark the 10 years since the start of SIP Express Router (aka SER - also the origin for kamailio (openser) and now the two are back together). From last year: * http://sip-router.org/wiki/history/ser-9-years This is a good time to celebrate and I am thinking of at least a social event (e.g., dinner) in Berlin, Germany. Based of private discussion so far, other cities are interested in organizing something similar locally, e.g., Barcelona. If you are a group of people that can gather together for a drink in same place to celebrate the event, let me know and I will add in the list to coordinate - hopefully we will be even able connect between sites with some video/audio if the timezones are convenient. I'm looking as well to do a more formal event, something like a day long conference style, on Friday, 2nd of September. Berlin is the natural choice for location, being the place where the project started. I can take care of location when it is clear that there are people willing to travel. Berlin area alone should be able to get about 20 people, among them several long term developers of the project and companies offering SER/Kamailio-based services. Coming over gives you the opportunity to connect directly with them. At this event, along with presenting shortly the history and future plans, the event should be open to everyone that want to present their usage of SER/Kamailio. Attendance is going to be free, in the limit of available seats. If you think it would be interesting and want to attend, let me know. Also, if you have other ideas of how to celebrate, write us back. Best regards, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

3 2

Re: [SR-Users] Playing with dlg_bridge()
by Stefan Broddesson 07 Jul '11

07 Jul '11

Brodde(a)yahoo.com Sent from my HTC ----- Reply message ----- From: "Roberto Fichera" <kernel(a)tekno-soft.it> To: "Henning Westerholt" <henning.westerholt(a)1und1.de> Cc: "sr-users(a)lists.sip-router.org" <sr-users(a)lists.sip-router.org> Subject: [SR-Users] Playing with dlg_bridge() Date: Thu, Jul 7, 2011 16:01 On 07/07/2011 02:51 PM, Henning Westerholt wrote: > On Thursday 07 July 2011, Roberto Fichera wrote: >>>> Do you know what does means the warning below? >>>> [..] >>>> 68.2.50 Jul 6 11:54:17 roberto /usr/sbin/kamailio[20457]: WARNING: >>>> dialog [dlg_handlers.c:884]: unable to find dialog for NOTIFY with >>>> route param '7d2.14447fc4' [727:1291273281] >>> This means that the module can't find an existing (stored) dialog for the >>> routed message in question, in this case a NOTIFY. >> Does this depends by the fact of own the dlg_bridge() works? And in case, >> how can I solve that problem? > Hi Roberto, Hi Henning, > I'm not 100% sure, I'm not the expert for dlg_bridge.. To be more specific > (after discussion with Timo..) - the warning is emmited if the module can't > find the dialog with the ID added as a route parameter to the respective > message. One particiular reason is (if you don't store the dialogs in a > database) a simple kamailio restart, then all existing dialog information is > lost. actually I've the following setup: #!define DBURL "postgres://openser:openserrw@localhost/openser" modparam("dialog", "dlg_flag", 2) modparam("dialog", "bridge_controller", "sip:controller@192.168.2.92") modparam("dialog", "db_url", DBURL ) Actually, my thought is that problem is more strictly related to the dlg_bridge() which might not track the bridged dialog. Anyway, I'm playing with dlg_bridge() because I want to realize the logic so that two peers, and if possible more, can talk together only if authorized. The authorization might be a unique tag or transaction id, stored in a database, and valid only for the give conversation {tag, caller, callee} once started, than the tag become invalid and can't be used again. I really like to have some suggestions about how to realize such logic, so which might be the best way to approach such problem since I'm actually a novice to Kamailio or in general with OpenSer logic. I'm actually more confident with Asterisk logic. > Best regards, > > Henning > _______________________________________________ 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

1 0

force_rtp_proxy/rtpproxy_offer/answer not replacing ip addr.
by MingHon 07 Jul '11

07 Jul '11

Hi List, im facing an issue that my kamailio proxy did not replace the ip address in the invite and 200OK sdp body. my rtpproxy is running: rtpproxy -l 192.168.1.3 -u:*:7722 -u user my kamailio is listening on 192.168.1.3, also define: advertised_address="175.136.223.112"; & advertised_port=5060; and my asterisk is on 192.168.1.23. sip signalling and rtp port forwarded to kamailio. uacs from another nat register successfully. if i put 2 lines of force_rtp_proxy("fcow","175.136.223.112"); i will get double ip addr in c and o but kamailio ignore my ip addr. example i will get c=IN IP4 192.168.1.3192.168.1.3 here is part of my simple script. hope you can help. thank you very much. ---------------cfg------------------- route[RTPPROXY] { #!ifdef WITH_NAT if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy("fcow","175.136.223.112"); #force_rtp_proxy("fcow","175.136.223.112"); xlog("L_INFO","offer"); } if (!has_totag()) add_rr_param(";nat=yes"); #!endif return; } -------------------------------------- and here is the wireshark for uac INVITE and OK. -----------INVITE----------------- ve0 EE;p9INVITE sip:102@192.168.2.132:5062 SIP/2.0 Record-Route: <sip:192.168.1.3;lr=on;ftag=as032358a3;nat=yes> Via: SIP/2.0/UDP 192.168.1.3;branch=z9hG4bK09d5.c5e9e8d2.0 Via: SIP/2.0/UDP 192.168.1.23:5080;branch=z9hG4bK71c27189;rport=5080 Max-Forwards: 69 From: "101" <sip:102@aextddns.dyndns.info>;tag=as032358a3 To: <sip:102@192.168.1.3:5060> Contact: <sip:102@192.168.1.23:5080> Call-ID: 416f6e09674ae9671bb7144a1cb11137(a)aextddns.dyndns.info CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.18 Date: Tue, 05 Jul 2011 07:20:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 327 v=0 o=root 1639709788 1639709788 IN IP4 192.168.1.3 s=Asterisk PBX 1.6.2.18 c=IN IP4 192.168.1.3 t=0 0 m=audio 10072 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=nortpproxy:yes -----------200OK--------------- e90 ElE;pX4tSIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.200:5062 ;rport=2788;received=175.138.21.31;branch=z9hG4bK2086380416 Record-Route: <sip:192.168.1.3;lr=on;ftag=1796959074;nat=yes> From: "101" <sip:101@aextddns.dyndns.info>;tag=1796959074 To: <sip:102@aextddns.dyndns.info>;tag=as2e4c0125 Call-ID: 1985782590(a)192.168.2.200 CSeq: 21 INVITE Server: Asterisk PBX 1.6.2.18 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: <sip:102@192.168.1.23:5080> Content-Type: application/sdp Content-Length: 286 v=0 o=root 403900934 403900934 IN IP4 192.168.1.23 s=Asterisk PBX 1.6.2.18 c=IN IP4 192.168.1.23 t=0 0 m=audio 14420 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv ------------------------------------ My kamailio log. -----------LOG------------------ DEBUG: rtpproxy [rtpproxy_funcs.c:148]: type <application/sdp> found valid DEBUG: rtpproxy [rtpproxy.c:2188]: proxy reply: 10070 192.168.1.3 INFO: <script>: offer ------------------------------------- double force_rtp_proxy --------kamailio -> asterisk [INVITE]--------- Pyi-}E7V@:#pINVITE sip:102@aextddns.dyndns.info SIP/2.0 Record-Route: <sip:192.168.1.3;lr=on;ftag=640933430;nat=yes> Via: SIP/2.0/UDP 192.168.1.3;branch=z9hG4bK89a5.53e9f766.0 Via: SIP/2.0/UDP 192.168.2.200:5062 ;rport=2788;received=175.138.21.31;branch=z9hG4bK1673765648 From: "101" <sip:101@aextddns.dyndns.info>;tag=640933430 To: <sip:102@aextddns.dyndns.info> Call-ID: 1909950509(a)192.168.2.200 CSeq: 21 INVITE Contact: <sip:101@175.138.21.31:2788> Content-Type: application/sdp Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE Max-Forwards: 69 User-Agent: T20 9.41.0.80 Allow-Events: talk,hold,conference,refer,check-sync Content-Length: 334 v=0 o=20073 20073 IN IP4 192.168.1.3192.168.1.3 s=SDP data c=IN IP4 192.168.1.3192.168.1.3 t=0 0 m=audio 1006410064 RTP/AVP 0 8 18 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:9 G722/8000 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv a=nortpproxy:yes a=nortpproxy:yes -----------LOG------------------ DEBUG: rtpproxy [rtpproxy_funcs.c:148]: type <application/sdp> found valid DEBUG: rtpproxy [rtpproxy.c:2188]: proxy reply: 10068 192.168.1.3 DEBUG: rtpproxy [rtpproxy_funcs.c:148]: type <application/sdp> found valid DEBUG: rtpproxy [rtpproxy.c:2188]: proxy reply: 10068 192.168.1.3 INFO: <script>: offer -----------LOG------------------ -- Regards, MingHon

3 19

Playing with dlg_bridge()
by Roberto Fichera 07 Jul '11

07 Jul '11

Hi All in the list, I want to implement a click and dial feature using the xhttp interface but also keeping working the XCAP functionality. Than I added some code about the dialog module (it's my first time I try it) so that the URL look like http://localhost:5060/sip-bridge/<from>/<to> #!ifdef WITH_XCAPSRV #!define WITH_XHTTPAUTH event_route[xhttp:request] { xdbg("===== xhttp: request [$rv] $rm => $hu\n"); if($hu=~"^/xcap-root/") { #!ifdef WITH_XHTTPAUTH if (!www_authorize("xcap", "subscriber")) { www_challenge("xcap", "0"); exit; } #!endif set_reply_close(); set_reply_no_connect(); # 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; xlog("===== xhttp: $xcapuri(u=>auid) : $xcapuri(u=>xuid)\n"); if($xcapuri(u=>auid)=="xcap-caps") { $var(xbody) = "<?xml version='1.0' encoding='UTF-8'?> <xcap-caps xmlns='urn:ietf:params:xml:ns:xcap-caps'> <auids> <auid>rls-services</auid> <auid>pidf-manipulation</auid> <auid>xcap-caps</auid> <auid>resource-lists</auid> <auid>pres-rules</auid> <auid>org.openmobilealliance.pres-rules</auid> </auids> <extensions> </extensions> <namespaces> <namespace>urn:ietf:params:xml:ns:rls-services</namespace> <namespace>urn:ietf:params:xml:ns:pidf</namespace> <namespace>urn:ietf:params:xml:ns:xcap-caps</namespace> <namespace>urn:ietf:params:xml:ns:resource-lists</namespace> <namespace>urn:ietf:params:xml:ns:pres-rules</namespace> </namespaces> </xcap-caps>"; xhttp_reply("200", "ok", "application/xcap-caps+xml", "$var(xbody)"); exit; } #!ifdef WITH_XHTTPAUTH # be sure auth user access only its documents if ($au!=$(var(uri){uri.user})) { xhttp_reply("403", "Forbidden", "text/html", "<html><body>$si:$sp</body></html>"); exit; } #!endif switch($rm) { case "PUT": xcaps_put("$var(uri)", "$hu", "$rb"); if($xcapuri(u=>auid)=~"pres-rules") { xlog("===== xhttp put: refreshing watchers for $var(uri)\n"); pres_update_watchers("$var(uri)", "presence"); pres_refresh_watchers("$var(uri)", "presence", 1); } exit; break; case "GET": xlog("===== xhttp: get $var(uri) => $hu\n"); xcaps_get("$var(uri)", "$hu"); exit; break; case "DELETE": xcaps_del("$var(uri)", "$hu"); if($xcapuri(u=>auid)=~"pres-rules") { xlog("===== xhttp del: refreshing watchers for $var(uri)\n"); pres_update_watchers("$var(uri)", "presence"); pres_refresh_watchers("$var(uri)", "presence", 1); } exit; break; } } else if($hu=~"^/sip-bridge/") { if ( ! ( src_ip == 127.0.0.1 ) ) { xhttp_reply( "403", "Forbidden", "text/html", "<html><body>Invalid Access</body></html>" ); exit; } $var(uri) = $hu; $var(from) = $(var(uri){s.select,2,/}); $var(to) = $(var(uri){s.select,3,/}); xdbg( "=== GOT sip-bridge between $var(caller) => $var(callee) ===" ); *dlg_bridge( $var(from), $var(to), "sip:192.168.2.92:5060" );* } # http ops xhttp_reply("200", "ok", "text/html", "<html><body>OK: $si:$sp</body></html>"); exit; } #!endif but I got the error below: Not starting : invalid configuration file! 0(22256) : <core> [cfg.y:3409]: parse error in config file //etc/kamailio/kamailio.cfg, line 986, column 17-26: function dlg_bridge: parameter 1 is not constant 0(22256) : <core> [cfg.y:3412]: parse error in config file //etc/kamailio/kamailio.cfg, line 986, column 63: bad command ERROR: bad config file (2 errors) so my question is how can I call the dlg_bridge() passing some parameters coming from the http request? Thanks in advance. Roberto Fichera.

2 6

Problem with worst case scenario of accessing 15 gateways over the DISPATCHER module - MAX_BRANCHES is reached
by Klaus Feichtinger 07 Jul '11

07 Jul '11

Hi, I have a problem in my scenario where I use the dispatcher module for (round-robin based) distribution of incoming calls to 15 (!) different gateways. In idle case (= all gateways are up and running or just a couple of them does not accept calls) my solution is working fine. The solution is very primitive and based on the example of the dispatcher module's ReadMe. However, when e.g. 12 of 15 gateways are out of service (in detail: rejecting the call with "480 - Temporarily Unavailable") and only the 13th would be okay, I can not reach that gateway. This is because the TM module is interrupting the distribution with an error message indicating that the maximum number of branches (per transaction) is reached (ERROR:/tm/:/add_uac/: /maximum number of branches/ exceeded).... My first idea for solving this problem was/is in manipulating the definition of the "max_branches" constant in the config.h file. However, I prefer using the precompiled Debian packages. Therefore that solution is not optimal. I have not found a method for decreasing the destination_set on an easy way... Does anybody know an alternative solution / workaround? Thanks in advance, Klaus

3 6

new to ser
by Pradeep Sharma 07 Jul '11

07 Jul '11

Hi, I am installed ser 0.9.6 onto my mac machine and with following command i get it to listen to 5060 port. but when i use a sip client and send register message i get a 501 not implemented as a response from the server. i am attaching ser.cfg. your help is appreciated. -- Thanks Paul

2 1

updates to config preprocessor
by Daniel-Constantin Mierla 07 Jul '11

07 Jul '11

Hello, just to let you know about some recent additions regarding the config file preprocessor, hopefully they will help you keep the config cleaner and easy to maintain. - you can use either #! or !! as the start marker of preprocessor directive (to make happy those complaining about not able to distinguish quickly from comments) - subsdef was added some time ago to allow defining a token that will replace ID tokens as well as update inside string values. It has the same format as #!subst. For example:| #!substdef "!ABC!XYZ!g"| is pretty much same as: #!define ABC XYZ #!subs "/ABC/XYZ/" - substdefs, like substdef byt the defined value will be added enclosed in double quotes: |#!substdef "!ABC!XYZ!g"| is pretty much same as: #!define ABC "XYZ" #!subs "/ABC/XYZ/" - trydef - similar to define, but if the ID is defined already, ignore this time - redef - similar to define, but if the ID is defined, redefine its value (note that define itself throws error if the ID is already defined) Regarding the inclusion of files, a new option is available, 'import_file'. It is similar to 'include_file', but if the file is not found, it does not throw error. Bot include_file and import_file can be started as well with #! or !!. You can still use them without #! or !!, whatever you like more. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

RFC3311 troublefix
by Andrew O. Zhukov 06 Jul '11

06 Jul '11

Please add in all samples, default installed /etc/*.cfg files etc, etc, etc... RFC3311 processinging in a branch route: } else if (is_method("INVITE|UPDATE")){ force_rtp_proxy(); instead of } else if (is_method("INVITE")){ force_rtp_proxy(); Or in last cvs version sample if (nat_uac_test("3")) { .... if ( is_method("INVITE") ) { fix_nated_sdp("1"); # Add direction=active to SDP }; is_method("INVITE|UPDATE"); Thanks in advance !!!

4 6

strange behavior with pseuvars not being found
by Bret McDanel 06 Jul '11

06 Jul '11

I am in the process of migrating from opensips 1.4 to kamailio 3.2 (GIT current as of this morning). I think I have the config fully ported but I appear to be missing something. I get the following error unknown script pseudo variable $cl I also get the same error for $rU and some others. I wanted to see if this is a common problem that generates this error before the config has to be looked at, ie made public. I am making heavy use of #!define and #!ifdef statements, and from what I can tell I am not missing any curly braces or #!endifs. Thanks for any pointers, -- Trixter aka Bret McDanel website: http://www.0xdecafbad.com pgp key: http://bit.ly/9XYK4b

2 1

auth_db question
by Fabian Borot 06 Jul '11

06 Jul '11

Hello I am trying to store the password on the subscriber db as encrypted instead of plain text and when I do it then the registration fails, modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "ha1") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) If I change the 3rd line to this it works but it shows the password in clear text: modparam("auth_db", "password_column", "password") This is how my subscriber table is defined mysql> describe subscriber; +---------------+------------------+------+-----+---------+----------------+ | Field | Type | Null | Key | Default | Extra | +---------------+------------------+------+-----+---------+----------------+ | id | int(10) unsigned | NO | PRI | NULL | auto_increment | | username | varchar(64) | NO | MUL | | | | domain | varchar(64) | NO | | | | | password | varchar(25) | NO | | | | | email_address | varchar(64) | NO | | | | | ha1 | varchar(64) | NO | | | | | ha1b | varchar(64) | NO | | | | | rpid | varchar(64) | YES | | NULL | | +---------------+------------------+------+-----+---------+----------------+ and this is how it looks like in the db: [root@registrar ~]# /usr/local/kamailio-3.1/sbin/kamctl db show subscriber +----+----------+---------------+----------+---------------+----------------------------------+----------------------------------+------+ | id | username | domain | password | email_address | ha1 | ha1b | rpid | +----+----------+---------------+----------+---------------+----------------------------------+----------------------------------+------+ | 1 | 100 | 192.168.169.1 | abc | | 4a169916d58247f9e7eac55251fa7467 | 28787e5b5d3a2c6e2e0a80d3e9e3c6de | NULL | | 2 | 200 | 192.168.169.1 | abc | | 0b15f8f9e348ec64a2e7d5fca6b629c3 | 99eea8b456956d67e7915ed9c96e420b | NULL | +----+----------+---------------+----------+---------------+----------------------------------+----------------------------------+------+ please give me hand figuring this one out, I can't see what I am doing wrong fborot

2 4

kamailio fails to start with kamctl start
by Fabian Borot 06 Jul '11

06 Jul '11

Hello I just installed kamailio-3.1 and using the default config file "kamctl start" fails [root@registrar sbin]# ./kamctl start INFO: Starting Kamailio : ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed [root@registrar sbin]# but when I use this command it starts: <kamailio -f path_to_config_file [root@registrar sbin]# ./kamailio -f /usr/local/kamailio-3.1/etc/kamailio/kamailio.cfg loading modules under /usr/local/kamailio-3.1//lib64/kamailio/modules_k/:/usr/local/kamailio-3.1//lib64/kamailio/modules/ Listening on udp: 127.0.0.1:5060 udp: 192.168.168.10:5060 tcp: 127.0.0.1:5060 tcp: 192.168.168.10:5060 Aliases: tcp: localhost:5060 tcp: localhost.localdomain:5060 tcp: registrar:5060 tcp: registrar.mytestdomain.com:5060 udp: localhost:5060 udp: localhost.localdomain:5060 udp: registrar:5060 udp: registrar.mytestdomain.com:5060 [root@registrar sbin]# Also, when I use "./kamctl start" the syslog file does not show anything at all that would help me why it is failing. But when I use "/usr/local/kamailio-3.1/sbin/kamailio -f /usr/local/kamailio-3.1/etc/kamailio/kamailio.cfg" it does update the syslog file. Also, after kamailio start with "./kamailio -f .." I can use ./kamctl fifo <any option>. Example: [root@registrar sbin]# ./kamctl fifo version Server:: kamailio (3.1.4 (x86_64/linux)) Build:: mi_core.c compiled on 15:09:40 Jun 24 2011 with gcc 4.1.2 Flags:: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES GIT:: e2d6f0 [root@registrar sbin]# [root@registrar sbin]# ./kamctl fifo ps Process:: ID=0 PID=27078 Type=attendant Process:: ID=1 PID=27079 Type=udp receiver child=0 sock=127.0.0.1:5060 Process:: ID=2 PID=27080 Type=udp receiver child=1 sock=127.0.0.1:5060 Process:: ID=3 PID=27081 Type=udp receiver child=2 sock=127.0.0.1:5060 Process:: ID=4 PID=27082 Type=udp receiver child=3 sock=127.0.0.1:5060 Process:: ID=5 PID=27083 Type=udp receiver child=0 sock=192.168.168.10:5060 Process:: ID=6 PID=27084 Type=udp receiver child=1 sock=192.168.168.10:5060 Process:: ID=7 PID=27085 Type=udp receiver child=2 sock=192.168.168.10:5060 Process:: ID=8 PID=27086 Type=udp receiver child=3 sock=192.168.168.10:5060 Process:: ID=9 PID=27087 Type=slow timer Process:: ID=10 PID=27088 Type=timer Process:: ID=11 PID=27089 Type=MI FIFO Process:: ID=12 PID=27090 Type=ctl handler Process:: ID=13 PID=27091 Type=tcp receiver child=0 Process:: ID=14 PID=27092 Type=tcp receiver child=1 Process:: ID=15 PID=27093 Type=tcp receiver child=2 Process:: ID=16 PID=27094 Type=tcp receiver child=3 Process:: ID=17 PID=27095 Type=tcp main process [root@registrar sbin]#

2 6

Muenster germany next week
by Olle E. Johansson 06 Jul '11

06 Jul '11

Friends, Going to be in Muenster next week. If anyone is nearby, I'll be happy to take part in a VoIP dinner :-) /O

1 0

Kamailio 3.1.4 Compilation Error ('TMCB_DONT_ACK' undeclared)
by Bruce McAlister 05 Jul '11

05 Jul '11

Hi All, Would someone be able to point me in the right direction here, I am trying to compile Kamailio 3.1.4 from git checkout and I am ending up with the following error: gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -m64 -minline-all-stringops -falign-loops -ftree-vectorize -mtune=opteron -Wall -DNAME='"ser"' -DVERSION='"3.1.4"' -DARCH='"x86_64"' -DOS='linux_' -DOS_QUOTED='"linux"' -DCOMPILER='"gcc 4.1.2"' -D__CPU_x86_64 -D__OS_linux -DSER_VER=3001004 -DCFG_DIR='"/usr/local/loadbalancer/lb1/etc/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST -DUSE_NAPTR -DDBG_QM_MALLOC -DUSE_TLS -DTLS_HOOKS -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_SCHED_SETSCHEDULER -DUSE_RAW_SOCKS -DHAVE_EPOLL -DHAVE_SIGIO_RT -DSIGINFO64_WORKARROUND -DUSE_FUTEX -DHAVE_SELECT -DOPENSER_MOD_INTERFACE -DMOD_NAME='"seas"' -c seas_action.c -o seas_action.o -MMD -MP seas_action.c: In function 'ac_uac_req': seas_action.c:1097: error: 'TMCB_DONT_ACK' undeclared (first use in this function) seas_action.c:1097: error: (Each undeclared identifier is reported only once seas_action.c:1097: error: for each function it appears in.) make[1]: *** [seas_action.o] Error 1 make: *** [modules_k] Error 1 I am fairly certain I have all required -devel packages installed, but I just cannot see where that identifier is meant to be defined. Any tips/suggestions would be greatly appreciated. If you require more info, please let me know what you need. thanks Bruce

4 5

job offer: System administrator VoIP
by Henning Westerholt 05 Jul '11

05 Jul '11

Job offer: Interested in working with one of the biggest Voice over IP networks in europe? We're hiring and have one additional IT operation position open in our team in Karlsruhe, Germany at the moment. You would be responsible in a team of motivated system administrators for the operation and maintenance of the VoIP kamailio backend at 1&1, providing services for more than three million customers. You should have some experience in operating or developing Voice over IP services with kamailio, asterisk or others similar servers. Additionally you should be able to communicate professionally in the german language. For more informations please contact me directly by email. Best regards, Henning Westerholt -- Henning Westerholt - Head of IT Operations Internet Access & Communications 1&1 Internet AG, Brauerstraße 48, 76135 Karlsruhe, Germany

1 0

more on asynchronous processing in config
by Daniel-Constantin Mierla 05 Jul '11

05 Jul '11

Hello, over the weekend I exported to config file the two main functions from tm that allow to suspend the execution of a SIP request and resume it in another place. They are exported through tmx module: - t_suspend() - will suspend the execution of current sip request and then you can do something else (like processing the next request) - t_continue(tindex, tlabel, rtname) - continue the processing of SIP request suspended in the transaction identified by tupple (tindex,tlable) with the actions in route[rtname] After t_suspend() you can get the transaction identifier (tindex, tlable) via pseudo-variables $T(id_index) and $T(id_label), e.g.: if(t_suspend()) { xlog("sip request suspended in transaction [$T(id_index)/$T(id_label)]\n"); exit; } One example of usage can be passing the $T(id_index) and $T(id_label) via mqueue to another process for further handling. So, besides the new additions, now following cfg tools can be used to do asynchronous operations directly via config: - the mqueue + rtimer module - pass a task via mqueue and get it in rtimer specific processes to run it further - async module - intended to collect asynchronous specific implementation, for now able to sleep and execute a specific route after a time interval Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

2 2

dbtext module
by Bruno Bresciani 05 Jul '11

05 Jul '11

Hi All, I'm using kamailio with database in text mode and i'd like to know if dbtext module has support the following parameters to search: ilike; limit; offset; Regards

3 3

"select count(*)" with avp_db_query always returns null
by Andrade Ricardo (CI/AFU1) 05 Jul '11

05 Jul '11

Hi list, Perhaps this fits better into a bug report, but I'd like to know if somebody out there experienced a similar issue. I am executing a "select count" query using the avp_db_query function, but it is not storing the results in any avp. Other queries are working fine. I use db_mysql connected with a mysql 5.1 server. I have tested this with kamailio-3.1.0 and kamailio-3.1.3, both didn't work. In an old box (version 1.3.x), the same query was returning the correct value. --------------------------------------------------- Here is the case which is not working: Code: avp_delete("$avp(s:count)"); $var(ret) = avp_db_query("SELECT count(*) FROM subscriber where username='foo'", "$avp(s:count)"); xlog("L_INFO", "var(ret)=$var(ret) avp(s:count)=$avp(s:count)"); Output: INFO: <script>: var(ret)=1 avp(s:count)=<null> (notice the return code 1, which means that the query was successfull and there should be some output value stored in the avp, but it is <null>) --------------------------------------------------- And here is one case which works as expected: $var(ret) = avp_db_query("SELECT id FROM subscriber where username='foo'", "$avp(s:count)"); Output: INFO: <script>: var(ret)=1 avp(s:count)=11 --------------------------------------------------- Thanks in advance for any feedback. Best regards, Ricardo Andrade.

4 8

two uac behind same nat one uac remain Cflag:: 0
by MingHon 05 Jul '11

05 Jul '11

Hi List, both uac register successfully but the issue is the first uac who register the Cflag remain 0. im using nat_uac_test("19"). thank you in advance. here is the ul show: AOR:: 102 Contact:: sip:102@175.136.221.60:5062 Q= Expires:: 103 Callid:: 678936531(a)175.136.221.60 Cseq:: 2 User-agent:: T22 7.3.0.50 Received:: sip:175.136.221.60:1056 State:: CS_SYNC Flags:: 0 Cflag:: 192 Socket:: udp:60.53.82.184:5060 Methods:: 16383 AOR:: 103 Contact:: sip:103@175.136.221.60:5062 Q= Expires:: 62 Callid:: 1621958925(a)175.136.221.60 Cseq:: 3 User-agent:: Yealink SIP-T18 18.0.0.70 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: udp:60.53.82.184:5060 Methods:: 16383 and here is the wireshark on kamailio for ua 103 and 102. E@< \<<5Rw REGISTER sip:aextddns.dyndns.info SIP/2.0 Via: SIP/2.0/UDP 175.136.221.60:5062;branch=z9hG4bK2080454766 From: "103" <sip:103@175.136.221.60>;tag=681558690 To: "103" <sip:103@aextddns.dyndns.info> Call-ID: 1621958925(a)175.136.221.60 CSeq: 2 REGISTER Contact: <sip:103@175.136.221.60:5062> Authorization: Digest username="103", realm="aextddns.dyndns.info", nonce="Tgp7mU4Kem3Qqyd66hwhWgmU8bNpL463", uri="si$ Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE Max-Forwards: 70 User-Agent: Yealink SIP-T18 18.0.0.70 Expires: 3600 Content-Length: 0 E@<5R<PSIP/2.0 200 OK Via: SIP/2.0/UDP 175.136.221.60:5062;branch=z9hG4bK2080454766;rport=5062 From: "103" <sip:103@175.136.221.60>;tag=681558690 To: "103" <sip:103@aextddns.dyndns.info >;tag=cbf8071f5b02e3e2fd329ee5d72c3f65.d24e Call-ID: 1621958925(a)175.136.221.60 CSeq: 2 REGISTER Contact: <sip:103@175.136.221.60:5062>;expires=120 Server: kamailio (3.1.4 (i386/linux)) Content-Length: 0 -------------------------------------------------- E@< j<<5R 7REGISTER sip:aextddns.dyndns.info SIP/2.0 Via: SIP/2.0/UDP 175.136.221.60:5062;branch=z9hG4bK1080036976 From: "102" <sip:102@175.136.221.60>;tag=544575245 To: "102" <sip:102@aextddns.dyndns.info> Call-ID: 678936531(a)175.136.221.60 CSeq: 2 REGISTER Contact: <sip:102@175.136.221.60:5062> Authorization: Digest username="102", realm="aextddns.dyndns.info", nonce="Tgp8AU4KetWGb79MEU5MUHq+xA4xy+OW", uri="si$ Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE Max-Forwards: 70 User-Agent: T22 7.3.0.50 Expires: 3600 Content-Length: 0 E@<5R< {SIP/2.0 200 OK Via: SIP/2.0/UDP 175.136.221.60:5062;branch=z9hG4bK1080036976;rport=1056 From: "102" <sip:102@175.136.221.60>;tag=544575245 To: "102" <sip:102@aextddns.dyndns.info >;tag=cbf8071f5b02e3e2fd329ee5d72c3f65.5cb8 Call-ID: 678936531(a)175.136.221.60 CSeq: 2 REGISTER Contact: <sip:102@175.136.221.60:5062>;expires=120;received="sip: 175.136.221.60:1056" Server: kamailio (3.1.4 (i386/linux)) Content-Length: 0 -- Regards, MingHon

2 2

Hello,can you send me the useage of siremis?Thank you
by 吴明 05 Jul '11

05 Jul '11

HI: I'm a user of siremis from China.Now I install the siremis successfully.But I would not use it. So I want the useage of it. I'll wait for your letter.Thank you! Best regards. Calvin wu 2011-6-24

2 1

Kamailio 64bit Compile Errors
by Nathaniel L Keeling 05 Jul '11

05 Jul '11

I am trying to install Kamailio 3.1.0 on Solaris 10 that is a sparc machine (v120) and is running a 64bit OS. When I originally performed the compile utilizing the instructions from the website, I was getting warnings about using 32 bit headers. I was able to compile and install successfully, but when I started Kamailio, loading of modules were failing due to "not found" error. The warning messages: "atomic/atomic_sparc.h:41:2: warning: #warning "sparc32 atomic operations support not tested"" I thought was because the target architecture specified sparc and not sparc64, so I tried specified the target architecture as listed below which got rid of the warning messages but then I got ELFCLASS32 error messages. What is the proper way to create a 64bit version of Kamailio on a sparc Solaris 10. It seems like the target architecure is not being configured properly and that may be causing other problems. I have included output from different commands about the environment and the commands that I entered to compile Kamailio. # make FLAVOUR=kamailio cfg target architecture <sparc>, host architecture <sparc64> # isainfo -n sparcv9 # uname -p sparc # uname -m sun4u # isainfo -v 64-bit sparcv9 applications vis 32-bit sparc applications vis v8plus div32 mul32 # isainfo -b 64 Here is what I entered to try and force the building of a 64 bit version and a sample error that I was getting. make FLAVOUR=kamailio prefix=/usr/local/kamailio-3.1 CPU=ultrasparc ARCH=sparc64 cfg LD (gcc) [M lcr.so] lcr.so ld: fatal: file /usr/local/lib/libpcre.so: wrong ELF class: ELFCLASS32 ld: fatal: File processing errors. No output written to lcr.so collect2: ld returned 1 exit status make[1]: *** [lcr.so] Error 1 make: *** [modules] Error 1 LD_LIBRARY_PATH=/lib/sparcv9:/usr/lib/sparcv9:/usr/sfw/lib/sparcv9:/usr/local/postgres/9.0-pgdg/lib/64 LD_RUN_PATH=/lib/sparcv9:/usr/lib/sparcv9:/usr/sfw/lib/sparcv9:/usr/local/postgres/9.0-pgdg/lib/64 PATH=/opt/csw/bin:/usr/sfw/bin:/usr/local/bin:/usr/local/postgres/9.0-pgdg/bin:/usr/sbin:/usr/bin:/usr/ccs/bin: # find . -name libpcre.so -print ./usr/local/lib/libpcre.so Thanks

2 1

Playing with XCAP_SERVER and Postgresql
by Roberto Fichera 05 Jul '11

05 Jul '11

Hi all in the List, I'm playing with the xcap_server using the jitsi client. Kamailio is v3.1.4 and postgresql is v8.4.8 under Fedora 14 64bit. Basically I'm getting the error below. Jul 5 10:36:36 roberto /usr/sbin/kamailio[8529]: ERROR: <script>: ===== xhttp: xcap-caps : <null> Jul 5 10:36:36 roberto /usr/sbin/kamailio[8530]: ERROR: <script>: ===== xhttp: resource-lists : sip:roberto@192.168.2.92 Jul 5 10:36:36 roberto /usr/sbin/kamailio[8530]: ERROR: <script>: ===== xhttp: get sip:roberto@192.168.2.92 => /xcap-root/resource-lists/users/sip:roberto@192.168.2.92/index Jul 5 10:36:36 roberto /usr/sbin/kamailio[8531]: ERROR: <script>: ===== xhttp: pres-rules : sip:roberto@192.168.2.92 Jul 5 10:36:36 roberto /usr/sbin/kamailio[8531]: ERROR: <script>: ===== xhttp: get sip:roberto@192.168.2.92 => /xcap-root/pres-rules/users/sip:roberto@192.168.2.92/presrules Jul 5 10:36:36 roberto /usr/sbin/kamailio[8521]: INFO: presence [notify.c:1588]: NOTIFY sip:roberto@192.168.2.92 via sip:roberto@192.168.2.92:5090;transport=udp;registering_acc=192_168_2_92 on behalf of sip:mattia@192.168.2.92 for event presence Jul 5 10:36:36 roberto /usr/sbin/kamailio[8520]: ERROR: presence [subscribe.c:711]: Missing or unsupported event header field value Jul 5 10:36:36 roberto /usr/sbin/kamailio[8520]: ERROR: presence [subscribe.c:714]: #011event= message-summary Jul 5 10:36:36 roberto /usr/sbin/kamailio[8522]: INFO: presence [notify.c:1588]: NOTIFY sip:roberto@192.168.2.92 via sip:roberto@192.168.2.92:5090;transport=udp;registering_acc=192_168_2_92 on behalf of sip:roberto@192.168.2.92 for event presence.winfo *Jul 5 10:36:36 roberto /usr/sbin/kamailio[8520]: ERROR: db_postgres [km_dbase.c:427]: invalid query, execution aborted Jul 5 10:36:36 roberto /usr/sbin/kamailio[8520]: ERROR: db_postgres [km_dbase.c:428]: driver error: PGRES_FATAL_ERROR, ERROR: null value in column "sender" violates not-null constraint#012 Jul 5 10:36:36 roberto /usr/sbin/kamailio[8520]: WARNING: db_postgres [km_dbase.c:473]: unexpected result returned* Any idea? Best regards, Roberto Fichera.

2 2

Presence and Presence_mwi
by Marvin59 05 Jul '11

05 Jul '11

Hi all, I am trying to configure kamailio-3.1.4 in order to send me MWI Notify when there is change in messages. I have already configure it, in order to use a FreeRadius to authenticate my users but I have no idea to use Presence and Presence_mwi module. I need this behavior : - The phone send me a subscribe, then if it authorize i send it the Notify with the messages-waiting. - I must write myself the notify request because an external script will tell me if there are messages waiting. I have looked here (http://www.kamailio.org/docs/modules/3.1.x/modules_k/presence.html#id2614273 and http://www.kamailio.org/docs/modules/3.1.x/modules_k/presence_mwi.html). If someone has further informations .... Thanks for your help. Best regards, Marvin59 -- View this message in context: http://old.nabble.com/Presence-and-Presence_mwi-tp31946526p31946526.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

2 1

Session authorization just before initiate it
by Roberto Fichera 05 Jul '11

05 Jul '11

Hi All in the list, I'm new to Kamailio but I'm actually able to quickly setup a server, including also the SIP presence, looking at the example I found at http://kb.asipto.com/kamailio:presence:k31-made-simple which explain very well how to do it. I also made some changes in the final setup and everything seems working well, I can have audio/video/chat session including the presence status of any user. Now I would like to improve my testing setup so that for each call request Kamailio receive, I would like to perform some SQL lookup for deciding if the conversation might be established or not. About the SQL lookup I read about the SQLOPS module, it seems really easy to use, but I don't know where to put such check before the INVITE. Could someone give me some hints? P.S: Indeed, I need to read some good documentation about how to code a Kamailio config file. Any URL to suggest? Thanks in advance, Roberto Fichera.

2 4

route_graph.py
by s.maertens＠telenet.be 04 Jul '11

04 Jul '11

Hi, I have a kamailio 3.1.4 installation installed from the APT repository kamailio.org. I have read about a python script route_graph which I would like to test. The only thing is, i cannot seem to find out what deb package I should install to have this. Could someone point me in the right direction please? :) Best regards Stefaan

3 2

force_rtp_proxy[flag, ipaddress] not rewriting c and o ?
by MingHon 04 Jul '11

04 Jul '11

im trying to replace the ip address in c= and o= with force_rtp_proxy. i tried different flag co,rc,ro,fo,fc,rfco but still doesnt work. hope some one can help me with this.. below is my noob simple script. hope someone can help.. thanks you :) route[RTPPROXY] { #!ifdef WITH_NAT if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ xlog("L_INFO","force_rtp_proxy!!!! $rm from $fu (IP:$si:$sp)\n"); force_rtp_proxy("rfco","publicip"); } if (!has_totag()) add_rr_param(";nat=yes"); #!endif return; } -- Regards, MingHon

2 4

MWI Notify Request
by Marvin59 01 Jul '11

01 Jul '11

Hi all, I would like to know how kamailio sends Notify request by using the handle_subscribe of presence module. My purpose is to modify the request to add information for "message waiting indication" like : Messages-Waiting: yes, Voice-Message: 2/0 (0/0), ... I need your help ! Thanks in advance, Best regards Marvin -- View this message in context: http://old.nabble.com/MWI-Notify-Request-tp31954500p31954500.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

2 2

Internet RTP filters
by Dominguez Jover, Ricardo 30 Jun '11

30 Jun '11

Hi everybody, First to say sorry if this is not the right list to send this post, anyway we hope somebody can help us. We have Kamalio 3.1.3 working since many time ago. We use an RTP Proxy for NAT issues in the server side and STUN for NAT issues in the internet client side. Calls in Spain are working fine. This month one of our users has gone to Rwanda (our University has created a Hospital and a School in a region called Nemba). They are using an internet connection by satellite, with a low bandwith (128kb down, 16 up). In some countries we have trouble with internet calls. SIP signaling works fine, the call rings and is stablished, but there is a problem with RTP streams: - RTP streams sent by RTP proxy to client in internet never arrives to the client in the internet (although they RTP is sent to the right IP and port) - RTP streams sent by client in the internet never arrives to RTP proxy (although RTP is sent to the right IP and port) As I said this happens sometimes, and it's happening in Rwanda. We usually solve this problem making a VPN connection to the University, then the RTP stream works fine. However when our Rwanda's users connect to VPN, the Bandwith goes down (45 kbps down and 8kbps up), so they can hear as perfectly, but we can't hear them. Does anyone know if T companies are filtering RTP traffic? I've tried to change ports in RPT Proxy with no result. How could we solve this? Thanks in advance and best regards, Ricardo Dominguez Universidad Miguel Hernández de Elche

2 2

Re: [SR-Users] MWI handling
by Ovidiu Sas 30 Jun '11

30 Jun '11

If you are using asterisk as a voicemail server, let asterisk to handle the SUBSCRIBE and don't involve the kamailio presence server in this process. If you really want to use kamailio as a mwi presence server, then you will need to build some external scripts for pooling the asterisk mwi db and send PUBLISH out. I would stick with asterisk handling the mwi subscription and configure kamailio as a relay only for those events. Regards, Ovidiu Sas On Thu, Jun 30, 2011 at 9:30 AM, Spinov Evgeniy <spinov_evgeniy(a)intalisan.com> wrote: > This makes the main problem as in documents I've found on asterisk is > that it doesn't support PUBLISH, i.e. there is no way to force sending > them on MWI event. It works only on SUBSCRIBE-NOTIFY scheme. > > Did you succeeded with PUBLISH on Asterisk in order to inform Kamailio > about MWI? If yes, how? Cause right now, for me, it looks like a theory, > cause no mechanics to force asterisk send PUBLISHes. > > Thank you. > > Regards, > Evgeniy Spinov > > >> If you are using asterisk for voicemail, you should let asterisk >> notify the subscribers about mwi. >> If you want to use kamailio for notifications, then you need to >> configure asterisk to send PUBLISH requests to kamailio and kamailio >> will notify the subscribers. > > >> Regards, >> Ovidiu Sas > >

1 0

Re: [SR-Users] MWI handling
by Spinov Evgeniy 30 Jun '11

30 Jun '11

I've done that in this fashion: Kamailio is only proxing SUBSCRIBEs to Asterisk and it's replying with NOTIFYs to Kamailio, Kamailio is just proxying reply to UAC. So Kamailio is transparent for MWI. But it looks like spike. I'd better send PUBLISHes from Asterisk to Kamailio for all the clients and Kamailio reply with NOTIFY to subscribed ones. This is like Kamailio MWI module was designed. Anyway, thanks for the help. Scripts for making PUBLISHES is spike even more. On Thu, 2011-06-30 at 09:53 -0400, Ovidiu Sas wrote: > If you are using asterisk as a voicemail server, let asterisk to > handle the SUBSCRIBE and don't involve the kamailio presence server in > this process. > If you really want to use kamailio as a mwi presence server, then you > will need to build some external scripts for pooling the asterisk mwi > db and send PUBLISH out. > > I would stick with asterisk handling the mwi subscription and > configure kamailio as a relay only for those events. > > > Regards, > Ovidiu Sas > > On Thu, Jun 30, 2011 at 9:30 AM, Spinov Evgeniy > <spinov_evgeniy(a)intalisan.com> wrote: > > This makes the main problem as in documents I've found on asterisk is > > that it doesn't support PUBLISH, i.e. there is no way to force sending > > them on MWI event. It works only on SUBSCRIBE-NOTIFY scheme. > > > > Did you succeeded with PUBLISH on Asterisk in order to inform Kamailio > > about MWI? If yes, how? Cause right now, for me, it looks like a theory, > > cause no mechanics to force asterisk send PUBLISHes. > > > > Thank you. > > > > Regards, > > Evgeniy Spinov > > > > > >> If you are using asterisk for voicemail, you should let asterisk > >> notify the subscribers about mwi. > >> If you want to use kamailio for notifications, then you need to > >> configure asterisk to send PUBLISH requests to kamailio and kamailio > >> will notify the subscribers. > > > > > >> Regards, > >> Ovidiu Sas > > > >

1 0

Re: [SR-Users] MWI handling
by Spinov Evgeniy 30 Jun '11

30 Jun '11

This makes the main problem as in documents I've found on asterisk is that it doesn't support PUBLISH, i.e. there is no way to force sending them on MWI event. It works only on SUBSCRIBE-NOTIFY scheme. Did you succeeded with PUBLISH on Asterisk in order to inform Kamailio about MWI? If yes, how? Cause right now, for me, it looks like a theory, cause no mechanics to force asterisk send PUBLISHes. Thank you. Regards, Evgeniy Spinov > If you are using asterisk for voicemail, you should let asterisk > notify the subscribers about mwi. > If you want to use kamailio for notifications, then you need to > configure asterisk to send PUBLISH requests to kamailio and kamailio > will notify the subscribers. > Regards, > Ovidiu Sas

1 0

mq_fetch and while loops
by Watkins, Bradley 30 Jun '11

30 Jun '11

I'm having an issue trying to use the construct show in the module documentation. The docs show: while(mq_fetch("myq")) { xlog("$mqk(myq) - $mqv(myq)\n"); } And I'm doing essentially the same thing (differently named mqueue, but otherwise identical), but invariably I get this in the logs: ERROR: <core> [action.c:1388]: ERROR: runaway while (559, 28): more then 100 loops Am I doing something wrong, or is mq_fetch not returning appropriate values for successful versus failed fetches? This is on Kamailio 3.1.4, FWIW. Regards, - Brad The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it.

4 5

Table location, methods field
by Andrew O. Zhukov 30 Jun '11

30 Jun '11

Can someone send me bitmasks for the Subj. Or just the reference to include file with comments in a source code. I can't find it in all set of availible document's since the SER router. :( I just have to know which UA except Grandstreams an CSipsample use RFC3311 :) Thanks in advance. Andrew O. Zhukov

2 5

kamailio + rtpproxy behind nat possible?
by MingHon 30 Jun '11

30 Jun '11

will kamailio as proxy behind nat and UACs behind another nat work? port forward sip and rtp done in the router. UACs register successfully but no audio. advertised_address = public_ip advertised_port = sip both define after the line of listen=public_ip please advice. thanks. -- Regards, MingHon

3 6

Proxy setup in a non-recursive mode
by Abdul Hakeem 29 Jun '11

29 Jun '11

Hello, Could anyone help with suggestions on how to setup a Proxy ( R-Proxy) that only does a route lookup and pass the parameter back to origination Proxy/Gateway A to forward the call based on the new parameter. On receipt of an invite from GW/Proxy A, R- Proxy will do a route lookup for the called number 447988870290, If it finds a route then it should append the network prefix to the new INVITE or rewrite the INVITE 23430442088870290 in the 302 Moved Temporarily message sent back to GW/Proxy A in the Contact field and there are two CC-Diversion header fields included; at GW/Proxy (IP address) who originated the call and already has a route or dial plan for the new prefix and also another one for same at GW/Proxy (IP address). Is there any way to do this without creating a loop in the network ? Previously this setup can be achieved with the originating network having 2 GW/Proxies i.e. GW/Proxy A and GW/Proxy B. In the 302 Temporarily moved message the Contact will show GW/Proxy B, while the 2 CC-Diversion header fields will add GW/Proxy B and GW Proxy A. Regards, Abdul Hakeem

2 3

MWI handling
by Spinov Evgeniy 29 Jun '11

29 Jun '11

Hello. Trying to understand how to work with the presence_mwi module. It seems to me, that it's just for my purpose - inform subscribed peers about waiting messages. But it do not export any functions, noting. The scheme is simple: Peer->Kamailio->Asterisk. Asterisk can accept voicemail and when the message is saved I want to receive NOTIFY for subscribed peer. As I understand, peer should register on Kamailio, kamailio should accept this subscription and send subscription to asterisk. After this, need to proxy all NOTIFY to the peer. Anyway, cannot find any way to do that. How it's usually done, please, advice. Thanks in advance.

2 1

Re: [SR-Users] Questions about LCR module
by Fabian Borot 29 Jun '11

29 Jun '11

Thanks Juha I populated the lcr tables by hand. This is how they look now, I have 2 lcr_ids [1 and 2], each instance has its own gw. And each instance has its own prefix to match for [4444 and 5555]. The idea is that a call comes in with R-URI 444413051234567, it should match the lcr_id= 200, then it should go out using gw 192.168.169.200 etc mysql> select * from lcr_gw; +----+--------+------------+-----------------+-------------------+------+--------+------------+-----------+-------+------+-------+---------+ | id | lcr_id | gw_name | ip_addr | hostname | port | params | uri_scheme | transport | strip | tag | flags | defunct | +----+--------+------------+-----------------+-------------------+------+--------+------------+-----------+-------+------+-------+---------+ | 1 | 200 | switchvox | 192.168.169.200 | pbx1.mydomain.com | 5060 | none | 1 | 1 | 0 | | 0 | 0 | | 2 | 201 | switchvox2 | 192.168.169.201 | pbx1.mydomain.net | 5060 | none | 1 | 1 | 0 | | 0 | 0 | +----+--------+------------+-----------------+-------------------+------+--------+------------+-----------+-------+------+-------+---------+ mysql> select * from lcr_rule; +----+--------+--------+----------+---------+---------+ | id | lcr_id | prefix | from_uri | stopper | enabled | +----+--------+--------+----------+---------+---------+ | 1 | 200 | 4444 | | 0 | 1 | | 2 | 201 | 5555 | | 0 | 1 | +----+--------+--------+----------+---------+---------+ mysql> select * from lcr_rule_target; +----+--------+---------+-------+----------+--------+ | id | lcr_id | rule_id | gw_id | priority | weight | +----+--------+---------+-------+----------+--------+ | 1 | 200 | 1 | 1 | 5 | 1 | | 2 | 201 | 2 | 2 | 5 | 1 | +----+--------+---------+-------+----------+--------+ Then I tried the sercmd commands to reload and dump and I do not see the contents o the screen. Is this how is supposed to work now?. with debug level set to 3 I do this "/usr/local/kamailio-3.1/sbin/sercmd lcr.dump_rules" and see these lines on the syslog: Jun 28 11:53:57 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: ctl [../../io_wait.h:390]: DBG: io_watch_add(0x2b772e494660, 12, 3, 0x9679e8), fd_no=1 Jun 28 11:53:57 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: ctl [io_listener.c:452]: handle_stream read: new connection (1) on /tmp/ser_ctl Jun 28 11:53:57 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: ctl [io_listener.c:505]: handle_stream read: eof on /tmp/ser_ctl Jun 28 11:53:57 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: ctl [../../io_wait.h:617]: DBG: io_watch_del (0x2b772e494660, 12, -1, 0x10) fd_no=2 called if I do this: "/usr/local/kamailio-3.1/sbin/sercmd lcr.reload" I see these: Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: ctl [../../io_wait.h:390]: DBG: io_watch_add(0x2b772e494660, 12, 3, 0x9679e8), fd_no=1 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: ctl [io_listener.c:452]: handle_stream read: new connection (1) on /tmp/ser_ctl Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db.c:294]: connection 0x951b78 found in pool Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:118]: allocate 48 bytes for result set at 0x952728 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:68]: 5 columns returned from the query Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:155]: allocate 40 bytes for result names at 0x9527b8 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:165]: allocate 20 bytes for result types at 0x952610 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[0] at 0x9528b8 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x9528b8)[0]=[id] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[1] at 0x952928 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x952928)[1]=[prefix] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:136]: use DB1_STRING result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[2] at 0x951a98 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x951a98)[2]=[from_uri] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:136]: use DB1_STRING result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[3] at 0x951b08 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x951b08)[3]=[stopper] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[4] at 0x952998 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x952998)[4]=[enabled] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_dbase.c:326]: no rows returned from the query Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: lcr [lcr_mod.c:985]: loading, cycle 0 with <0> rows Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:81]: freeing 5 columns Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[0] at 0x9528b8 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[1] at 0x952928 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[2] at 0x951a98 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[3] at 0x951b08 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[4] at 0x952998 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:94]: freeing result names at 0x9527b8 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:99]: freeing result types at 0x952610 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:136]: freeing result set at 0x952728 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:118]: allocate 48 bytes for result set at 0x952728 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:68]: 12 columns returned from the query Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:155]: allocate 96 bytes for result names at 0x951ef8 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:165]: allocate 48 bytes for result types at 0x951c38 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[0] at 0x951b08 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x951b08)[0]=[gw_name] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:136]: use DB1_STRING result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[1] at 0x951a98 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x951a98)[1]=[ip_addr] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:136]: use DB1_STRING result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[2] at 0x952928 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x952928)[2]=[port] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[3] at 0x9528b8 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x9528b8)[3]=[uri_scheme] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[4] at 0x952610 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x952610)[4]=[transport] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[5] at 0x952998 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x952998)[5]=[params] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:136]: use DB1_STRING result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[6] at 0x9527b8 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x9527b8)[6]=[hostname] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:136]: use DB1_STRING result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[7] at 0x951fb8 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x951fb8)[7]=[strip] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[8] at 0x952028 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x952028)[8]=[tag] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:136]: use DB1_STRING result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[9] at 0x952098 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x952098)[9]=[flags] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[10] at 0x952108 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x952108)[10]=[defunct] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[11] at 0x951288 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x951288)[11]=[id] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:169]: no rows returned from the query Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:81]: freeing 12 columns Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[0] at 0x951b08 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[1] at 0x951a98 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[2] at 0x952928 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[3] at 0x9528b8 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[4] at 0x952610 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[5] at 0x952998 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[6] at 0x9527b8 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[7] at 0x951fb8 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[8] at 0x952028 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[9] at 0x952098 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[10] at 0x952108 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[11] at 0x951288 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:94]: freeing result names at 0x951ef8 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:99]: freeing result types at 0x951c38 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:136]: freeing result set at 0x952728 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:118]: allocate 48 bytes for result set at 0x952728 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:68]: 4 columns returned from the query Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:155]: allocate 32 bytes for result names at 0x952998 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:165]: allocate 16 bytes for result types at 0x951288 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[0] at 0x952098 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x952098)[0]=[rule_id] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[1] at 0x952028 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x952028)[1]=[gw_id] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[2] at 0x951fb8 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x951fb8)[2]=[priority] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:85]: allocate 16 bytes for RES_NAMES[3] at 0x9528b8 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x9528b8)[3]=[weight] Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_res.c:100]: use DB1_INT result type Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: db_mysql [km_dbase.c:326]: no rows returned from the query Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: lcr [lcr_mod.c:1344]: loading, cycle 0 with <0> rows Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:81]: freeing 4 columns Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[0] at 0x952098 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[1] at 0x952028 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[2] at 0x951fb8 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[3] at 0x9528b8 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:94]: freeing result names at 0x952998 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:99]: freeing result types at 0x951288 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_res.c:136]: freeing result set at 0x952728 Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: <core> [db_pool.c:97]: connection still kept in the pool Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: ctl [io_listener.c:505]: handle_stream read: eof on /tmp/ser_ctl Jun 28 11:55:04 registrar /usr/local/kamailio-3.1/sbin/kamailio[4309]: DEBUG: ctl [../../io_wait.h:617]: DBG: io_watch_del (0x2b772e494660, 12, -1, 0x10) fd_no=2 called I am trying to port our setup from kamailio 1.4.3 to 3.1.4 and I see > huge differences on the LCR module. there is a script somewhere (read wiki) that upgrades the tables. > I have these main questions: > > 1- does lcr_id replaces the grp_id? not really. lcr_ids denote independent lcr instances. > 2- what happened to the fifo commands to add gws etc: fifo commands to manipulate lcr tables are not supported by me. -- juha From: fborot(a)hotmail.com To: sr-users(a)lists.sip-router.org Subject: Questions about LCR module Date: Mon, 27 Jun 2011 17:23:05 -0400 I am trying to port our setup from kamailio 1.4.3 to 3.1.4 and I see huge differences on the LCR module. I have these main questions: 1- does lcr_id replaces the grp_id? 2- what happened to the fifo commands to add gws etc: lcr addgw <gw_name> <ip> <port> <scheme> <transport> <grp_id> <flags> <tag> <strip> ............... add a gateway with flags, tag and strip ............ ................(flags, tag, and strip are optional arguments) ..... lcr rmgw <gw_name> .......................................................... ............... delete a gateway lcr addroute <prefix> <from> <grp_id> <prio> ................................. .............. add a route ( use '' to match anything in <from> ) lcr rmroute <prefix> <from> <grp_id> <prio> ................................. .............. delete a route 3- it looks like the kamctl fifo lcr reload can not be used anymore, instead I found that there is a sercmd command that can be used to "show" and "reload" only the lcr tables What would be the best way to use this module on this version? Any help/hint would be greatly appreciated. thank you fborot

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Kamailio+ Asterisk 491 error code ????
by Sébastien Cramatte 28 Jun '11

28 Jun '11

Hello, I've just setup Kamailio+ Asterisk following Daniel Constantin Mierla howto http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb Everything seemts to works and I'm able to place a call to local extensions. I'm able to establish a call but strangely I get and 491 error code on my SIP Phones In my Asterisk console I get this error : [Jun 28 18:14:42] WARNING[24432]: chan_sip.c:18087 handle_response_invite: just did sched_add waitid(23) for sip_reinvite_retry for dialog LIFuQLu1l3ZtKrNy0XDX7lEk.Y5JdFko in handle_response_invite == Spawn extension (from-local, 102, 1) exited non-zero on 'SIP/103-00000000' Anybody can give me an hand to solve this issue ? Best regards

2 1

Dialog module question
by Javier Gallart 28 Jun '11

28 Jun '11

Hello I have little experience with the dialog module -so sorry if the quesiton is silly-, and I was wondering if I could use it for doing accounting (acc works just fine, but I was looking for a unique stop record that includes all the call information). The basic idea would be to assign the initial INVITE to a profile; and then, when BYE is received, build a cdr based of the dialog attributes...but afaik , upon reception of the BYE the dialog is destroyed. is there a way to access the dlg attributes inside the script when the BYE transaction is processed? Thanks in advance Javi

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Questions about LCR module
by Fabian Borot 28 Jun '11

28 Jun '11

I am trying to port our setup from kamailio 1.4.3 to 3.1.4 and I see huge differences on the LCR module. I have these main questions: 1- does lcr_id replaces the grp_id? 2- what happened to the fifo commands to add gws etc: lcr addgw <gw_name> <ip> <port> <scheme> <transport> <grp_id> <flags> <tag> <strip> ............... add a gateway with flags, tag and strip ............ ................(flags, tag, and strip are optional arguments) ..... lcr rmgw <gw_name> .......................................................... ............... delete a gateway lcr addroute <prefix> <from> <grp_id> <prio> ................................. .............. add a route ( use '' to match anything in <from> ) lcr rmroute <prefix> <from> <grp_id> <prio> ................................. .............. delete a route 3- it looks like the kamctl fifo lcr reload can not be used anymore, instead I found that there is a sercmd command that can be used to "show" and "reload" only the lcr tables What would be the best way to use this module on this version? Any help/hint would be greatly appreciated. thank you fborot

2 1

Re: [SR-Users] [sr-dev] new module: async - asynchronous SIP request processing in cfg
by Dan-Cristian Bogos 28 Jun '11

28 Jun '11

Hey Daniel, As said in the past, this module is good news, thanks for making it available. As a big fan of delayed billing, sleep() does a great job of delaying 200 OK from the upstream for me (useful for example with analogue/CDMA gateways sending 200 OK on ringback). Cheers, DanB

1 0

Kamailio and DTMF
by Phillman25 Kyriacou 28 Jun '11

28 Jun '11

Hi I was wondering what DTMF modes Kamailio supports? If i wanted to force a DTMF mode on Kamailio how could i go about doing it? thanking you in advance Phillip

3 3

rtpproxy and kamailio doesnt work out from the box.
by MingHon 28 Jun '11

28 Jun '11

hey list, i currently test on my asterisk 1.6 box on on pc with private ip and tested kamailio 3.1.3/3.1.4 + rtpproxy 1.2.1/1.2.0/1.1 on another pc with public ip and private ip. everything installed successfully. i follow this tutorial realtime intergration with asterisk http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb my config. rtpproxy -l publicip -s udp:127.0.0.1:7722 -u user #!define WITH_NAT modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") modparam("nathelper", "sipping_from", "sip:pinger@publicip") nat_uac_test("19") # uncomment next line to do SIP NAT pinging setbflag(FLB_NATSIPPING); is my config correct? may i know the best version work out from the box? please advice.. thanks in adv. :) -- Regards, MingHon

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new module: async - asynchronous SIP request processing in cfg
by Daniel-Constantin Mierla 27 Jun '11

27 Jun '11

Hello, a new module is now in GIT master branch, named async. Its purpose is to provide asynchronous SIP request processing using t_suspend()/t_continue() mechanism from tm module. While the tm module had it for quite some time, it was not available for usage in configuration file. There are two functions implemented by now: - async_route(routename, interval) - execute asynchronously a route block after a time interval - aync_sleep(interval) - sleep asynchronously for a time interval and then resume SIP request processing See the readme for more details: http://kamailio.org/docs/modules/devel/modules/async.html Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

4 4

LCR module
by Fabian Borot 27 Jun '11

27 Jun '11

After installing the latest stable version 3.1.4 I noticed that I can not run any 'lcr' commands from the 'kamctl fifo' interface. I noticed that the lcr.so is not on the lib64/modules_k or lib64/modules_s folders so I guess is was not included at installation time: [root@registrar kamailio]# cd /usr/local/kamailio-3.1/lib64/kamailio/modules_k/ [root@registrar modules_k]# ls acc.so db_text.so drouting.so mi_datagram.so permissions.so rtimer.so statistics.so usrloc.so alias_db.so db_unixodbc.so exec.so mi_fifo.so pike.so rtpproxy.so textops.so xlog.so auth_db.so dialog.so group.so msilo.so pua_mi.so siptrace.so tmx.so auth_diameter.so dispatcher.so htable.so nathelper.so pv.so siputils.so uac_redirect.so benchmark.so diversion.so imc.so nat_traversal.so qos.so speeddial.so uac.so call_control.so domainpolicy.so kex.so path.so registrar.so sqlops.so uri_db.so cfgutils.so domain.so maxfwd.so pdt.so rr.so sst.so userblacklist.so [root@registrar modules_k]# cd /usr/local/kamailio-3.1/lib64/kamailio/modules_s/ [root@registrar modules_s]# ls acc_db.so avp.so diversion.so fifo.so nathelper.so pike.so speeddial.so uri_db.so acc_syslog.so db_ops.so domain.so gflags.so options.so print.so textops.so uri.so auth_db.so dialog.so eval.so maxfwd.so pdt.so registrar.so timer.so usrloc.so avp_db.so dispatcher.so exec.so msilo.so permissions.so rr.so uac.so xlog.so [root@registrar modules_s]# I checked the modules.lst file and it is not on the exclude list, because I did removed it along with db_mysql and db_odbc, so I expected it to be included but it is not. Am I missing something? This is from the syslog Jun 27 11:44:00 registrar /usr/local/kamailio-3.1/sbin/kamailio[3974]: ERROR: mi_fifo [fifo_fnc.c:470]: fifo command lcr is not available Jun 27 11:44:00 registrar /usr/local/kamailio-3.1/sbin/kamailio[3974]: DEBUG: mi_fifo [fifo_fnc.c:549]: entered consume Jun 27 11:44:00 registrar /usr/local/kamailio-3.1/sbin/kamailio[3974]: DEBUG: mi_fifo [fifo_fnc.c:549]: **** done consume and this is from the fifo interface: [root@registrar kamailio]# /usr/local/kamailio-3.1/sbin/kamctl fifo lcr reload 500 command 'lcr' not available [root@registrar kamailio]#

1 1

async processing in a module
by Matthew Williams 27 Jun '11

27 Jun '11

Hi, I'm working on a jsonrpc (client) module that we would like to make async, using t_suspend and t_continue from the tm module. I have a couple questions related to this. The way I understand these functions, t_suspend will freeze the transaction in shared memory and immediately return and t_continue will unfreeze the transaction. What I'm not clear on here is whether t_continue will continue processing of the transaction within the process that calls t_continue, or will it delegate to a worker process? I assume I should spawn an io process in child_init which will be responsible for sending/receiving on the socket(s), but what is the best way to communicate between processes? Is there a way to get a fd from fork_process (that can be written to by any child and read from using select() in the io thread)? This is my first attempt at a real module, so any pointers are greatly appreciated. Regards, Matthew WIlliams

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