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37456 discussions

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?

2 1

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

29 Aug '11

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

2 1

enum_pv_query
by Andrew Melton 28 Aug '11

28 Aug '11

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

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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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