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11 participants
37526 discussions

Kamailio as a sip client
by Amit Nepal 11 Feb '11

11 Feb '11

Hi, Is it possible to have kamailio register to a different proxy and route calls through that proxy ? -- Thank You Amit Nepal Systems Administrator Phoenix Internet Phone: 602-385-0731 602-234-0917#112 http://www.phoenixinternet.net

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Kamailio 1.5.4 and beyond - mhomed issue under Linux
by Sean O'Donnell 10 Feb '11

10 Feb '11

Hi all, I just started deploying Kamailio release 1.5.4, and I think there's an issue with how Kamailio identifies an outgoing interface when mhomed is enabled under Linux. I use Kamailio as a call distributor/proxy between a soft-switch/SBC and a voicemail platform. It runs on a CentOS 5.3 (Linux 2.6 kernel) host with two network interfaces and is configured such that it listens on both interfaces. One interface (public interface) handles traffic with the SBC, the other (private interface) handles with the VM platform. The 'mhomed' option is enabled. After upgrading from 1.5.3 to 1.5.4, I started noticing problems with UDP packets coming out of the public interface. After looking at some ngrep captures on that interface, I noticed that some packets had the source IP address of the private interface and also had Record-Route and Via headers for the private interface only - no headers for the public interface were there. Usually when I see the wrong source IP in a UDP packet, it's an issue with how routes are set up on the host. However, I had our network engineer double check them, and they seem fine (no ambiguous routes). The fact that I captured these messages on the public interface also indicates to me that the kernel is routing the message correctly. The missing Record-Route and Via for the public interface, however, lead me to believe that the proxy didn't correctly identify the outgoing interface in the first place. After looking at the ChangeLog for 1.5.4, I noticed that the some new logic was put in to improve performance when mhomed is enabled (r5971) in forward.c, and I think this is the issue. As I understand it, prior to 1.5.4, when mhomed was enabled, Kamailio determined the outgoing interface by creating a temporary UDP socket, invoking connect() on the socket with the packet destination, then checking the source IP of the socket that the kernel assigned using getsockname(). After the source address was determined, the temp socket was closed closed. As of 1.5.4, this was modified to reuse the temporary socket and just re-invoking connect() with a new destination address. The problem with the enhancement is that Linux (again, at least in the 2.6 kernels I'm using) doesn't seem to rebind a new source address to the socket when connect() is called more than once on a UDP socket. Instead, it keeps the original one, and thus the wrong interface is assumed. I wrote a small program to confirm this - basically creates a UDP socket, calls connect()/getsockname() multiple times using different destination addresses. I ran it on several 2.6 kernels, including Centos4.x and Centos5. The result was always that the source address of the socket wasn't changed after the first connect(), regardless of the destination address. The only way I could get it work as required was to first do a connect() using a zero'd out AF_UNSPEC address before doing the connect() to the remote address. I also ran it on Solaris and it worked. Go figure. I've downloaded the latest stable release (3.1.2) but I think the issue is still there, and I don't see anything in the user groups that addresses this. Any help would be appreciated. Thanks, Sean

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Merging nathelper modules
by Ovidiu Sas 10 Feb '11

10 Feb '11

Hello all, The nathelper module in modules_k was split in two: - nathelper (dealing with signaling); - rtpproxy (dealing with rtpproxy protocol). I would like to move the rtpproxy module from "modules_k" into "modules" and remove rtpproxy functionality from nathelper (s). This will give to (s) and (k) users: - rtpproxy: a single module for dealing with rtpproxy servers; - nathelper: two variants for dealing with NAT signaling. Next step, will be to merge the two nathelper modules into a single one. Thoughts? Regards, Ovidiu Sas

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nathelper-module and sdp offer in 200 OK
by Emil Kroymann 10 Feb '11

10 Feb '11

Hi, We recently had a problem with the nathelper module and rtpproxy in a scenario where the SDP offer is sent only in the 200 OK. We use sip-router 3.1 and rtp-proxy from git master. The sip-router configuration uses the rtpproxy_offer() and rtpproxy_answer() functions in appropriate places. The problem is, that the arguments sent to the rtpproxy, when the ACK with the sdp answer arrives, seems to be not in the order, that rtpproxy expects. On the 200 OK, the nathelper module sends callid, to-tag, from-tag to rtpproxy. On the ACK, the nathelper module sends callid, from-tag, to-tag (with different command prefixes, of course, but I cannot remember them atm). The version of rtpproxy that we are using seems to expect, that the order of arguments sent on the ACK request is the same as on the 200 OK. My question: are there any module parameters, to correct this behaviour? Regards, Emil -- Emil Kroymann VoIP Services Engineer Email: emil.kroymann(a)isaco.de Tel: +49-30-203899885 ISACO GmbH Kurfürstenstraße 79 10787 Berlin Germany Amtsgericht Charlottenburg, HRB 112464B Geschäftsführer: Daniel Frommherz

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Use different certificate for different client with TLS
by Daniel GROTTI 09 Feb '11

09 Feb '11

Hi all, I would like to use kamailio 3.1 with TLS and verified also a client certificate. My tls.cfg file is as follow: --- tls.cfg ---- ..... ..... [server:MY_IP:5061] method = TLSv1 verify_certificate = yes require_certificate = yes private_key = default_key.pem certificate = default_cert.pem ca_list = default_ca.pem [client:default] verify_certificate = no require_certificate = no .... .... So I want to verify the client certificate, to do that I use "is_peer_verified()" function in kamailio.cfg. As tls.cfg shows, I have to send to my clients the CA certificate and the client certificate (default_cert.pem + default_key.pem - signing by the CAcert). This client certificate is unique for all clients. Everything works fine. But suppose I wanted to create a client certificate for client 1 (cert_1.crt), and a different client certificate for client 2 (cert_2.crt) and I want to configure kamailio to be able to verified this different certificates. Does it possible ? How can I configure the tls.cfg file to do that ? I try to do something like this: [server:MY_IP:5061] method = TLSv1 verify_certificate = yes require_certificate = yes private_key = default_key_1.pem certificate = default_cert_1.pem private_key = default_key_2.pem certificate = default_cert_2.pem ca_list = default_ca.pem But when kamailio restart it seems that it read only the last couple of row certifcate/private_key. Regards, Daniel G

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[OpenSER-Users] uaCSTA support in openser
by Hugo Koblmueller 09 Feb '11

09 Feb '11

high.all! i'm wondering if there is any support of uaCSTA in openser (planned)? i'm just working on the integration of asterisk (*) environment to OCS 2007 environment, having openSER in the middle (mainly for TCP/UDP translation and smoothing out the protocol deficienes on both sides). in this setup the * having the openSER in front is talking to the OCS (and vice versa) via the OCS mediation server, which is moreorless sending standard SIP messages, which enables normal softphone (integration to *) of the office communicator. this configuration is already working... now i'm planning to go for the CTI integration, where there is no OCS mediation server in between OCS and openSER, doing the translation of SIP/CSTA to SIP. i'm thinking about using openSER for this task, that's why i'm looking for a CSTA module or perl programm, which is capable of this functionality. afaik for the CTI communication there isn't the full complexity of CSTA needed, just a subset mainly for call setup and call clearing. anyone having experience on this topic? thx & cheers -hugo Great Ideas for Small Devices Hugo Koblmueller Senior Staff Engineer Software Development COMNEON electronic technology GmbH & Co. OHG Freistaedter Strasse 400 4040 Linz Austria hugo.koblmueller(a)comneon.com tel: fax: mobile: Skype ID: +43 (5) 1777 - 15730 <http://www.plaxo.com/click_to_call?lang=en&src=jj_signature&To=%2B43+%285%2 9+1777+%2D+15730&Email=hugo(a)koblmueller.com> +43 (5) 1777 - 15810 +43 (676) 82051280 <http://www.plaxo.com/click_to_call?lang=en&src=jj_signature&To=%2B43+%28676 %29+82051280&Email=hugo(a)koblmueller.com> drhookson Want to always have my latest info? <https://www.plaxo.com/add_me?u=21475050628&src=client_sig_212_1_banner_join &invite=1&lang=en> Want a signature like <http://www.plaxo.com/signature?src=client_sig_212_1_banner_sig&lang=en> this?

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Re: [SR-Users] module postgres SER
by Daniel-Constantin Mierla 09 Feb '11

09 Feb '11

please keep the mailing list cc-ed, sending private messages is not in the spirit of public mailing lists. Others may want to follow up the discussion now or later. Thanks, Daniel On 2/9/11 5:01 PM, Bruno Bresciani wrote: > Daniel, > > thanks for your reply, really ser-0.8.1.4 is too old but i need to > solve this problem on that version. My great doubt is Why the aug_free > function corrupt the url of database after some attemps to reconnect. > Well, I'll try to understand this question... > > Best Regards > > > 2011/2/9 Daniel-Constantin Mierla <miconda(a)gmail.com > <mailto:miconda@gmail.com>> > > Hello, > > > On 2/9/11 2:19 PM, Bruno Bresciani wrote: > > Hi, > > I've seen the problem in a postgres module (SER-0.8.1.4), if > the connection fails and module tries to reparse url it fails > as CON_SQLURL(_h) is corrupted by the function aug_free() > after some reconnect attempts . When the postgres database > back to work, some modules doesn't get reconnect because the > db_url is corrupted. Why this is happening? There are some > solution for this problem? > > ser 0.8.1.4 is soooo old and I cannot fully remember, but I think > postgres module had no reconnect functionality at all by that time. > > However, version 3.1.x of SER (as well as Kamailio flavour) has db > reconnect functionality for postgres. You can try it and report if > something is not working, it will be fixed in 3.x, but I think > nobody is still developing on 0.8.x to backport anything there. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla > http://www.asipto.com > > -- Daniel-Constantin Mierla http://www.asipto.com

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problem with bye using rtpproxy
by Amit Nepal 09 Feb '11

09 Feb '11

I have been trying to figure this out While using kamailio and rtpproxy, the caller is not receiving the bye when callee hangs up but audio is two way and everything seems to be working fine, any one had this issue ? -- Thank You Amit Nepal Systems Administrator Phoenix Internet Phone: 602-385-0731 602-234-0917#112 http://www.phoenixinternet.net

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Multiple call accounting from an IP
by Amit Nepal 09 Feb '11

09 Feb '11

Hi everyone, I am sure someone have been working with this scenario, how about accounting multiple calls from same account or ip address while using ip auth ? -- Thank You Amit Nepal Systems Administrator Phoenix Internet Phone: 602-385-0731 602-234-0917#112 http://www.phoenixinternet.net

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fosdem 2011 presentation about p_usrloc
by Henning Westerholt 09 Feb '11

09 Feb '11

Hello, this year the FOSDEM developer conference was a again a really nice event, the first time with an own room dedicated completely to open source telephony solutions! If you're interested in our presentation about the new p_usrloc module and how to scale location services with Kamailio, you can find it at the usual place on our webserver: http://kamailio.org/events/2011-fosdem/p_usrloc.pdf Best regards, Henning

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