sr-users November 2016

sr-users@lists.kamailio.org

57 participants
123 discussions

Re: [SR-Users] kamailio + farm of Elastix + rtp direct

by Kato

Any comments on this please? Thanks! De: Kato [mailto:elkato@gmail.com] Enviado el: domingo, 6 de noviembre de 2016 11:29 a. m. Para: 'sr-users(a)lists.sip-router.org' <sr-users(a)lists.sip-router.org> Asunto: kamailio + farm of Elastix + rtp direct Hi, I'm just starting with Kamailio, and trying to validate a possible solution design. I started offering hosted PBXs (FreePBX or Elastix as customer wish), each one with a public IP, connected with SIP trunks from Net2Phone provider. RTP from/to customers calls is going in and out through PBX, and not directly customer <----> sip trunk provider. Billing is provided by SIP trunk provider, so with just CDR from freepbx/Elastix is enough. But, I would like to introduce 2 main changes. First a sip router/proxy in front of PBXs like kamailio, and doing just a basic domain based routing to each PBX. Kamailio with 2 interfaces (1 public, 1 internal) and change PBXs to internal IPs. Second, to know if would be possible to redirect RTP from/to customer-net2phone provider directly, not going through PBXs. Net2phone has public IPs for RTP, customer normally in their homes behind NAT with a SOHO router. Elastix 2.5 using Asterisk 11, and FreePBX using Asterisk 12 (could be 11 or 13 too tbe o reach this). I would like to kamilio be as simple and static as possible, not handling REGISTER or LOCATION. Just to route all traffic based on each domain to each corresponding PBX. My idea will be to pre-provide all subdomains to each internal IP using subdomains like "customer0001.domain.com, customer0002.domain.com." and customers would login as 100(a)customer0001.domain.com <mailto:100@customer0001.domain.com> , 101(a)customer0001.domain.com <mailto:101@customer0001.domain.com> to PBX0001 and 100(a)customer0002.domain.com <mailto:100@customer0002.domain.com> , 101(a)customer0002.domain.com <mailto:101@customer0002.domain.com> to PBX0002 for example. Just using as it was a firewall doing NAT and Asterisk (freepbx/elastix) behinds handles everything as they are doing already. Maybe kamailio could check/filter some malicious packets like I'm doing now with fail2ban on PBXs. Can be this possible? In that case where do I need to setup SIP trunk? On kamailio or on internal PBXs as I have it now? For second part, just to know and understand if would be possible customer "talks" RTP directly to Net2phone SIP trunk provider on doing or receiving external calls. Not going to my DC to save BW. How I could do that? Thanks a lot in advance! Regards, elkato

8 years

Next Kamailio IRC Devel Meeting, Nov 09, 2016

by Daniel-Constantin Mierla

Hello, I am proposing a new IRC meeting to discuss the current major issues and the plans for next Kamailio release, on Wednesday, Nov 09, 2016, with alternatives for previous or next day. If many developers are not available, we can postpone it to another date in the near future (make proposals if that is the case for you). I created a wiki page for it: * https://www.kamailio.org/wiki/devel/irc-meetings/2016b Add there the topics that you want to be discussed. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

8 years

Contacts replacing each other when registering

by Sebastian Damm

Hi, we are running a Registrar server with 4.4.3, the registrar module is configured with the following parameters: modparam("registrar", "max_contacts", 50) modparam("registrar", "use_path", 1) modparam("registrar", "path_mode", 0) modparam("registrar", "path_use_received", 1) modparam("registrar", "received_avp", "$(avp(i:801))") modparam("registrar", "default_expires", 600) modparam("registrar", "min_expires", 60) modparam("registrar", "max_expires", 600) modparam("registrar", "gruu_enabled", 0) A customer has configured 4 phones with the same username. Normally, all 4 phones would be visible in the location database, and incoming calls would fork out to all 4 phones. However, only one of the devices is always online, the others always replace each other on each registration. Client details: * All phones are snom 715 devices with the same firmware version. * All phones sit behind the same router. Registrations: client 1: * Via: SIP/2.0/UDP 192.168.208.91:35812;branch=z9hG4bK-4q0mmxgu6gm7;rport * Contact: <sip:1794388e3@192.168.208.91:35812;line=gicode6v>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:0a14174a-77be-424c-864e-0004137F1C9D>" client 2: * Via: SIP/2.0/UDP 192.168.208.90:41331;branch=z9hG4bK-3cbxdvbmk0di;rport * Contact: <sip:1794388e3@192.168.208.90:41331;line=0j6veso8>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:0a14174a-77be-424c-864e-0004137F1C9D>" client 3: * Via: SIP/2.0/UDP 192.168.208.92:5360;branch=z9hG4bK-4drsogakez94;rport * Contact: <sip:1794388e3@192.168.208.92:5360;line=0n7jfzih>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:0a14174a-77be-424c-864e-0004137F1C9D>" client 4: * Via: SIP/2.0/UDP 192.168.208.93:5260;branch=z9hG4bK-1ci0qia3yxyl;rport * Contact: <sip:1794388e3@192.168.208.93:5260;line=mh9sgxa1>;reg-id=1;q=1.0 Only client 4 is reachable all the time. The other 3 clients replace other registrations. So after client 1 registers, the contacts returned in the 200 OK looks like this: Contact: <sip:1794388e3@192.168.208.91:35812;line=gicode6v>;q=1;expires=600;received="sip:1.2.3.4:6411";+sip.instance="<urn:uuid:0a14174a-77be-424c-864e-0004137F1C9D>";reg-id=1, <sip:1794388e3@192.168.208.93:5260;line=mh9sgxa1>;q=1;expires=303;received="sip:1.2.3.4:8923" And after client 2 registers, the contacts returned in the 200 OK looks like this: Contact: <sip:1794388e3@192.168.208.90:41331;line=0j6veso8>;q=1;expires=600;received="sip:1.2.3.4:49061";+sip.instance="<urn:uuid:0a14174a-77be-424c-864e-0004137F1C9D>";reg-id=1, <sip:1794388e3@192.168.208.93:5260;line=mh9sgxa1>;q=1;expires=301;received="sip:1.2.3.4:8923" So you see, one contact stays the same, the other one gets replaced. Does anybody know what could cause this kind of behavior? I think, it could be GRUU. I see that those three clients all send the same GRUU parameter. From my understanding, this UUID should be different from client to client. I don't know whether this is a bug in the snom firmware always sending the same GRUU UUID or whether som "intelligent" SIP aware router is putting this in the Contact header. However, since we have disabled gruu for the registrar module (see above), I think, Kamailio should ignore it. Ideas? Thanks in advance. Best Regards, Sebastian

8 years

TCP Keepalive for NAT'd Clients

by Colin Morelli

Hey all, Looking to figure out the best way to allow TCP connections to stay alive for NAT'd clients, however, to protect against people just opening TCP connections to the server, I'm hoping to only keepalive TCP connections for connections that have sent an INVITE and received a 200 OK. Does this type of per-socket connection lifetime setting exist? For a bit more background: this is specifically for clients that are using the TCP connection for an INVITE. They don't register with the server, and as a result there's no active flow timer for their connection. Is there a way to enable this type of behavior without registering? Best, Colin

8 years

How to get Destination port UDP

by andrzej.ciupek-asterisk.edu.pl

Hello I am using Kamailio to collect data about active calls. I write INVITEs to database, and delete it on BYE or any REL. I would like to write source and destination port too, but don't know how to get destination port. I have tried variables: $dp, $rp, $Rp, $op But when I don't have port in URI like INVITE IP:PORT, $rp have default value of 5060. I would like to get real UDP port, that INVITE does to. Is it possible using PSEUDO VARIABLES? Greetings Andrzej

8 years

Out of memory in UB 210: OOM killed process 12261 (kamailio) score 0 vm:1614768kB, rss:280200kB, swap:131408kB

by Jurijs Ivolga

Hi, Our Kamailio server is crashing once per week, with following error: Oct 1 06:25:06 kamailio kernel: [26982632.803789] Out of memory in UB 210: OOM killed process 12261 (kamailio) score 0 vm:1614768kB, rss:280200kB, swap:131408kB Core dump was never created, probably it is because of my environment, but I will try to get it. Server constantly eats memory, maybe some kind of memory leak? Any help is highly appreciated! Jurijs

8 years

NER and ASR

by Gholamreza Sabery

Are there any real data about number of concurrent calls, number of on-line users, NER (Network Effectiveness Ratio) and ASR (Answer Seizure Ratio) of a real VoIP system? I want this data for making comparisons. Can anyone provide a reference, article, etc..

8 years

cdr on remote POSTGRES server

by beer Ll

Hi all I have a kamailio server configured with accounting on a separate Postgres SQL server by an internal network with acc module and db_postgres module Everything works good but what can I do if I have problem on the internal network or on the postgres server ? Is it possible cache or save the data if the insert data is failed ?

8 years

fnmatch

by Slava Bendersky

Hello Everyone, I trying fnmatch to find which sip status is returned by B2BUA. But some reason it doesn't work. Nov 6 17:02:15 cavprx00 kamailio: 2(3747) ERROR: *** cfgtrace:onreply_route=[MESSAGE_FORWARD] c=[/etc/kamailio/kamailio-asterisk.cfg] l=456 a=26 n=xlog Nov 6 17:02:15 cavprx00 kamailio: 2(3747) INFO: <script>: Message 1 Accepted by B2BUA --> 202 with Accepted Nov 6 17:02:15 cavprx00 kamailio: 2(3747) ERROR: *** cfgtrace:onreply_route=[MESSAGE_FORWARD] c=[/etc/kamailio/kamailio-asterisk.cfg] l=469 a=16 n=if Nov 6 17:02:15 cavprx00 kamailio: 2(3747) ERROR: *** cfgtrace:onreply_route=[MESSAGE_FORWARD] c=[/etc/kamailio/kamailio-asterisk.cfg] l=457 a=26 n=fnmatch Nov 6 17:02:15 cavprx00 kamailio: 2(3747) ERROR: *** cfgtrace:onreply_route=[DEFAULT_ROUTE] c=[/etc/kamailio/kamailio-asterisk.cfg] l=370 a=2 n=return if(fnmatch("$rs", "200") && (fnmatch("$rr", "Accepted"))) { xlog("L_INFO", "Message Accepted by B2BUA --> $rs with $rr \n"); $var(tu) = $tu; xlog("L_INFO", "New To --> $var(tu)\n"); } Is this fnmatch correct for this operation ? Slava.

8 years

kamailio + farm of Elastix + rtp direct

by Kato

Hi, I'm just starting with Kamailio, and trying to validate a possible solution design. I started offering hosted PBXs (FreePBX or Elastix as customer wish), each one with a public IP, connected with SIP trunks from Net2Phone provider. RTP from/to customers calls is going in and out through PBX, and not directly customer <----> sip trunk provider. Billing is provided by SIP trunk provider, so with just CDR from freepbx/Elastix is enough. But, I would like to introduce 2 main changes. First a sip router/proxy in front of PBXs like kamailio, and doing just a basic domain based routing to each PBX. Kamailio with 2 interfaces (1 public, 1 internal) and change PBXs to internal IPs. Second, to know if would be possible to redirect RTP from/to customer-net2phone provider directly, not going through PBXs. Net2phone has public IPs for RTP, customer normally in their homes behind NAT with a SOHO router. Elastix 2.5 using Asterisk 11, and FreePBX using Asterisk 12 (could be 11 or 13 too tbe o reach this). I would like to kamilio be as simple and static as possible, not handling REGISTER or LOCATION. Just to route all traffic based on each domain to each corresponding PBX. My idea will be to pre-provide all subdomains to each internal IP using subdomains like "customer0001.domain.com, customer0002.domain.com." and customers would login as 100(a)customer0001.domain.com <mailto:100@customer0001.domain.com> , 101(a)customer0001.domain.com <mailto:101@customer0001.domain.com> to PBX0001 and 100(a)customer0002.domain.com <mailto:100@customer0002.domain.com> , 101(a)customer0002.domain.com <mailto:101@customer0002.domain.com> to PBX0002 for example. Just using as it was a firewall doing NAT and Asterisk (freepbx/elastix) behinds handles everything as they are doing already. Maybe kamailio could check/filter some malicious packets like I'm doing now with fail2ban on PBXs. Can be this possible? In that case where do I need to setup SIP trunk? On kamailio or on internal PBXs as I have it now? For second part, just to know and understand if would be possible customer "talks" RTP directly to Net2phone SIP trunk provider on doing or receiving external calls. Not going to my DC to save BW. How I could do that? Thanks a lot in advance! Regards, elkato

8 years

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sr-users November 2016