Yes sorry, it is a typo. I meant public ip...
But indeed, it does go once again through the firewall. I am thinking the problem is here: In the tcpdump (on kamailio) I see it says incoming call from: sip:number@IP OF CARRIER Ans it is forwarded this way to pbx I "think" that this way the PBX will send a reply directly to the IP of the carrier for RTP. When I look at the tcpdump (on the pbx) it does seem this way. (I am still trying to comprehend the whole sip dialog, learning bit by bit) So, I am trying to change the "from" and "to" header to use the ip of kamailio, and not of the carrier. For this I am using uac module. But no success until now... I had to do this to get call from extension on pbx, through kamailio , to carrier to work (And this works (hurray)). (Maybe I'm saying stupid stuff and I'm totally wrong) Thanks a lot for the help! Arnout Op di 17 sep. 2019 om 11:01 schreef Daniel-Constantin Mierla < [email protected]>: > Hello, > > might be just a typing mistake, but: > > rtpproxy -A "private ip" -F -l "local ip" -m 10000 -M 20000 -s udp:*:7722 > -d INFO > > says that -A is private ip, it should be public IP. > > Then, towards asterisk, is it everything on a private network, or does it > go again through some FW with public ip and port forwarding? > > Cheers, > Daniel > > On Tue, Sep 17, 2019 at 10:23 AM Arnout Van Den Kieboom < > [email protected]> wrote: > >> Hi, >> >> Firewall (PFsense) does port forwarding: >> >> inbound publicIP:5060 --> private ip (192.168.150.119) >> >> inbound PublicIP:10000 to 20000 -> private ip (192.168.150.119) >> >> The public ip is set in the config file (kamailio.cfg) and RTP proxy is >> started with >> >> rtpproxy -A "private ip" -F -l "local ip" -m 10000 -M 20000 -s >> udp:*:7722 -d INFO >> >> In the cfg file NATtin is defined. (define with_nat) >> rtpproxy module is loaded. >> >> This works for most audio situations, except with the carrier >> Also very important: the carrier i am using for testing requires >> authentication, so I am using the uac module. >> >> I can see the registration with the provider, i see the inbound call in >> kamailio, I see the inbound call in asterisk. >> A connection is set up like it should, but then there is One way audio. >> >> The weird thing is: >> this route works without a problem: carrier --> FW -->kamailio --> (back >> through firewall) FW --> user subscribed to kamaiolio. >> (which should imply the same NAT'ing problems?) >> >> If you need more info, feel free to ask! >> >> PS: What i haven't tried yet is to call from asterisk through kamailio to >> the carrier number, this is what i'll set up and test next) >> >> Arnout >> >> Op ma 16 sep. 2019 om 13:26 schreef Daniel-Constantin Mierla < >> [email protected]>: >> >>> Hello, >>> >>> is Kamailio and RTPProxy (or Asterisk) using public IP addresses? Or >>> they listen on a private address and the firewall does port forwarding of a >>> public IP address? >>> >>> Cheers, >>> Daniel >>> >>> On Mon, Sep 16, 2019 at 10:51 AM Arnout Van Den Kieboom < >>> [email protected]> wrote: >>> >>>> Hi, >>>> >>>> First of all, I'm really new to Kamailio. So sorry if I ask a stupid >>>> question, or perhaps a really weird one. >>>> >>>> I started using kamailio in a test environment about a week ago. >>>> >>>> for setup i have the situation for inbound calls: Carrier --> FW (NAT) >>>> --> Kamailio >>>> for outbound to an asterisk box i have: Kamailio --> FW (NAT) --> >>>> Asterisk >>>> for outbound to users i have: Kamailio --> FW (NAT) --> sip-phone >>>> (yealink) or grandstream >>>> >>>> During that time i was able to : >>>> Set up calls to users, >>>> Use the dispatcher module >>>> use the avpops module to do data lookups. >>>> Get calls from asterisk to kamailio user to work. >>>> Also calls between users (even though behind nat) are having good audio. >>>> >>>> I have installed rtpproxy and it works nicely. >>>> >>>> There is just one situation where it fails: >>>> >>>> Carrier --> FW (NAT) Kamailio --> asterisk >>>> >>>> There is only one way audio (from asterisk to carrier) but not from >>>> carrier to asterisk. >>>> >>>> I believe I need to do something with the rtp proxy module ... and >>>> tried different things (forcing the r flag, forcing the w flag) >>>> Any idea where I might need to start looking? It's been driving me >>>> crazy... >>>> >>>> Thanks >>>> >>>> Pemp >>>> >>>> >>>> _______________________________________________ >>>> Kamailio (SER) - Users Mailing List >>>> [email protected] >>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>> >>> >>> >>> -- >>> Daniel-Constantin Mierla - https://www.asipto.com >>> https://twitter.com/miconda - https://www.linkedin.com/in/miconda >>> Kamailio Advanced Training - https://www.asipto.com/u/kat >>> _______________________________________________ >>> Kamailio (SER) - Users Mailing List >>> [email protected] >>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>> >> _______________________________________________ >> Kamailio (SER) - Users Mailing List >> [email protected] >> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> > > > -- > Daniel-Constantin Mierla - https://www.asipto.com > https://twitter.com/miconda - https://www.linkedin.com/in/miconda > Kamailio Advanced Training - https://www.asipto.com/u/kat > _______________________________________________ > Kamailio (SER) - Users Mailing List > [email protected] > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >
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