Hi, thanks for your reply and sorry for my late answer.
I checked my sipxecs settings and checked the link that you gave to me in your post. Everything is set properly. The situation is this after tests. When from hardware voip phone we do outbound calls, then there is voice in both sides, but when it receives calls, then there is no voice in both sides. When we do calls using sofphones everything is ok. They work properly for inbound and outbound calls, without using STUN! The model of the hardware phone: Siemens Gigaset C470IP The phones are behind remote NAT. The router in front is: Linksys WAG200G (Firmware 1.01.05), this one is also a DSL modem We have the same model of hardware phone in other place too and we have the same problem there too. The router there in front is: D-Link DIR-655 with firmware 1.21EU Important: SIP Settings are deactivated in routers. We tried these settings in the hardware phone, but still not success: STUN: no Proxy server address: mydomain.tld Proxy server port: 5060 Registrar server: mydomain.tld Registrar server port: 5060 Registration refresh time: 180 sec. NAT refresh time: 20 sec. Outbound proxy mode: always Outbound proxy: mydomain.tld Outbound port: 5060 Also these without proxy settings: STUN: no Registrar server: mydomain.tld Registrar server port: 5060 Registration refresh time: 180 sec. NAT refresh time: 20 sec. So we have these results at the end: outbound calls from siemens phone -> (works) voice in both sides (with and without using proxy settings) inbound calls to siemens phone -> (does not work) no voice in both sides (with and without using proxy settings) using STUN on hardware phone -> works for outbound and inbound calls On Fri, 2009-11-20 at 09:18 -0500, Robert Joly wrote: > > we have installed 4.0.2-016420 on Centos 5. We have applied > > latest patch for sipXbridge too (4.0.4). > > > > We are experience a problem with audio on our calls. > > > > When we do internal calls from extension to extension phones > > are ringing, but we don't have audio when we pickup the > > phones. > > Are these phones behind remote NATs or not? > > In this case we don't have set STUN on the phones. > > Especially this happens using hardware over IP phones. Using > > softphones seems to be ok. When we set STUN on hardware phone > > and the others, then the voice is comming back anad > > everything is ok. This problem appear if we call to a real > > number and on the line is put hardware voip phone. > > Can you specify the brand/model/firmware version of phones you are > using? > > Also, what is the brand/model/firmware version of the remote NATs you > use? Also please be sure to disable any SIP ALGs that may be enabled on > your remote NATs. > > You should not have to run STUN on the phones. In fact, we strongly > recommend that you don't if you're using sipXecs's remote NAT traversal > feature. > > > > How to deal with this problem? I don't see any problems in > > the logs. I know that this is related with NAT Traversal. > > > > Server is not behind NAT and it has properly configured DNS > > records, and it uses static IP. > > > > My NAT settings on sipxecs are this way: > > > > System -> Internet Calling -> NAT Traversal -> Enable NAT > > Traversal (checked), Server behind NAT (not checked). > > > > System -> Servers -> choose the server -> NAT -> Specify IP Address. > > ( as i understand, this option is related to way if Server is > > behind NAT or not, or if server has dynamic address. I > > suppose that this setting does not have any relation with > > remote workers behind NAT). > > This setting is used by the remote NAT traversal feature. Just put the > public IP address of your sipXecs in there. > > > > > > Is there a way to set in sipxecs remote workers which stun > > server to use, or something like this or there is another solution? > > > > So how to go with this problem? > > Please verify that you have followed all the steps in > http://sipx-wiki.calivia.com/index.php/Configuring_remote_workers_cheats > heet > > Pay special attention to step #3. In your case, you need to remove all > the default values and add the subnet that sipXecs is a part of. > > > > > > I found this thread: > > http://www.mail-archive.com/[email protected]/msg > 05291.html > > > > My problem is the same or very similar. What i expect is to > > not be necessary to set on the phones (hardware, softphone, > > etc.) stun. I suppose that this functionality must be covered > > from sipXbridge, but maybe i'm wrong. > > > > If you need log parts, please tell me and i will post logs here. > > > > Thanks in advanced! > > > > > > > > _______________________________________________ > > sipx-users mailing list [email protected] List > > Archive: http://list.sipfoundry.org/archive/sipx-users > > Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users > > sipXecs IP PBX -- http://www.sipfoundry.org/ > > _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users sipXecs IP PBX -- http://www.sipfoundry.org/
