Thank you for your replay Tony. But how can I use SBC with a single phone? I thought that SBC can be "fixed" onlu using a trunk?
On Tue, 12 Jul 2011 04:09:00 -0400, Tony Graziano <[email protected]> wrote: > because sip provider has a way handle your device randomizing portal. > that's what a sbc does.. that device was designed for consumer use . > sipx is a system geared toward businesses. either get border > controller with that phone or get another phone. > On Jul 12, 2011 3:40 AM, wrote: >> I tried everything i could, and it still doesn't work. >> An interesting thing is that there are 2 more connections via SIP > > between this phone and other SIP-providers, and everything is > fine! >>>> >>>> On Fri, Jul 8, 2011 at 4:48 AM, wrote: >>>> Hey there! >>>> This is my first post in sipx-users mailing list, so I'm sorry > if > >>> I do >>>> something wrong. >>>> Well.. the problem is strange enough, and I can't find out how > to >>>> fix >>>> it during 2 days, hm.. >>>> So, i've got a sipX station, and it has public IP. So "Server > >>> behind >>>> NAT" is unchecked. >>>> There are about 5 users, whos phones use NAT. Everything is ok >>>> with >>>> softphones and SIP mobile phones. The only settings i fix in > the > >>> softphones and SIP mobiles phones are REG server and >>>> Domain/Outbound >>>> Proxy server. Everything works fine! I use this pretty good > guide >>>> >>>> >>>> > > _http://wiki.sipfoundry.org/display/sipXecs/Remote+User+NAT+Traversal > [2] > >>> [3] >>>> But! The only phone (Siemens Gigaset C470 IP) doesn't work as > good >>>> as i >>>> need. The scheme of my VoIP network looks like this: >>>> SIPX Server - > IP-network -> Remote NAT + Siemens Gigaset C470 > IP > >>> So the sipX station does'n use any nat, but the phone does. > SIP is >>>> ok, >>>> so I can call this user and he can call me. When the call goes >>>> from >>>> Siemens Gigaset C470 IP we can speak and everything is ok. But > if > >>> I call >>>> him, there is no media between our phones. So the remote user > can >>>> hear >>>> the ring, pick up the phone, but we can't hear each other.. >>>> I've already tried to fix everything I could. There are those > >>> types of >>>> options in the Siemens Gigaset C470 IP web GUI: >>>> >>>> Domain >>>> Proxy server address and port >>>> REG server addres and port > >>> STUN yes/no >>>> STUN server >>>> Outbound proxy server and port >>>> >>>> I've alredy tried different combinations of those settings > (even >>>> checking "use STUN"), but there is no result. No media when > anyone > >>> call >>>> him. >>>> I will be happy if someone could help me to find out where the >>>> problem >>>> is. >>>> Regards, kga. >>>> >>>> _______________________________________________ > >>> sipx-users mailing list >>>> [email protected] [3] [4] >>>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ [4] > [5] > >>> >>>> -- >>>> ====================== >>>> Tony Graziano, Manager >>>> Telephone: 434.984.8430 >>>> sip: [email protected] [5] [6] > >>> Fax: 434.326.5325 >>>> >>>> Email: [email protected] [6] [7] >>>> >>>> LAN/Telephony/Security and Control Systems Helpdesk: > >>> Telephone: 434.984.8426 >>>> sip: [email protected] [7] [8] >>>> >>>> Helpdesk Contract Customers: >>>> http://support.myitdepartment.net [8] [9] > >>> >>>> [10]Blog: >>>> http://blog.myitdepartment.net [9] [11] >>>> >>>> Linked-In >>>> Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 > [10] [12] > >>> >>>> >>>> Links: >>>> ------ >>>> [1] >>>> >>>> > > http://wiki.sipfoundry.org/display/sipXecs/Display+SIP+message+flow+using+Sipviewer > [11] > >>> [2] mailto:[email protected] [12] >>>> [3] >>>> > http://wiki.sipfoundry.org/display/sipXecs/Remote+User+NAT+Traversal > [13] > >>> [4] mailto:[email protected] [14] >>>> [5] http://list.sipfoundry.org/archive/sipx-users/ [15] > >>> [6] mailto:[email protected] [16] >>>> [7] mailto:[email protected] [17] > >>> [8] mailto:[email protected] [18] >>>> [9] http://support.myitdepartment.net [19] >>>> [10] http://support.myitdepartment.net [20] > >>> [11] http://blog.myitdepartment.net [21] >>>> [12] http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 [22] > > >> _______________________________________________ >> sipx-users mailing list >> [email protected] [23] >> List Archive: http://list.sipfoundry.org/archive/sipx-users/ [24] > > > Links: > ------ > [1] mailto:[email protected] > [2] > http://wiki.sipfoundry.org/display/sipXecs/Remote+User+NAT+Traversal > [3] mailto:[email protected] > [4] http://list.sipfoundry.org/archive/sipx-users/ > [5] mailto:[email protected] > [6] mailto:[email protected] > [7] mailto:[email protected] > [8] http://support.myitdepartment.net > [9] http://blog.myitdepartment.net > [10] http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 > [11] > > http://wiki.sipfoundry.org/display/sipXecs/Display+SIP+message+flow+using+Sipviewer > [12] mailto:[email protected] > [13] > http://wiki.sipfoundry.org/display/sipXecs/Remote+User+NAT+Traversal > [14] mailto:[email protected] > [15] http://list.sipfoundry.org/archive/sipx-users/ > [16] mailto:[email protected] > [17] mailto:[email protected] > [18] mailto:[email protected] > [19] http://support.myitdepartment.net > [20] http://support.myitdepartment.net > [21] http://blog.myitdepartment.net > [22] http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 > [23] mailto:[email protected] > [24] http://list.sipfoundry.org/archive/sipx-users/ _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/
