sbc's can do trucking remote users or both. it sounds like it would be easier to use a different phone with it On Jul 12, 2011 4:13 AM, <[email protected]> wrote: > 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/
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