I find that 1:1 is best with access rules only allowing the ports you want, this way server always goes out with the same IP. Also, make sure the firewall does not do outbound port randomization.
On Thu, Oct 11, 2012 at 12:18 PM, Henry Kwan <[email protected]> wrote: > Do I need one-to-one NAT, or symmetric NAT? I bought this router because > it had been tested to work with VoIP, whatever that means, but I forgot the > source of this information. > > *From:* Tony Graziano <[email protected]> > *To:* Henry Kwan <[email protected]>; Discussion list for users of sipXecs > software <[email protected]> > *Sent:* Thursday, October 11, 2012 9:28:30 AM > *Subject:* Re: [sipx-users] External calls cannot be transferred to voice > mail (sipXecs 4.4.0) > > I don't think the router is compatible with the ability to 1:1 NAT or > do NAT without changing (randomizing) the source port. I would get > thee to a router that will do thusly. Even if you do all of the above, > you will likely have frequent or all the time broken audio. > > On Thu, Oct 11, 2012 at 10:37 AM, Henry Kwan <[email protected]> wrote: > > I am a total newbie on SipXecs. I am also green when it comes to the SIP > > and VoIP PBX scene. Please excuse my seemingly simple question. > > > > The problem that I am encountering, essentially, is that external calls > > cannot be transferred to voice mail when a call is not answered. > Internal > > calls that were not answered were transferred to voice mail without a > > problem. > > > > My setup: > > - SipXecs 4.4.0 installed from the download ISO and updated to the latest > > patches with yum. OS is also updated to Centos 5.8, with the latest > > patches. > > - Phones are Linksys SPA942 only, no other phones are on the system. > Only 3 > > phones are on the system. > > - Domain: mydomain.company.com. company.com is registerd but > > mydomain.company.com is local/internal and the DNS server is the Sipx > PBX. > > - Sipx PBX is the DHCP server, dhcp.conf file is edited to assign a > limited > > range of IP addresses. No other dhcp servers are on the subnet. > > - The workarounds stated on the sipfoundry wiki for the SPA942 are > > implemented, i.e.: > > a. MOH Server: [email protected] > > b. Message Waiting: checked > > c. Mailbox ID: $USER_ID > > d. Voice Mail Server: [email protected]. I have > > also changed mydomain.company.com to the IP address of the sipx server. > > - Use internal sipXbridge to connect to my SIP trunk. SIP trunk > > authenticated successfully and works. > > - Router used is Linksys WRVS4400N. Port 5080 and 30000 to 31000 are > > forwarded to the SipX PBX. > > - Aliases are setup for these 3 phones are set for DID. > > > > With the above setup, I can dial extensions and have their respective > voice > > mail kick-in when a call is not answered. Dial out and DID work as well. > > The problem that I am encountering now is that voice mail does not > kick-in > > when an external call is not answered. Voice mail does work for internal > > calls, though. > > > > I've also added domain aliases of the IP address of the PBX and > > PBX.mydomain.company.com to the setup but that did not help. > > > > I also setup one of the phones to call forward to another phone, then > voice > > mail. The call forwart to another extension worked but call forward to > > voice mail did not. > > > > In desperation, I also added an A record for mydomain.company.com in my > DNS > > server but that did not help. > > > > Lacking the experience of sipX, VoIP PBX, SIP, and network debug tools, I > > hope experienced SipXecs users can shed some on my plight. > > > > Thank you. > > > > Henry Kwan > > > > _______________________________________________ > > sipx-users mailing list > > [email protected] > > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > > > > -- > ~~~~~~~~~~~~~~~~~~ > Tony Graziano, Manager > Telephone: 434.984.8430 > sip: [email protected] > Fax: 434.465.6833 > ~~~~~~~~~~~~~~~~~~ > Linked-In Profile: > http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 > Ask about our Internet Fax services! > ~~~~~~~~~~~~~~~~~~ > > Using or developing for sipXecs from SIPFoundry? Ask me about sipX-CoLab > 2013! > > -- > LAN/Telephony/Security and Control Systems Helpdesk: > Telephone: 434.984.8426 > sip: [email protected] > > Helpdesk Customers: http://myhelp.myitdepartment.net/ > Blog: http://blog.myitdepartment.net/ > > > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > -- Michael Picher, Director of Technical Services eZuce, Inc. 300 Brickstone Square**** Suite 201**** Andover, MA. 01810 O.978-296-1005 X2015 M.207-956-0262 @mpicher <http://twitter.com/mpicher> linkedin <http://www.linkedin.com/profile/view?id=35504760&trk=tab_pro> www.ezuce.com ------------------------------------------------------------------------------------------------------------ There are 10 kinds of people in the world, those who understand binary and those who don't.
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