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/
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