Thank you to all who have given me suggestions. I'll follow-up on those
suggestions.
Best regards,
Henry Kwan
________________________________
From: Michael Picher <[email protected]>
To: Discussion list for users of sipXecs software
<[email protected]>
Cc: Henry Kwan <[email protected]>
Sent: Friday, October 12, 2012 4:42:25 AM
Subject: Re: [sipx-users] External calls cannot be transferred to voice mail
(sipXecs 4.4.0)
And beware of those Cisco RV series 'firewalls'. In the past with 1-to-1 NAT
I've noted that they actually just open up ALL ports. Scary as hell...
Run away... run towards pfSense.
Mike
On Fri, Oct 12, 2012 at 2:39 AM, Todd Hodgen <[email protected]> wrote:
Henry, I can’t speak to the router, or your ITSP provider. I can state that
I have a site running on 4.4 with a single server, server provides DHCP and
DNS, and works with SPA942 phones. I did not use the wiki recommendations. I
simply provisioned them via the management templates and they work perfectly.
>
>Trunks are provided via a PRI gateway – I’ve used Epygi and Patton gateways at
>this site with great results from both of them.
>
>I would suggest router or ITSP are your issue, as others have.
>
>VOIP.ms is a low cost ITSP provider that for a minimum investment you can use
>to test. We know they work, and for a few bucks you can save yourself some
>time in troubleshooting.
>
>From:[email protected]
>[mailto:[email protected]] On Behalf Of Henry Kwan
>Sent: Thursday, October 11, 2012 8:24 PM
>To: Tony Graziano
>
>Cc: Discussion list for users of sipXecs software
>
>Subject: Re: [sipx-users] External calls cannot be transferred to voice mail
>(sipXecs 4.4.0)
>
>The router, Linksys WRVS4400N, that I am using is not a home router. It is a
>small business router. Having said that it still may not mean it is a
>suitable router for SipX.
>
>I managed to obtain another router and do more testing tonight. The router is
>a Linksys/Cisco RV016. It has one-to-one NAT and I set it up to have a
>one-to-one NAT entry between my internal sipx server and the router's external
>interface.
>
>Using the RV016, the following test results were obtained (please note that I
>had to port forward 5080, and 30000 to 31000, otherwise external calls would
>come through with just one-to-one NAT setup and enabled):
>
>All the previous test results remained exactly the same. That is to say
>internal calls could be transferred to voice mail when no one answer the calls
>but external calls could not.
>
>I then setup forwarding directly to voice mail by calling the external voice
>mail DID number that I setup. That worked!!
>
>I am beginning to think that it may have to do with how the SPA942 operates or
>it was not setup properly via the sipxecs web interface. But I am not
>knowledgeable enough to examine and change the settings on the SPA942.
>
>If anyone can give me suggestions to troubleshoot this problem, I'd much
>appreciate it.
>
>Best regards,
>
>Henry Kwan
>
>________________________________
>
>From:Tony Graziano <[email protected]>
>To: Henry Kwan <[email protected]>
>Cc: Discussion list for users of sipXecs software
><[email protected]>
>Sent: Thursday, October 11, 2012 11:35:38 AM
>Subject: Re: [sipx-users] External calls cannot be transferred to voice mail
>(sipXecs 4.4.0)
>
>Tested by who? Just because it works as a home router for voip doesn't
>mean it will probably work for your office hosting a PBX, BIG FAT
>difference.
>
>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/
>>
>>
>
>
>
>--
>~~~~~~~~~~~~~~~~~~
>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
www.ezuce.com
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