Hi Todd,
Thank you for your response and your assurance that the combination of SPA942
and SipXecs 4.4 works.
I am just curious regarding the transfer to voice mail since I am not
knowledgeable on the sequence of operation. How is the signalling different
between transfer to voice mail from an internal call and that for an external
call? Is it correct to say that for an internal call to voice mail transfer,
only the phone and the SIP server are involved; for an external call, the ITSP,
SIP server, and phone are involved (therefore the router and ITSP may affect
this operation)? But the call has already been handed to the SIP server, so
why does the ITSP need to get into the scene? If the ITSP is not involved,
what is the difference in handling transfer to voice mail between an internal
and external call?
I apologize for all these questions but I just am mystified by my encounters
and observations.
Thanks and best regards,
Henry Kwan
________________________________
From: Todd Hodgen <thod...@frontier.com>
To: 'Henry Kwan' <hslk...@yahoo.ca>; 'Discussion list for users of sipXecs
software' <sipx-users@list.sipfoundry.org>
Sent: Friday, October 12, 2012 12:39:02 AM
Subject: RE: [sipx-users] External calls cannot be transferred to voice mail
(sipXecs 4.4.0)
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:sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] 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 <tgrazi...@myitdepartment.net>
To: Henry Kwan <hslk...@yahoo.ca>
Cc: Discussion list for users of sipXecs software
<sipx-users@list.sipfoundry.org>
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 <hslk...@yahoo.ca> 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 <tgrazi...@myitdepartment.net>
> To: Henry Kwan <hslk...@yahoo.ca>; Discussion list for users of sipXecs
> software <sipx-users@list.sipfoundry.org>
> 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 <hslk...@yahoo.ca> 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: ~~mh~@mydomain.company.com
>> b. Message Waiting: checked
>> c. Mailbox ID: $USER_ID
>> d. Voice Mail Server: extens...@mydomain.company.com. 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
>> sipx-users@list.sipfoundry.org
>> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>
>
>
> --
> ~~~~~~~~~~~~~~~~~~
> Tony Graziano, Manager
> Telephone: 434.984.8430
> sip: tgrazi...@voice.myitdepartment.net
> 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: helpd...@voice.myitdepartment.net
>
> Helpdesk Customers: http://myhelp.myitdepartment.net/
> Blog: http://blog.myitdepartment.net/
>
>
--
~~~~~~~~~~~~~~~~~~
Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.net
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: helpd...@voice.myitdepartment.net
Helpdesk Customers: http://myhelp.myitdepartment.net/
Blog: http://blog.myitdepartment.net/
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