First of all, allow me to thank everyone who had given me advice on the problem
that I have been encountering: The problem of the inability to have external
calls transferred to voice mail when external calls were not answered, using
version 4.4.0. I am, somewhat, happy that I've resolved the problem. The
resolution has nothing to do with actually identifying the problem but, rather,
avoiding it. Let me explain.
All the advices in suggesting that my router, WRVS4400N or the RV016, may have
been the root of the problem turned out to be wrong. My ITSP was also not the
source of the problem. I downloaded, installed, used pfSense as my
firewall/router and the same problem persisted. After much reading and
searching on the net, I've decided to use 4.2.1 instead of 4.4.0. Viola this
version worked just fine with the same settings/configuration and same
hardware, including the routers. All the routers, WRVS4400N, RV016, and
pfSense, worked.
My obvious conclusion is that one, or more, bug was introduced into 4.4.0 that
caused this behaviour but regression testing of the release did not catch it.
I am, however, surprised that no one else is reporting this behaviour, or bug.
Perhaps someone already did but I simply missed it.
It is all good that this problem is out of my way.
I have another observation that I'd like to seek advice. This observation is
applicable to both 4.2.1 and 4.4.0. I've observed that sometimes after making
changes to configurations and restarted the required processes, as prompted by
sipXecs, I could not make external calls but internal calls and receiving
external calls were just fine. Then I did "Send Profile" to my server to
restart everything but that was also a hit and miss (meaning sometimes the
problem of not able to make external calls went away but sometime not). I then
did "service sipxecs restart" on the command line but that was also a hit and
miss. This problem was also observed even if no configuration changes were
made but simply restarting the sipxecs processes using methods mentioned above
would cause the same observed problem. There were no changes on internal and
external hardware either. So the observed problem had nothing to do with
configuration changes. When this
problem occurred, my phone (SPA942) would show "Calling" then quickly show
"Forbidden".
My off-the-cuff conclusion is that there must be some race conditions or
out-of-order events, for lack of a better term, that sipXecs encountered but
could not consistently resolve or handle properly, leading to this condition.
I may be totally wrong here but I cannot explain why restarting the application
without changing any configuration and hardware will cause this inconsistent
behaviour on the side of the application.
Please excuse my long submission and thank you for your attention.
Best regards,
Henry Kwan
________________________________
From: Richard Bruce <[email protected]>
To: 'Discussion list for users of sipXecs software'
<[email protected]>
Sent: Wednesday, October 17, 2012 6:00:23 AM
Subject: Re: [sipx-users] External calls cannot be transferred to voice mail
(sipXecs 4.4.0)
You should probably check the DNS settings
on the gateway. I have had this problem on multiple analog gateways, having
forgotten to set this.
Richard Bruce
Dimensional Communications
7915 S. Emerson Ave, Suite 131
Indianapolis, IN 46237
(317) 215-4199- office
(317) 946-1899 - cell
________________________________
From:[email protected]
[mailto:[email protected]] On Behalf Of Tony Graziano
Sent: Friday, October 12, 2012
10:31 AM
To: Henry Kwan; Sipx-users list
Subject: Re: [sipx-users] External
calls cannot be transferred to voice mail (sipXecs 4.4.0)
Are you
configuring the spa942 manually? If so, do t do that and let sipx configure it.
Resist the urge to change the configuration for the phone within sipx.
Explain how you are configured (is sipx DNS and dhcp server), etc.
On Oct 12, 2012 10:27 AM, "Henry Kwan" <[email protected]> wrote:
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 <[email protected]>
To: 'Henry Kwan' <[email protected]>; ' Discussion list for users of sipXecs
software ' <[email protected]>
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:[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/
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
sip: [email protected]. net
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/
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