Re: [sipx-users] External calls cannot be transferred to voice mail (sipXecs 4.4.0)

2012-10-11 Thread Todd Hodgen 
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 3 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 
To: Henry Kwan  
Cc: Discussion list for users of sipXecs software
 
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  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 
> To: Henry Kwan ; Discussion list for users of sipXecs
> software 
> 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  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

Re: [sipx-users] Hacked SipXecs 4.4

2012-10-11 Thread Davide Poletto 
Hi, could be something related to Polycom's phones FTP provisioning ? I've
read that the default FTP user name for that is 'PlcmSpIp' and the default
password is the same (so well-known credentials).

Over ther internet there are some references about that (AFAIK see
this 
one,
just as example, that has a good explanation about logged messages).

Regards, Davide.


On Fri, Oct 12, 2012 at 5:48 AM, Noah Mehl  wrote:

> All,
>
> I just realized that my emails from my SipXecs 4.4 server were not being
> delivered.  Upon further investigation, I found that my SipXecs VM had a
> sendmail queue with over 13000 messages in it.  I'm trying to figure out
> how my machine was sending mail, and it doesn't look like the relay is
> open, but I found something curious:
>
> [root@sipx1 log]# cat secure | grep "pam_unix(sshd:session): session
> opened"
> Oct 11 06:09:25 sipx1 sshd[22059]: pam_unix(sshd:session): session opened
> for user PlcmSpIp by (uid=0)
> Oct 11 18:36:02 sipx1 sshd[29185]: pam_unix(sshd:session): session opened
> for user PlcmSpIp by (uid=0)
> Oct 11 18:36:16 sipx1 sshd[29192]: pam_unix(sshd:session): session opened
> for user PlcmSpIp by (uid=0)
> Oct 11 18:36:21 sipx1 sshd[29195]: pam_unix(sshd:session): session opened
> for user PlcmSpIp by (uid=0)
> Oct 11 20:57:58 sipx1 sshd[30561]: pam_unix(sshd:session): session opened
> for user PlcmSpIp by (uid=0)
>
> Those are what I think to be successful ssh logins with the user PlcmSplp.
>  Is this user part of the SipXecs install?
>
> ~Noah
>
> Scanned for viruses and content by the Tranet Spam Sentinel service.
> ___
> sipx-users mailing list
> sipx-users@list.sipfoundry.org
> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>
___
sipx-users mailing list
sipx-users@list.sipfoundry.org
List Archive: http://list.sipfoundry.org/archive/sipx-users/

[sipx-users] Hacked SipXecs 4.4

2012-10-11 Thread Noah Mehl 
All,

I just realized that my emails from my SipXecs 4.4 server were not being 
delivered.  Upon further investigation, I found that my SipXecs VM had a 
sendmail queue with over 13000 messages in it.  I'm trying to figure out how my 
machine was sending mail, and it doesn't look like the relay is open, but I 
found something curious:

[root@sipx1 log]# cat secure | grep "pam_unix(sshd:session): session opened"
Oct 11 06:09:25 sipx1 sshd[22059]: pam_unix(sshd:session): session opened for 
user PlcmSpIp by (uid=0)
Oct 11 18:36:02 sipx1 sshd[29185]: pam_unix(sshd:session): session opened for 
user PlcmSpIp by (uid=0)
Oct 11 18:36:16 sipx1 sshd[29192]: pam_unix(sshd:session): session opened for 
user PlcmSpIp by (uid=0)
Oct 11 18:36:21 sipx1 sshd[29195]: pam_unix(sshd:session): session opened for 
user PlcmSpIp by (uid=0)
Oct 11 20:57:58 sipx1 sshd[30561]: pam_unix(sshd:session): session opened for 
user PlcmSpIp by (uid=0)

Those are what I think to be successful ssh logins with the user PlcmSplp.  Is 
this user part of the SipXecs install?

~Noah

Scanned for viruses and content by the Tranet Spam Sentinel service.
___
sipx-users mailing list
sipx-users@list.sipfoundry.org
List Archive: http://list.sipfoundry.org/archive/sipx-users/

Re: [sipx-users] External calls cannot be transferred to voice mail (sipXecs 4.4.0)

2012-10-11 Thread Henry Kwan 
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 3 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 
To: Henry Kwan  
Cc: Discussion list for users of sipXecs software 
 
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  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 
> To: Henry Kwan ; Discussion list for users of sipXecs
> software 
> 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  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 3 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 experie

Re: [sipx-users] sipxecs-4.6.0: Framebuffer (or something) broken on first boot

2012-10-11 Thread Adrien Guillon 
Okay, so I tried to CentOS 6.3 minimal install.  The issue remains with a
normal install, however CentOS also has a low-end video mode, and that
install works just fine.

It might be worth considering, if simple enough, to add an option to SipX
to install CentOS in a low video mode.  For now I can adjust GRUB, but
something to consider for future releases.

AJ

On Thu, Oct 11, 2012 at 3:16 PM, Adrien Guillon wrote:

> I'll download a minimal CentOS install now to try to help narrow down the
> problem to CentOS or SipX.  Thanks for pointing out the potential install
> parameters, if X runs during the install (which it did) I would assume that
> the framebuffer would be alright.
>
> Tony: I am well aware of how to change parameters and grub, and do not
> need your help using Google.  I am attempting to help a project isolate an
> issue that is affecting me, and potentially others.  This is me
> volunteering to help a project I like in some small way, because I have a
> problem in front of me that I can help debug.
>
> Thanks.
>
> AJ
>
>
> On Thu, Oct 11, 2012 at 1:34 PM, Tony Graziano <
> tgrazi...@myitdepartment.net> wrote:
>
>> It's a centos issue.
>>
>> It's probably a resolution issue. it's "still centos and your hardware"
>>
>> let me google that FOR you...
>> http://lmgtfy.com/?q=framebuffers+centos+6
>>
>> the answer lies in "probably" changing grub.conf to match what the
>> hardware might support...
>>
>> example:
>> kernel /boot/vmlinuz-2.6.18-164.11.1.el5.centos.plus ro root=LABEL=/
>> vga=0x305
>>
>> Value of VGA :
>>
>> 0x307=1280x1024
>> 0x305=1024x768
>> 0x303=800x600
>> 0x301=640x480
>>
>> Something is probably calling hires mode (In centos, not linux).
>>
>> Either way, it's a linux/centos hardware thing. I've never had this
>> issue but hardware remains pretty current all the time.
>>
>> Good luck.
>>
>> On Thu, Oct 11, 2012 at 11:56 AM, Adrien Guillon 
>> wrote:
>> > It's an ISO install, 4.6.
>> >
>> > Yes, it could be an issue with my hardware and CentOS, however the
>> issue is
>> > that not all hardware is going to support framebuffers properly.
>>  Regardless
>> > of whether the bug is in CentOS or the install scripts of the sipxecs
>> ISO,
>> > it is still an issue that there is no option to disable the framebuffer
>> if
>> > it doesn't work.
>> >
>> > AJ
>> >
>> >
>> > On Thu, Oct 11, 2012 at 11:38 AM, Tony Graziano
>> >  wrote:
>> >>
>> >> Realize you have not stated what version of sip and how you are
>> >> installing it. Is it sipx 4.4 or 4.6? Are you installing via RPM or
>> >> from ISO?
>> >>
>> >> Typically it means your hardware has an issue with linux. If you know
>> >> what hardware you are using (we don't) and what version of sipx you
>> >> are installing (4.4 using centos 5.x and 4.6 uses centos 6.x), try
>> >> googling it. Even if it is garbled on your display, you have already
>> >> set the password and IP during the setup. This probably means you can
>> >> ssh to it and access sipxconfig. The video driver (assumed) is
>> >> something you can deal with separately I would think.
>> >>
>> >> On Thu, Oct 11, 2012 at 11:35 AM, Adrien Guillon > >
>> >> wrote:
>> >> > Hi,
>> >> >
>> >> > I'm not sure if this is a known bug.  I've installed SipX twice now,
>> and
>> >> > upon booting the screen is completely garbled.  I suspect a
>> framebuffer
>> >> > or
>> >> > something is being used for boot, but it doesn't like my hardware.
>> >> > Anyone
>> >> > else experience this?
>> >> >
>> >> > AJ
>> >> >
>> >> > ___
>> >> > 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
>> >>
>> >> ___
>> >> sipx-users mailing list
>> >> sipx-users@list.sipfoundry.org
>> >> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>> >
>> >
>> >
>> > ___
>> > 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:
>> htt

Re: [sipx-users] sipxecs-4.6.0: Framebuffer (or something) broken on first boot

2012-10-11 Thread Adrien Guillon 
I'll download a minimal CentOS install now to try to help narrow down the
problem to CentOS or SipX.  Thanks for pointing out the potential install
parameters, if X runs during the install (which it did) I would assume that
the framebuffer would be alright.

Tony: I am well aware of how to change parameters and grub, and do not need
your help using Google.  I am attempting to help a project isolate an issue
that is affecting me, and potentially others.  This is me volunteering to
help a project I like in some small way, because I have a problem in front
of me that I can help debug.

Thanks.

AJ

On Thu, Oct 11, 2012 at 1:34 PM, Tony Graziano  wrote:

> It's a centos issue.
>
> It's probably a resolution issue. it's "still centos and your hardware"
>
> let me google that FOR you...
> http://lmgtfy.com/?q=framebuffers+centos+6
>
> the answer lies in "probably" changing grub.conf to match what the
> hardware might support...
>
> example:
> kernel /boot/vmlinuz-2.6.18-164.11.1.el5.centos.plus ro root=LABEL=/
> vga=0x305
>
> Value of VGA :
>
> 0x307=1280x1024
> 0x305=1024x768
> 0x303=800x600
> 0x301=640x480
>
> Something is probably calling hires mode (In centos, not linux).
>
> Either way, it's a linux/centos hardware thing. I've never had this
> issue but hardware remains pretty current all the time.
>
> Good luck.
>
> On Thu, Oct 11, 2012 at 11:56 AM, Adrien Guillon 
> wrote:
> > It's an ISO install, 4.6.
> >
> > Yes, it could be an issue with my hardware and CentOS, however the issue
> is
> > that not all hardware is going to support framebuffers properly.
>  Regardless
> > of whether the bug is in CentOS or the install scripts of the sipxecs
> ISO,
> > it is still an issue that there is no option to disable the framebuffer
> if
> > it doesn't work.
> >
> > AJ
> >
> >
> > On Thu, Oct 11, 2012 at 11:38 AM, Tony Graziano
> >  wrote:
> >>
> >> Realize you have not stated what version of sip and how you are
> >> installing it. Is it sipx 4.4 or 4.6? Are you installing via RPM or
> >> from ISO?
> >>
> >> Typically it means your hardware has an issue with linux. If you know
> >> what hardware you are using (we don't) and what version of sipx you
> >> are installing (4.4 using centos 5.x and 4.6 uses centos 6.x), try
> >> googling it. Even if it is garbled on your display, you have already
> >> set the password and IP during the setup. This probably means you can
> >> ssh to it and access sipxconfig. The video driver (assumed) is
> >> something you can deal with separately I would think.
> >>
> >> On Thu, Oct 11, 2012 at 11:35 AM, Adrien Guillon 
> >> wrote:
> >> > Hi,
> >> >
> >> > I'm not sure if this is a known bug.  I've installed SipX twice now,
> and
> >> > upon booting the screen is completely garbled.  I suspect a
> framebuffer
> >> > or
> >> > something is being used for boot, but it doesn't like my hardware.
> >> > Anyone
> >> > else experience this?
> >> >
> >> > AJ
> >> >
> >> > ___
> >> > 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
> >>
> >> ___
> >> sipx-users mailing list
> >> sipx-users@list.sipfoundry.org
> >> List Archive: http://list.sipfoundry.org/archive/sipx-users/
> >
> >
> >
> > ___
> > 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
> ___
> sipx-users mailing list
> sipx-users@list.sipfoundry.org
> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>

Re: [sipx-users] External calls cannot be transferred to voice mail (sipXecs 4.4.0)

2012-10-11 Thread Tony Graziano 
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  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 
> To: Henry Kwan ; Discussion list for users of sipXecs
> software 
> 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  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 3 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

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Blog: http://blog.myitdepartment.net
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List Ar

Re: [sipx-users] sipxecs-4.6.0: Framebuffer (or something) broken on first boot

2012-10-11 Thread Tony Graziano 
It's a centos issue.

It's probably a resolution issue. it's "still centos and your hardware"

let me google that FOR you...
http://lmgtfy.com/?q=framebuffers+centos+6

the answer lies in "probably" changing grub.conf to match what the
hardware might support...

example:
kernel /boot/vmlinuz-2.6.18-164.11.1.el5.centos.plus ro root=LABEL=/ vga=0x305

Value of VGA :

0x307=1280x1024
0x305=1024x768
0x303=800x600
0x301=640x480

Something is probably calling hires mode (In centos, not linux).

Either way, it's a linux/centos hardware thing. I've never had this
issue but hardware remains pretty current all the time.

Good luck.

On Thu, Oct 11, 2012 at 11:56 AM, Adrien Guillon  wrote:
> It's an ISO install, 4.6.
>
> Yes, it could be an issue with my hardware and CentOS, however the issue is
> that not all hardware is going to support framebuffers properly.  Regardless
> of whether the bug is in CentOS or the install scripts of the sipxecs ISO,
> it is still an issue that there is no option to disable the framebuffer if
> it doesn't work.
>
> AJ
>
>
> On Thu, Oct 11, 2012 at 11:38 AM, Tony Graziano
>  wrote:
>>
>> Realize you have not stated what version of sip and how you are
>> installing it. Is it sipx 4.4 or 4.6? Are you installing via RPM or
>> from ISO?
>>
>> Typically it means your hardware has an issue with linux. If you know
>> what hardware you are using (we don't) and what version of sipx you
>> are installing (4.4 using centos 5.x and 4.6 uses centos 6.x), try
>> googling it. Even if it is garbled on your display, you have already
>> set the password and IP during the setup. This probably means you can
>> ssh to it and access sipxconfig. The video driver (assumed) is
>> something you can deal with separately I would think.
>>
>> On Thu, Oct 11, 2012 at 11:35 AM, Adrien Guillon 
>> wrote:
>> > Hi,
>> >
>> > I'm not sure if this is a known bug.  I've installed SipX twice now, and
>> > upon booting the screen is completely garbled.  I suspect a framebuffer
>> > or
>> > something is being used for boot, but it doesn't like my hardware.
>> > Anyone
>> > else experience this?
>> >
>> > AJ
>> >
>> > ___
>> > 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
>>
>> ___
>> sipx-users mailing list
>> sipx-users@list.sipfoundry.org
>> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>
>
>
> ___
> 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
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Re: [sipx-users] External calls cannot be transferred to voice mail (sipXecs 4.4.0)

2012-10-11 Thread Douglas Hubler 
On Thu, Oct 11, 2012 at 12:18 PM, Henry Kwan  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.

Disable "SIP algorithm" if it's enabled router.
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Re: [sipx-users] sipxecs-4.6.0: Framebuffer (or something) broken on first boot

2012-10-11 Thread Douglas Hubler 
On Thu, Oct 11, 2012 at 11:56 AM, Adrien Guillon  wrote:
> It's an ISO install, 4.6.
>
> Yes, it could be an issue with my hardware and CentOS, however the issue is
> that not all hardware is going to support framebuffers properly.  Regardless
> of whether the bug is in CentOS or the install scripts of the sipxecs ISO,
> it is still an issue that there is no option to disable the framebuffer if
> it doesn't work.

Try installing CentOS 6 minimal ISO on the machine just to give us an
idea if it's centos or sipxecs.  Either sipxecs or centos, you can hit
tab in the boot menu and change the installation parameters.  All sort
of wacky options

 
http://www.linuxtopia.org/online_books/rhel6/rhel_6_installation/rhel_6_installation_ap-admin-options.html

Assuming you get that far.  It's not clear exactly where "upon boot"
is that you're describing.
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Re: [sipx-users] Snom 370 Presence Monitoring

2012-10-11 Thread Michael Picher 
I'll add one more thing that works...

On the Windows version of Bria 3.x (not the mac, linux, or any of the
tablet/iphone/android) there is the ability to setup RLS.  You can enter
your RLS subscription and allow other phones to monitor your softphone's
presence.  For a workgroup server enter:  sip:~~rl~C~@sipdomain where
 is your extension and sipdomain is your sip domain.

And check the box (allow others to see your presence).

This also works with all versions of the Voice Operator Panel software.

Your XMPP users will also then be able to see your presence as RLS and XMPP
are synced by sipXecs/openUC.

Thanks,
  Mike

On Wed, Oct 10, 2012 at 3:54 PM, Geoff Musgrave <
geoff.musgr...@cacionline.net> wrote:

> Thanks Tony. I got too hung up on it being a snom/softphone compatibility
> issue to step back and look and the rest of the situation.
>
> Thank you for the second set of eyes!
>
> -Original Message-
> From: sipx-users-boun...@list.sipfoundry.org [mailto:
> sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
> Sent: Wednesday, October 10, 2012 2:29 PM
> To: Discussion list for users of sipXecs software
> Subject: Re: [sipx-users] Snom 370 Presence Monitoring
>
> You need at least Bria 3.x. None of the other phones support the methods
> needed. The phone presence and xmpp need to be integrated into the same
> software package (bria 3.x), the polycom phone can only be monitored from a
> softphone if the polycom user is also running a proper version of xmpp.
>
>
> See the wiki please:
>
> http://wiki.sipfoundry.org/pages/viewpage.action?pageId=6520862
>
>
> On Wed, Oct 10, 2012 at 3:18 PM, Geoff Musgrave <
> geoff.musgr...@cacionline.net> wrote:
> > I have 2 snom 370 phones with expansion bases connected to sipXecs 4.6
> > working properly. They will monitor and display the presence of users
> > that have Polycom phones however none of the softphone users can be
> > monitored from the snom phones.
> >
> >
> >
> > I'ved tried several different softphones but no luck. Xlite 3 & 5,
> > zoiper classic/communicator, bria 2.4, 3cx, jitsi, VOP
> >
> >
> >
> > Can anyone give any suggestions as to a softphone that may work with
> > the Snom 370 presence monitoring or another suggestion that I may be
> > overlooking.
> >
> >
> >
> > Thanks in advance for any and all help.
> >
> >
> >
> >
> >
> >
> >
> >
> > ___
> > 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
> ___
> sipx-users mailing list
> sipx-users@list.sipfoundry.org
> List Archive: http://list.sipfoundry.org/archive/sipx-users/
> ___
> sipx-users mailing list
> sipx-users@list.sipfoundry.org
> 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 
linkedin 
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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Re: [sipx-users] External calls cannot be transferred to voice mail (sipXecs 4.4.0)

2012-10-11 Thread Michael Picher 
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  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 
> *To:* Henry Kwan ; Discussion list for users of sipXecs
> software 
> *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  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 3 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/
>
>
>
> ___
> sipx-users mailing list
> sipx-users@list.sipfoundry.org
> 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 
linkedin 
www.ezuce.com

---

Re: [sipx-users] External calls cannot be transferred to voice mail (sipXecs 4.4.0)

2012-10-11 Thread Henry Kwan 
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 
To: Henry Kwan ; Discussion list for users of sipXecs 
software  
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  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 3 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/___
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Re: [sipx-users] sipxecs-4.6.0: Framebuffer (or something) broken on first boot

2012-10-11 Thread Michael Picher 
Well, you could use the RPM installation method and go that route...  Then
you'd have full control over the installation.

Thanks,
   Mike

On Thu, Oct 11, 2012 at 11:56 AM, Adrien Guillon wrote:

> It's an ISO install, 4.6.
>
> Yes, it could be an issue with my hardware and CentOS, however the issue
> is that not all hardware is going to support framebuffers properly.
> Regardless of whether the bug is in CentOS or the install scripts of the
> sipxecs ISO, it is still an issue that there is no option to disable the
> framebuffer if it doesn't work.
>
> AJ
>
>
>
> On Thu, Oct 11, 2012 at 11:38 AM, Tony Graziano <
> tgrazi...@myitdepartment.net> wrote:
>
>> Realize you have not stated what version of sip and how you are
>> installing it. Is it sipx 4.4 or 4.6? Are you installing via RPM or
>> from ISO?
>>
>> Typically it means your hardware has an issue with linux. If you know
>> what hardware you are using (we don't) and what version of sipx you
>> are installing (4.4 using centos 5.x and 4.6 uses centos 6.x), try
>> googling it. Even if it is garbled on your display, you have already
>> set the password and IP during the setup. This probably means you can
>> ssh to it and access sipxconfig. The video driver (assumed) is
>> something you can deal with separately I would think.
>>
>> On Thu, Oct 11, 2012 at 11:35 AM, Adrien Guillon 
>> wrote:
>> > Hi,
>> >
>> > I'm not sure if this is a known bug.  I've installed SipX twice now, and
>> > upon booting the screen is completely garbled.  I suspect a framebuffer
>> or
>> > something is being used for boot, but it doesn't like my hardware.
>>  Anyone
>> > else experience this?
>> >
>> > AJ
>> >
>> > ___
>> > 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
>> ___
>> sipx-users mailing list
>> sipx-users@list.sipfoundry.org
>> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>>
>
>
> ___
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> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>



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300 Brickstone Square

Suite 201

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O.978-296-1005 X2015
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Re: [sipx-users] sipxecs-4.6.0: Framebuffer (or something) broken on first boot

2012-10-11 Thread Adrien Guillon 
It's an ISO install, 4.6.

Yes, it could be an issue with my hardware and CentOS, however the issue is
that not all hardware is going to support framebuffers properly.
Regardless of whether the bug is in CentOS or the install scripts of the
sipxecs ISO, it is still an issue that there is no option to disable the
framebuffer if it doesn't work.

AJ


On Thu, Oct 11, 2012 at 11:38 AM, Tony Graziano <
tgrazi...@myitdepartment.net> wrote:

> Realize you have not stated what version of sip and how you are
> installing it. Is it sipx 4.4 or 4.6? Are you installing via RPM or
> from ISO?
>
> Typically it means your hardware has an issue with linux. If you know
> what hardware you are using (we don't) and what version of sipx you
> are installing (4.4 using centos 5.x and 4.6 uses centos 6.x), try
> googling it. Even if it is garbled on your display, you have already
> set the password and IP during the setup. This probably means you can
> ssh to it and access sipxconfig. The video driver (assumed) is
> something you can deal with separately I would think.
>
> On Thu, Oct 11, 2012 at 11:35 AM, Adrien Guillon 
> wrote:
> > Hi,
> >
> > I'm not sure if this is a known bug.  I've installed SipX twice now, and
> > upon booting the screen is completely garbled.  I suspect a framebuffer
> or
> > something is being used for boot, but it doesn't like my hardware.
>  Anyone
> > else experience this?
> >
> > AJ
> >
> > ___
> > 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
> ___
> sipx-users mailing list
> sipx-users@list.sipfoundry.org
> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>
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Re: [sipx-users] sipxecs-4.6.0: Framebuffer (or something) broken on first boot

2012-10-11 Thread Tony Graziano 
Realize you have not stated what version of sip and how you are
installing it. Is it sipx 4.4 or 4.6? Are you installing via RPM or
from ISO?

Typically it means your hardware has an issue with linux. If you know
what hardware you are using (we don't) and what version of sipx you
are installing (4.4 using centos 5.x and 4.6 uses centos 6.x), try
googling it. Even if it is garbled on your display, you have already
set the password and IP during the setup. This probably means you can
ssh to it and access sipxconfig. The video driver (assumed) is
something you can deal with separately I would think.

On Thu, Oct 11, 2012 at 11:35 AM, Adrien Guillon  wrote:
> Hi,
>
> I'm not sure if this is a known bug.  I've installed SipX twice now, and
> upon booting the screen is completely garbled.  I suspect a framebuffer or
> something is being used for boot, but it doesn't like my hardware.  Anyone
> else experience this?
>
> AJ
>
> ___
> 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
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[sipx-users] sipxecs-4.6.0: Framebuffer (or something) broken on first boot

2012-10-11 Thread Adrien Guillon 
Hi,

I'm not sure if this is a known bug.  I've installed SipX twice now, and
upon booting the screen is completely garbled.  I suspect a framebuffer or
something is being used for boot, but it doesn't like my hardware.  Anyone
else experience this?

AJ
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Re: [sipx-users] External calls cannot be transferred to voice mail (sipXecs 4.4.0)

2012-10-11 Thread Tony Graziano 
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  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 3 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
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Re: [sipx-users] External calls cannot be transferred to voice mail (sipXecs 4.4.0)

2012-10-11 Thread Gerald Drouillard 

On 10/11/2012 10:37 AM, Henry Kwan 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 3 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.


External calls not transferring usually have 2 causes: your ITSP does 
not support it, the call did not come in on 5080/registration, or a 
firewall issue.

Who is your ITSP?
Did you try to forward 5060 udp/tcp also?
Is your ITSP sending the calls to your based on your registration is it 
IP based.  If IP the call has to come in on 5080 to be able to transfer.

Did you do a "yum update"?
Send profiles to the server: System|Servers
Reboot

--
Regards
--
Gerald Drouillard
Technology Architect
Drouillard & Associates, Inc.
http://www.Drouillard.biz

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[sipx-users] External calls cannot be transferred to voice mail (sipXecs 4.4.0)

2012-10-11 Thread Henry Kwan 
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 3 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___
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Re: [sipx-users] update to latest 4.6 - send profiles required

2012-10-11 Thread Douglas Hubler 
On Thu, Oct 11, 2012 at 5:16 AM, George Niculae  wrote:
> On Thu, Oct 11, 2012 at 12:05 PM, Douglas Hubler  wrote:
>> On Thu, Oct 11, 2012 at 3:58 AM, George Niculae  wrote:
>>> please send profiles to server after updating to latest 4.6 RPMs -
>>> there was a change in conferences records so they need to be
>>> regenerated
>>
>> George,
>> When sipxconfig restarts, it regenerates profiles but does not rebuild
>> mongo tables. just checking, is the fix in regenerating mongo tables?
>
> Nope, that was not included yet. We should add the mechanism to
> regenerate mongo tables on sipxconfig restart only when new patch that
> needs this.

Ok, so if i understand you correctly :  your fix do require
regenerating mongo data therefore you ask folks to send server
profiles.  That's fine, I was just making sure you (and other
developers) understood that if any code change only requires that
config files are regenerated, then there's nothing you have to do.
restarting sipxconfig regenerates all config files.

George and for other's sake, we could automate regenerating mongo data
through SetupListener interface, but then the code to check if this
has been done will live in codebase forever. We'll reserve that sort
of option for after 4.6.0 is released.
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Re: [sipx-users] update to latest 4.6 - send profiles required

2012-10-11 Thread Douglas Hubler 
On Thu, Oct 11, 2012 at 5:16 AM, George Niculae  wrote:
> On Thu, Oct 11, 2012 at 12:05 PM, Douglas Hubler  wrote:
>> On Thu, Oct 11, 2012 at 3:58 AM, George Niculae  wrote:
>>> please send profiles to server after updating to latest 4.6 RPMs -
>>> there was a change in conferences records so they need to be
>>> regenerated
>>
>> George,
>> When sipxconfig restarts, it regenerates profiles but does not rebuild
>> mongo tables. just checking, is the fix in regenerating mongo tables?
>
> Nope, that was not included yet. We should add the mechanism to
> regenerate mongo tables on sipxconfig restart only when new patch that
> needs this.
>
>> if not, are you sure this is nec?
>
> Yes, send profiles or resave conference from sipXconfig UI
>
> George
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Re: [sipx-users] update to latest 4.6 - send profiles required

2012-10-11 Thread George Niculae 
On Thu, Oct 11, 2012 at 12:05 PM, Douglas Hubler  wrote:
> On Thu, Oct 11, 2012 at 3:58 AM, George Niculae  wrote:
>> please send profiles to server after updating to latest 4.6 RPMs -
>> there was a change in conferences records so they need to be
>> regenerated
>
> George,
> When sipxconfig restarts, it regenerates profiles but does not rebuild
> mongo tables. just checking, is the fix in regenerating mongo tables?

Nope, that was not included yet. We should add the mechanism to
regenerate mongo tables on sipxconfig restart only when new patch that
needs this.

> if not, are you sure this is nec?

Yes, send profiles or resave conference from sipXconfig UI

George
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Re: [sipx-users] update to latest 4.6 - send profiles required

2012-10-11 Thread Douglas Hubler 
On Thu, Oct 11, 2012 at 3:58 AM, George Niculae  wrote:
> please send profiles to server after updating to latest 4.6 RPMs -
> there was a change in conferences records so they need to be
> regenerated

George,
When sipxconfig restarts, it regenerates profiles but does not rebuild
mongo tables. just checking, is the fix in regenerating mongo tables?
if not, are you sure this is nec?
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[sipx-users] update to latest 4.6 - send profiles required

2012-10-11 Thread George Niculae 
Hi All,

please send profiles to server after updating to latest 4.6 RPMs -
there was a change in conferences records so they need to be
regenerated

Thanks
George
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