Re: [sipx-users] Bug fix release update: sipXecs 4.4.0 update #21

[Michael Picher]

we're pushing openuc today for you guys...  dave will contact when ready.

thanks,
  mike

On Fri, Oct 12, 2012 at 12:51 PM, Douglas Hubler  wrote:

> Update #21
> ==
> - ** No security updates in this update **
> - ISO has *not* been rebuilt as decided in release policy. Yum update
> after installation is recommended for getting these updates.
> - Thank you all for your continued testing and fixes.
>
> Build Log
> ==
> commit 9f44bd8eb22621ba48187dba989f1da7b61a0d4f
> Author: Joegen Baclor 
> Date:   Fri Oct 5 19:28:20 2012 +0800
>
> If request-uri is towards an alias domain and a user alias at the
> same time, normalize the request\
> -uri and do not add a contact.  If user is registered, regdb
> redirector will insert the final contact.
>
> commit bb6acf0e9eb997ba0c35701b6b84e848312419cc
> Author: George Niculae 
> Date:   Mon Oct 8 23:10:25 2012 +0300
>
> Revert "Revert "XX-10428 - Alias Redirector must be evaluate prior
> to regDB redirector so that the\
>  correct identity is used to search.""
>
> This reverts commit aaed0e5959b86ef1fbd4b3a49a33d1c62bb950d9.
>
> commit db4ea18bf23e28c9ed4b2c646473ce52b5d2f525
> Author: George Niculae 
> Date:   Mon Oct 8 23:10:01 2012 +0300
>
> Revert "Revert "XX-10428 - Alias Redirector does not match rule if
> domain is also a domain alias""
>
> This reverts commit 4152fa28c983974ea855acc20b3c5e0b4494d769.
>
>
> for past releases see
> http://download.sipfoundry.org/pub/sipXecs/ChangeLog-4.4.0
> ___
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> sipx-users@list.sipfoundry.org
> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>



-- 
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eZuce, Inc.

300 Brickstone Square

Suite 201

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[sipx-users] Bug fix release update: sipXecs 4.4.0 update #21

[Douglas Hubler]

Update #21
==
- ** No security updates in this update **
- ISO has *not* been rebuilt as decided in release policy. Yum update
after installation is recommended for getting these updates.
- Thank you all for your continued testing and fixes.

Build Log
==
commit 9f44bd8eb22621ba48187dba989f1da7b61a0d4f
Author: Joegen Baclor 
Date:   Fri Oct 5 19:28:20 2012 +0800

If request-uri is towards an alias domain and a user alias at the
same time, normalize the request\
-uri and do not add a contact.  If user is registered, regdb
redirector will insert the final contact.

commit bb6acf0e9eb997ba0c35701b6b84e848312419cc
Author: George Niculae 
Date:   Mon Oct 8 23:10:25 2012 +0300

Revert "Revert "XX-10428 - Alias Redirector must be evaluate prior
to regDB redirector so that the\
 correct identity is used to search.""

This reverts commit aaed0e5959b86ef1fbd4b3a49a33d1c62bb950d9.

commit db4ea18bf23e28c9ed4b2c646473ce52b5d2f525
Author: George Niculae 
Date:   Mon Oct 8 23:10:01 2012 +0300

Revert "Revert "XX-10428 - Alias Redirector does not match rule if
domain is also a domain alias""

This reverts commit 4152fa28c983974ea855acc20b3c5e0b4494d769.


for past releases see http://download.sipfoundry.org/pub/sipXecs/ChangeLog-4.4.0
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Re: [sipx-users] How to do transfer on unmanaged GW

[Tony Graziano]

Or your plugin will have to reference a db.

It reminds me of a couple of itsp's I struggled with in mexico and
Australia. Callcentric certainly comes to mind.
On Oct 12, 2012 11:25 AM, "Sven Evensen"  wrote:

> Thanks Tony, you confirmed what I was afraid of.
>
> And Yes, we are very close to saying we just cannot use this ITSP.
>
> Our last hope is that we have a plugin where we possibly can intercept the
> refer and replace the internal number with the full DID.
> Of course some internal numbers might not have a full DID, but we will
> have to live with that.
>
>
> On Fri, Oct 12, 2012 at 4:15 PM, Tony Graziano <
> tgrazi...@myitdepartment.net> wrote:
>
>> I do not blame you for not wanting to do it that way.
>>
>> It would be better to find an ITSP who supports REFER.
>>
>> If you use sipxbridge and set it up as a trunk it (sipxbridge) will
>> handle the refer locally. It would need the ITSP to support re-invite.
>> If you set it up as an unmanaged gateway then the proxy will send the
>> REFER to the ITSP (or their gateway, etc.), but it sounds like this
>> ITSP does not support reinvite AND requires the "Route by To Header".
>> Smacks to me of a best case of what not to look for.
>>
>> There is a way you can route by the to header in sipxbridge as a
>> siptrunk. There is not a way that I know of to do this as an unmanaged
>> gateway. I'm not sure how hard it would be to add that feature or what
>> else would break or become problematic with users who have lots of
>> unmanaged gateways out there since it also assumes the gateway
>> supports refer and looks at the INVITE and not the TO header.
>>
>>
>> On Fri, Oct 12, 2012 at 10:46 AM, Sven Evensen 
>> wrote:
>> > A customer is using an ITSP which does not support re-invite, therefore
>> we
>> > set it up an unmanaged gateway and REFER is used when doing a transfer.
>> If
>> > an external call comes through SIP trunk to an AA 200 and the AA does a
>> > transfer to say 211, the refer back to the SIP trunk will have refer-to
>> as
>> > 211 and that is not a number that the SIP trunk knows, it only knows the
>> > full DIDs assigned to that trunk.
>> >
>> > If I program the AA with the full DID of ext 211, then it works. Be we
>> do
>> > not want internal users to transfer to internal users with full DID.
>> >
>> > So is there any way we can do externally originated transfer?
>> >
>> > Thanks,
>> > Sven
>> >
>> > --
>> >
>> > Sven Evensen, Operations Consultant
>> >
>> > OnRelay
>> >
>> > Elizabeth House │ 39 York Road, London SE1 7NQ, UK │ +44 (0) 207 902
>> 8123 │
>> > mailto:sven.even...@onrelay.com │ www.onrelay.com
>> >
>> >
>> > This electronic message transmission contains information from OnRelay,
>> > Ltd., that may be confidential or privileged. The information is
>> intended
>> > solely for the recipient and use by any other party is not authorised.
>> If
>> > you are not the intended recipient, be aware that any disclosure,
>> copying,
>> > distribution or use of the contents of this information or any
>> attachment,
>> > is prohibited. If you have received this electronic transmission in
>> error,
>> > please notify us immediately by electronic mail (i...@onrelay.com) and
>> > delete this message, along with any attachments, from your computer.
>> > Registered in England No 04006093 ¦ Registered Office 1st Floor, 236
>> Gray's
>> > Inn Road, London WC1X 8HB
>> >
>> >
>> >
>> > ___
>> > 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/
>
>
>
>
> --
>
> *Sven Evensen, Operations Consultant*
>
> *OnRelay*
>
> Elizabeth House │ 39 York Road, London SE1 7NQ, UK │ +44 (0) 207 902 8123
> │ mailto:sven.even...@onrelay.com  │
> www.onrelay.com
>
>
> This electronic message transmission contains information from OnRelay,
> Ltd., that may be confidential or privileged. The information is intended
> solely for the recipient and use by any other party is not authorised. If
> you are not the intended recipient, be aware that any disclosure, copying,
> distribution or use of the contents of this information or any attachment,
> is prohibited. I

Re: [sipx-users] How to do transfer on unmanaged GW

[Sven Evensen]

Thanks Tony, you confirmed what I was afraid of.

And Yes, we are very close to saying we just cannot use this ITSP.

Our last hope is that we have a plugin where we possibly can intercept the
refer and replace the internal number with the full DID.
Of course some internal numbers might not have a full DID, but we will have
to live with that.


On Fri, Oct 12, 2012 at 4:15 PM, Tony Graziano  wrote:

> I do not blame you for not wanting to do it that way.
>
> It would be better to find an ITSP who supports REFER.
>
> If you use sipxbridge and set it up as a trunk it (sipxbridge) will
> handle the refer locally. It would need the ITSP to support re-invite.
> If you set it up as an unmanaged gateway then the proxy will send the
> REFER to the ITSP (or their gateway, etc.), but it sounds like this
> ITSP does not support reinvite AND requires the "Route by To Header".
> Smacks to me of a best case of what not to look for.
>
> There is a way you can route by the to header in sipxbridge as a
> siptrunk. There is not a way that I know of to do this as an unmanaged
> gateway. I'm not sure how hard it would be to add that feature or what
> else would break or become problematic with users who have lots of
> unmanaged gateways out there since it also assumes the gateway
> supports refer and looks at the INVITE and not the TO header.
>
>
> On Fri, Oct 12, 2012 at 10:46 AM, Sven Evensen 
> wrote:
> > A customer is using an ITSP which does not support re-invite, therefore
> we
> > set it up an unmanaged gateway and REFER is used when doing a transfer.
> If
> > an external call comes through SIP trunk to an AA 200 and the AA does a
> > transfer to say 211, the refer back to the SIP trunk will have refer-to
> as
> > 211 and that is not a number that the SIP trunk knows, it only knows the
> > full DIDs assigned to that trunk.
> >
> > If I program the AA with the full DID of ext 211, then it works. Be we do
> > not want internal users to transfer to internal users with full DID.
> >
> > So is there any way we can do externally originated transfer?
> >
> > Thanks,
> > Sven
> >
> > --
> >
> > Sven Evensen, Operations Consultant
> >
> > OnRelay
> >
> > Elizabeth House │ 39 York Road, London SE1 7NQ, UK │ +44 (0) 207 902
> 8123 │
> > mailto:sven.even...@onrelay.com │ www.onrelay.com
> >
> >
> > This electronic message transmission contains information from OnRelay,
> > Ltd., that may be confidential or privileged. The information is intended
> > solely for the recipient and use by any other party is not authorised. If
> > you are not the intended recipient, be aware that any disclosure,
> copying,
> > distribution or use of the contents of this information or any
> attachment,
> > is prohibited. If you have received this electronic transmission in
> error,
> > please notify us immediately by electronic mail (i...@onrelay.com) and
> > delete this message, along with any attachments, from your computer.
> > Registered in England No 04006093 ¦ Registered Office 1st Floor, 236
> Gray's
> > Inn Road, London WC1X 8HB
> >
> >
> >
> > ___
> > 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/




-- 

*Sven Evensen, Operations Consultant*

*OnRelay*

Elizabeth House │ 39 York Road, London SE1 7NQ, UK │ +44 (0) 207 902 8123 │
mailto:sven.even...@onrelay.com  │ www.onrelay.com


This electronic message transmission contains information from OnRelay,
Ltd., that may be confidential or privileged. The information is intended
solely for the recipient and use by any other party is not authorised. If
you are not the intended recipient, be aware that any disclosure, copying,
distribution or use of the contents of this information or any attachment,
is prohibited. If you have received this electronic transmission in error,
please notify us immediately by electronic mail (i...@onrelay.com) and
delete this message, along with any attachments, from your computer.
Registered in England No 04006093 ¦ Registered Office 1st Floor, 236 Gray's
Inn Road, London WC1X 8HB
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Re: [sipx-users] How to do transfer on unmanaged GW

[Tony Graziano]

I do not blame you for not wanting to do it that way.

It would be better to find an ITSP who supports REFER.

If you use sipxbridge and set it up as a trunk it (sipxbridge) will
handle the refer locally. It would need the ITSP to support re-invite.
If you set it up as an unmanaged gateway then the proxy will send the
REFER to the ITSP (or their gateway, etc.), but it sounds like this
ITSP does not support reinvite AND requires the "Route by To Header".
Smacks to me of a best case of what not to look for.

There is a way you can route by the to header in sipxbridge as a
siptrunk. There is not a way that I know of to do this as an unmanaged
gateway. I'm not sure how hard it would be to add that feature or what
else would break or become problematic with users who have lots of
unmanaged gateways out there since it also assumes the gateway
supports refer and looks at the INVITE and not the TO header.


On Fri, Oct 12, 2012 at 10:46 AM, Sven Evensen  wrote:
> A customer is using an ITSP which does not support re-invite, therefore we
> set it up an unmanaged gateway and REFER is used when doing a transfer. If
> an external call comes through SIP trunk to an AA 200 and the AA does a
> transfer to say 211, the refer back to the SIP trunk will have refer-to as
> 211 and that is not a number that the SIP trunk knows, it only knows the
> full DIDs assigned to that trunk.
>
> If I program the AA with the full DID of ext 211, then it works. Be we do
> not want internal users to transfer to internal users with full DID.
>
> So is there any way we can do externally originated transfer?
>
> Thanks,
> Sven
>
> --
>
> Sven Evensen, Operations Consultant
>
> OnRelay
>
> Elizabeth House │ 39 York Road, London SE1 7NQ, UK │ +44 (0) 207 902 8123 │
> mailto:sven.even...@onrelay.com │ www.onrelay.com
>
>
> This electronic message transmission contains information from OnRelay,
> Ltd., that may be confidential or privileged. The information is intended
> solely for the recipient and use by any other party is not authorised. If
> you are not the intended recipient, be aware that any disclosure, copying,
> distribution or use of the contents of this information or any attachment,
> is prohibited. If you have received this electronic transmission in error,
> please notify us immediately by electronic mail (i...@onrelay.com) and
> delete this message, along with any attachments, from your computer.
> Registered in England No 04006093 ¦ Registered Office 1st Floor, 236 Gray's
> Inn Road, London WC1X 8HB
>
>
>
> ___
> 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] How to do transfer on unmanaged GW

[Sven Evensen]

A customer is using an ITSP which does not support re-invite, therefore we
set it up an unmanaged gateway and REFER is used when doing a transfer. If
an external call comes through SIP trunk to an AA 200 and the AA does a
transfer to say 211, the refer back to the SIP trunk will have refer-to as
211 and that is not a number that the SIP trunk knows, it only knows the
full DIDs assigned to that trunk.

If I program the AA with the full DID of ext 211, then it works. Be we do
not want internal users to transfer to internal users with full DID.

So is there any way we can do externally originated transfer?

Thanks,
Sven

-- 

*Sven Evensen, Operations Consultant*

*OnRelay*

Elizabeth House │ 39 York Road, London SE1 7NQ, UK │ +44 (0) 207 902 8123 │
mailto:sven.even...@onrelay.com  │ www.onrelay.com


This electronic message transmission contains information from OnRelay,
Ltd., that may be confidential or privileged. The information is intended
solely for the recipient and use by any other party is not authorised. If
you are not the intended recipient, be aware that any disclosure, copying,
distribution or use of the contents of this information or any attachment,
is prohibited. If you have received this electronic transmission in error,
please notify us immediately by electronic mail (i...@onrelay.com) and
delete this message, along with any attachments, from your computer.
Registered in England No 04006093 ¦ Registered Office 1st Floor, 236 Gray's
Inn Road, London WC1X 8HB
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Re: [sipx-users] External calls cannot be transferred to voice mail (sipXecs 4.4.0)

[Tony Graziano]

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"  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 
> *To:* 'Henry Kwan' ; 'Discussion list for users of
> sipXecs software' 
> *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 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 <
> 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  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 

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

[Henry Kwan]

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 
To: 'Henry Kwan' ; 'Discussion list for users of sipXecs 
software'  
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 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

Re: [sipx-users] Hacked SipXecs 4.4

[Gerald Drouillard]

On 10/11/2012 11:48 PM, 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?
>
In your /etc/ssh/sshd_config you should have at the very least:
PermitRootLogin no
AllowUsers yoursecretusername
MaxAuthTries 3


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

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[sipx-users] 4.6: polycom and BLF

[George Niculae]

Hi All,

anyone with polycom and 4.6 handy to give BLF a try? Don't know why
yet but I cannot get buttons on TUI

Thanks
George
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Re: [sipx-users] Hacked SipXecs 4.4

[Tony Graziano]

... more -- its a user that does not have login to the OS itself, just
vsftpd, which is restricted to certain commands and must present a
request for its mac address in order to get a configuration file. It
is not logging into linux unless someone changed the rights of the
user.

On Fri, Oct 12, 2012 at 7:30 AM, George Niculae  wrote:
> On Fri, Oct 12, 2012 at 2:26 PM, Tony Graziano
>  wrote:
>> this is not a valid system user unless you have manually added it to the
>> system. I do think the logs would show more if access was granted. Why are
>> you exposing sshd to the outside world with an acl or by protecting it at
>> your firewall?
>>
>
> PlcmSpIp is the user used by polycom phones for fetching config from server
>
> George
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Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.net
Fax: 434.465.6833
~~
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[sipx-users] SipXopenfire timezone settings ?

[Jeff Ferrara]

Hello,

After a recent change to daylight savings (time moved forward one hour) 
all of our messages are coming through spark with timestamps an hour in 
the past

I have already corrected the device timezone, and the server time 
reported through both the Linux console and through sipxconfig are 
correct.  I have also enabled the openfire console and checked there - 
It was reporting the correct timezone, but even after changing it to 
something random, the message timestamp did not change  - Can anyone 
point me toward where I should be setting this?


Thanks for your help.
-Jeff
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Re: [sipx-users] External calls cannot be transferred to voice mail (sipXecs 4.4.0)

[Henry Kwan]

Thank you to all who have given me suggestions.  I'll follow-up on those 
suggestions.

Best regards,

Henry Kwan






 From: Michael Picher 
To: Discussion list for users of sipXecs software 
 
Cc: Henry Kwan  
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  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: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 b

Re: [sipx-users] Hacked SipXecs 4.4

[George Niculae]

On Fri, Oct 12, 2012 at 2:26 PM, Tony Graziano
 wrote:
> this is not a valid system user unless you have manually added it to the
> system. I do think the logs would show more if access was granted. Why are
> you exposing sshd to the outside world with an acl or by protecting it at
> your firewall?
>

PlcmSpIp is the user used by polycom phones for fetching config from server

George
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Re: [sipx-users] Hacked SipXecs 4.4

[Tony Graziano]

this is not a valid system user unless you have manually added it to the
system. I do think the logs would show more if access was granted. Why are
you exposing sshd to the outside world with an acl or by protecting it at
your firewall?

On Thu, Oct 11, 2012 at 11:48 PM, 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.
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> List Archive: http://list.sipfoundry.org/archive/sipx-users/

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Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.net
Fax: 434.465.6833
~~
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Ask about our Internet Fax services!
~~

Using or developing for sipXecs from SIPFoundry? Ask me about sipX-CoLab
2013!

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

[Michael Picher]

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  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:* 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 <
> 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  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 

Re: [sipx-users] Hacked SipXecs 4.4

[Michael Picher]

sipXecs 4.4.0 has no firewall enabled, so if you have your system raw on
the internet or you have port 25 open inbound to it you could have some
sort of DoS related thing going on.

clean out your mail directory, disallow external connections to the server
and see what happens.

doesn't sound like you're 'hacked', just broken.

mike

On Fri, Oct 12, 2012 at 1:55 AM, Davide Poletto wrote:

> 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.
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>>
>
>
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>



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