Re: [sipx-users] Strange behavior in dial plan regarding permissions

[Todd Hodgen]

If the 666 (devil) is the first dial rule, and other dial strings hit it, what 
causes the match?  Do you have 666 as optional possibly?  Look at advanced and 
ensure it is not optional or anyone dialing 10 digits will hit it.
Sent from my twiddling thumbs.

Henry Dogger  wrote:

>I have seen this interesting setup, but since we are also developing on 
>asterisk as we do on sipXecs, we would lose this option.
>But it really shouldn't behave like this in my opion, since I can also imagine 
>using such a dial rule when using just sipXecs
>Any thoughts on the bug/problem?
>
>-Original Message-
>From: sipx-users-boun...@list.sipfoundry.org 
>[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
>Sent: donderdag 10 januari 2013 14:06
>To: Discussion list for users of sipXecs software
>Subject: Re: [sipx-users] Strange behavior in dial plan regarding permissions
>
>I think this would behave differently using asterisk as a gateway.
>
>Have you considered this?
>http://wiki.sipfoundry.org/display/sipXecs/ACD+solution+based+on+Askozia
>
>Since it passes through a SBC it should not be required to make the dialplan 
>adjustments you are using.
>
>On Thu, Jan 10, 2013 at 7:59 AM, Henry Dogger  wrote:
>> Hi all,
>>
>>
>>
>> We stumbled some time ago on a strange behavior in the dial plan 
>> regarding the dial permissions.
>>
>> The situation is as follows:
>>
>>
>>
>> We have a few dial plan rules e.g.
>>
>> -  Mobile phones (required is the mobile call permission)
>>
>> -  Local numbers (required is the local call permission)
>>
>> -  International (required is the international call permission)
>>
>>
>>
>> This all works as aspected, a user without the mobile call permission 
>> is not allowed to call mobile phones.
>>
>> But part of our normal setup is a SIP connection between a sipXecs and 
>> a Asterisk, calls are being routed from asterisk to sipXecs and the 
>> other way around. (the reason why we use an Asterisk is because of the 
>> queue functionality, ACD in sipXecs is not satisfying and also openACD 
>> is still not good enough for us.)
>>
>> Since registering the asterisk as a user on sipXecs is a problem we 
>> decided to create a dial rule in the dial plan with a (to all users on 
>> the system) unknown prefix (e.g. 666).
>>
>> So the custom dial rule we created is 666 and 10 digits will result in 
>> a dial of the last 10 digits on the gateways configured for outbound calls.
>>
>> The problems we get with this dial rule are:
>>
>> -  The rule has to be on top of the other outbound dial rules
>> (Mobile, Local and International in this example) to work, otherwise 
>> sipXecs responds with a unauthorized to Asterisk.
>>
>> -  When this rule is active, all other outbound dial rules (Mobile,
>> Local and International in this example) can be called by all users, 
>> even the users without the desired call permissions, so somehow this 
>> rule breaks the entire permissions system
>>
>>
>>
>> I am curious if this is normal behavior, or did we stumble upon a bug?
>>
>> We are currently running on 4.4 updated till patch 16.
>>
>>
>>
>> Kind regards,
>>
>>
>>
>> Henry Dogger
>>
>> Telecats BV
>>
>>
>>
>>
>> ___
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>> 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] VIP extension

[Todd Hodgen]

What tony said.  Set DND on the line on the executive phone - it simply
turns off the ringer on it, or set it to a ring tone that is silent.  Create
a secondary extension that is in an odd range, and then create a dial plan
that excludes that range, and a dial plan that includes it.   Assigned the
inclusion dial plan to other members of the staff that are allowed to call
it, and the other to the rest of the users on the network.

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Monday, January 07, 2013 3:41 PM
To: Sipx-users list
Subject: Re: [sipx-users] VIP extension

 

By the same token you might be able to craft a rule (dial plan) to redirect
or null process that destination. It would be simpler to turn the ringer off
and simply ignore the calls and add the line to an admin assistant who would
field answering those calls.

 

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] Change MOH on Transfer from AA

[Todd Hodgen]

I'm not sure your question is clear.  Are you asking to have separate MOH
for auto attendant transfers, versus music that is heard when placed on hold
by an individual?

 

If that is your question, I've not seen a method that is designed
specifically to provide one type of Music for Auto Attendant calls, versus
other types.   However, if you setup MOH on a per User Basis, then you would
get different music when placed on hold by an individual user, versus what
the system MOH source is set to.

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Ali Ardestani
Sent: Thursday, January 03, 2013 2:07 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Change MOH on Transfer from AA

 

If this is a passed passing through SIPX bridge you need to set the it on
the bridge config page.

On Thu, Jan 3, 2013 at 12:54 PM, Tommy Laino  wrote:



Is there anyway to change the music that callers hear on
hold when they are transferred from the auto attendant? I
tried using user groups with personal MOH set for the group
but that did not work. This is SipXecs 4.4
--
Tommy Laino
Dome Technologies
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-- 

-- 

Ali S Ardestani   

Telephony Systems Engineer

Private National Mortgage Acceptance Company (PennyMac)

6101 Condor Drive

Moorpark, CA 93021

 

(805) 330-6004 Office

(818) 224-7442 x2654 Office

(626) 817-3512 Mobile

(818) 224-7397 Fax

 

ali.ardest...@pnmac.com

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Re: [sipx-users] Intercom in 4.6

[Todd Hodgen]

Seriously?  You did an intercom call to a softphone in 4.4?  I guess I never
tried it, assuming it wasn't a feature.  What softphones did you
successfully Intercom to?

-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Roman Gelfand
Sent: Wednesday, January 02, 2013 9:38 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Intercom in 4.6

This worked in 4.4 for all phones.  Is this a security feature?  Is there a
way to include all phones?

On Wed, Jan 2, 2013 at 12:06 PM, Todd Hodgen  wrote:
> Intercom for a Softphone?  I don't think you will find that supported.
>
> -Original Message-
> From: sipx-users-boun...@list.sipfoundry.org
> [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Roman 
> Gelfand
> Sent: Wednesday, January 02, 2013 3:00 AM
> To: George Niculae
> Cc: Discussion list for users of sipXecs software
> Subject: Re: [sipx-users] Intercom in 4.6
>
> Oh, I see. Yes, the phone group was not setup.
>
> What do I do if the softphone is not listed when trying to add a phone 
> to phone group?
>
> On Wed, Jan 2, 2013 at 5:40 AM, George Niculae  wrote:
>> Is intercom feature enabled in telephony tab?
>>
>> George
>>
>>
>> On Wednesday, January 2, 2013, Joegen Baclor  wrote:
>>> Compring the line below to a working test i did, there is one 
>>> difference.  A working intercom registrar log looks like this:
>>>
>>>
>>> "2013-01-02T05:27:00.231989Z":765:SIP:DEBUG:devat1.ossapp.com:SipRed
>>> i rectServer-9:7fd1f167c700:SipRegistrar:"[130-MAPPING]
>>> SipRedirectorMapping::lookUp got 0 UrlMapping Permission 
>>> requirements for 1 contacts"
>>>
>>> "2013-01-02T05:27:00.232006Z":766:SIP:DEBUG:devat1.ossapp.com:SipRed
>>> i rectServer-9:7fd1f167c700:SipRegistrar:"[130-MAPPING]
>>> SipRedirectorMapping::lookUp got 1 UrlMapping Contacts"
>>>
>>> "2013-01-02T05:27:00.232028Z":767:SIP:DEBUG:devat1.ossapp.com:SipRed
>>> i rectServer-9:7fd1f167c700:SipRegistrar:"[130-MAPPING]
>>> SipRedirectorMapping::lookUp contact =
>>>
>>>
> ' %3Aloc
> alhost%3E%3Banswer-after%3D0>'"
>>>
>>> "2013-01-02T05:27:00.232067Z":768:SIP:DEBUG:devat1.ossapp.com:SipRed
>>> i rectServer-9:7fd1f167c700:SipRegistrar:"[130-MAPPING]
>>> SipRedirectorMapping::lookUp contactUri =
>>>
>>>
> ' %3Aloc
> alhost%3E%3Banswer-after%3D0>'"
>>>
>>> "2013-01-02T05:27:00.232084Z":769:SIP:DEBUG:devat1.ossapp.com:SipRed
>>> i rectServer-9:7fd1f167c700:SipRegistrar:"[130-MAPPING]
>>> SipRedirectorMapping::lookUp RecordRouteField = 
>>> ''"
>>>
>>>
>>> In the first line, it says "got 0 UrlMapping Permission requirements 
>>> for
>>> 1 contacts" whereas the log you sent has a permission requirement.
>>> "got
>>> 1 UrlMapping Permission requirements"
>>>
>>>
>>> I think this is more in the realm of configuration issue.  I will let
>>> the sipxconfig experts take it from here.   first things first, make
>>> sure you have the phone as a member of the intercom group and you 
>>> have sent profiles after that.
>>>
>>>
>>> Joegen
>>>
>>>
>>>
>>>
>>> "2013-01-02T03:50:10.322912Z":368:SIP:DEBUG:sip.domain.com:SipRedire
>>> c tServer-10:b41f3b70:SipRegistrar:"[130-MAPPING]
>>> SipRedirectorMapping::lookUp got 1 UrlMapping Permission 
>>> requirements for 1 contacts"
>>>
>>>
>>>
>>>
>>>
>>> On 01/02/2013 12:06 PM, Roman Gelfand wrote:
>>>>
>>>> "2013-01-02T03:50:10.322912Z":368:SIP:DEBUG:sip.domain.com:SipRedir
>>>> e ctServer-10:b41f3b70:SipRegistrar:"[130-MAPPING]
>>>> SipRedirectorMapping::lookUp got 1 UrlMapping Permission 
>>>> requirements for 1 contacts"
>>>
>>> ___
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>>> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>>>
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>
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Re: [sipx-users] Intercom in 4.6

[Todd Hodgen]

Intercom for a Softphone?  I don't think you will find that supported.

-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Roman Gelfand
Sent: Wednesday, January 02, 2013 3:00 AM
To: George Niculae
Cc: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Intercom in 4.6

Oh, I see. Yes, the phone group was not setup.

What do I do if the softphone is not listed when trying to add a phone to
phone group?

On Wed, Jan 2, 2013 at 5:40 AM, George Niculae  wrote:
> Is intercom feature enabled in telephony tab?
>
> George
>
>
> On Wednesday, January 2, 2013, Joegen Baclor  wrote:
>> Compring the line below to a working test i did, there is one 
>> difference.  A working intercom registrar log looks like this:
>>
>>
>> "2013-01-02T05:27:00.231989Z":765:SIP:DEBUG:devat1.ossapp.com:SipRedi
>> rectServer-9:7fd1f167c700:SipRegistrar:"[130-MAPPING]
>> SipRedirectorMapping::lookUp got 0 UrlMapping Permission requirements 
>> for 1 contacts"
>>
>> "2013-01-02T05:27:00.232006Z":766:SIP:DEBUG:devat1.ossapp.com:SipRedi
>> rectServer-9:7fd1f167c700:SipRegistrar:"[130-MAPPING]
>> SipRedirectorMapping::lookUp got 1 UrlMapping Contacts"
>>
>> "2013-01-02T05:27:00.232028Z":767:SIP:DEBUG:devat1.ossapp.com:SipRedi
>> rectServer-9:7fd1f167c700:SipRegistrar:"[130-MAPPING]
>> SipRedirectorMapping::lookUp contact =
>>
>>
''"
>>
>> "2013-01-02T05:27:00.232067Z":768:SIP:DEBUG:devat1.ossapp.com:SipRedi
>> rectServer-9:7fd1f167c700:SipRegistrar:"[130-MAPPING]
>> SipRedirectorMapping::lookUp contactUri =
>>
>>
''"
>>
>> "2013-01-02T05:27:00.232084Z":769:SIP:DEBUG:devat1.ossapp.com:SipRedi
>> rectServer-9:7fd1f167c700:SipRegistrar:"[130-MAPPING]
>> SipRedirectorMapping::lookUp RecordRouteField = 
>> ''"
>>
>>
>> In the first line, it says "got 0 UrlMapping Permission requirements 
>> for
>> 1 contacts" whereas the log you sent has a permission requirement.  
>> "got
>> 1 UrlMapping Permission requirements"
>>
>>
>> I think this is more in the realm of configuration issue.  I will let
>> the sipxconfig experts take it from here.   first things first, make
>> sure you have the phone as a member of the intercom group and you 
>> have sent profiles after that.
>>
>>
>> Joegen
>>
>>
>>
>>
>> "2013-01-02T03:50:10.322912Z":368:SIP:DEBUG:sip.domain.com:SipRedirec
>> tServer-10:b41f3b70:SipRegistrar:"[130-MAPPING]
>> SipRedirectorMapping::lookUp got 1 UrlMapping Permission requirements 
>> for 1 contacts"
>>
>>
>>
>>
>>
>> On 01/02/2013 12:06 PM, Roman Gelfand wrote:
>>>
>>> "2013-01-02T03:50:10.322912Z":368:SIP:DEBUG:sip.domain.com:SipRedire
>>> ctServer-10:b41f3b70:SipRegistrar:"[130-MAPPING]
>>> SipRedirectorMapping::lookUp got 1 UrlMapping Permission 
>>> requirements for 1 contacts"
>>
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>> sipx-users@list.sipfoundry.org
>> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>>
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Re: [sipx-users] Backup Name

[Todd Hodgen]

Douglas - that is a nice improvement for the FTP.   :)   As always - Thanks!

-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tommy Laino
Sent: Friday, December 21, 2012 1:50 PM
To: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Backup Name



I am on 4.4 currently and I dont have plans to go to 4.6 yet. Todd that is a
good idea and I will use that solution most likely. Thanks
--
Tommy Laino
Dome Technologies
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Re: [sipx-users] Backup Name

[Todd Hodgen]

I handled this by creating multiple FTP usernames - each site has its own
username.  On the FTP site, the FTP user has their own directory.  This
created a group of directories, each containing its own backups - and each
directory has a sub-directory by date with the backup files in it.

Very simple and straightforward and nothing special done to sipxecs.


-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tommy Laino
Sent: Friday, December 21, 2012 1:05 PM
To: sipx-users@list.sipfoundry.org
Subject: [sipx-users] Backup Name



Is there anyway to change the name of the backup tar file that SipXecs uses?
I have multiple systems backing up to an FTP site but they all have the same
name so I can not tell which is which.
--
Tommy Laino
Dome Technologies
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Re: [sipx-users] ERROR: SipXbridge XML-RPC Exception

[Todd Hodgen]

After you reset the Bridge, and until it is fully reset, you typically see
this error.   Is this machine real slow possibly?

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Matt Nelson
Sent: Tuesday, December 18, 2012 7:30 PM
To: sipx-users
Subject: [sipx-users] ERROR: SipXbridge XML-RPC Exception

 

Ok, got the web interface to come up. Now having issues in the sbc report
area. Getting 



-- ERROR: SipXbridge XML-RPC Exception
 
any ideas?
 
Matt Nelson
Owner/Technical Director
Vitelity/Megapath/Yealink Partner
audiopivotPATC
LIC#10-00016519
Malibu, California
424.781.1666 itsp/fax
Yuma, Arizona
928.597.4777 itsp/fax
matthewe...@audiopivotpatc.com
 
 
"pure technical artistry"
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Re: [sipx-users] Registrations Expiring

[Todd Hodgen]

And you are sure someone's not turning the system off over the weekend by
mistake?  Had a customer turning off circuit breakers for the weekend, and
turning off more than they should have.  UPS died, and it was rebooted on
Monday - but phones were not registered.  Just an example.  I'd really
suggest you look at the logs to see what it is doing rather than locking
into an issue without any proof.  It could be any one of a dozen things
going on. 

-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tommy Laino
Sent: Sunday, December 16, 2012 7:55 AM
To: sipx-users@list.sipfoundry.org
Subject: [sipx-users] Registrations Expiring



For the last 2 weekends I have had a SipXecs 4.4 with 30 Polycom 335's that
has all the registrations expire. I have to resend the server profile and
all the phones re-register.
Weird thing is that this only happens over the weekend or when all the
phones are inactive for a long period of time.
Any ideas? I have the logs but not sure what I should be looking for to try
and troubleshoot this
--
Tommy Laino
Dome Technologies
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Re: [sipx-users] Registrations Expiring

[Todd Hodgen]

You can run log rotate to remove the log files and monitor smaller one.
logrotate -f /etc/logrotate.d/sipxchange

-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tommy Laino
Sent: Sunday, December 16, 2012 8:58 AM
To: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Registrations Expiring



The sipproxy.log file is so large that my text editors will not open it. I
am assuming that it is an attack. I am going to have the IT department close
that port on the firewall and see if we have any luck.
--
Tommy Laino
Dome Technologies
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Re: [sipx-users] You guys didn't make 4.4 to 4.6 easy... good lord!

[Todd Hodgen]

This is kind of a useless email to send out to the mailing list.   Vent, but
personally, I'd prefer my email box not get filled with it.   This release
has been out there for the most part for about 6 months.  Everyone has had
the same opportunity to load it on a test system, and test their own method
of migration.  I've seen many do it, and leave comments.  I've successfully
migrated systems without significant issues.As an open source product -
the Community is vital to assist in testing, validation, and cooperation to
make it a successful product.   

 

To the developers, thanks for your work on the project, it's appreciated by
most.

 

BTW, I hear Shoretel is offering an end of year discount on
licensing.

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Robert B
Sent: Thursday, December 13, 2012 2:37 PM
To: sipx-users@list.sipfoundry.org
Subject: [sipx-users] You guys didn't make 4.4 to 4.6 easy... good lord!

 

This should not be this difficult.

Disclaimer: I'm going to sound snarky and really pissed. It's how I vent
frustration. I apologize in advance.

Issues:

1. My BIND backups that I restore get destroyed every time sipx restarts.
Real nice, gents, real nice...

2. Why does iptables keep turning back on every time sipx restarts?
a. If SipX wants to be DNS, why on earth is port 53 blocked in the
default iptables configuration? Hello, MyFly...

3. After restoring my configuration from 4.4 to 4.6, I was also bitten by
the incorrect Superadmin PIN. Fortunately, I knew to use the reset-admin
from other posts.

4. Once my configuration is restored, what services do I need to re-enable?
The entire "Telephony Services" list is unchecked. Which do I need to have
what I had with 4.4? Boy howdy, these backups I was told to make sure are
proving useful...

5. It restored my Polycom firmware, but marked it as inactive. What the
hell?

6. Where can I monitor my registrations? That feature seems gone.

Did I maybe download the wrong installation ISO? I downloaded
"sipxecs-4.6.0-932.g00cded-x86_64.iso"... Is this the real GA release???

-- Robert

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Re: [sipx-users] Message Waiting Issue

[Todd Hodgen]

Maybe a stretch - Checked the POE switch?  Had a phone resetting just the
other day when they hit the messages button on the phone - as bizarre as
that sounds, it turned out to be a cable issue between switch and phone.  Go
figure?!?  Still scratching my head.

-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tommy Laino
Sent: Tuesday, December 11, 2012 2:16 PM
To: sipx-users@list.sipfoundry.org
Subject: [sipx-users] Message Waiting Issue



I got SipXecs 4.4 with all Polycom 335/550 phones with 3.2.6 firmware.
Everything was working just fine until today.
Suddenly people would take their voicemails and the light would not
extinguish. On the other hand others are receiving messages and not getting
any MWI. What gives? I checked the SUBSCRIBE under the messaging for the
line and they all show the users extension. What log am I looking for to see
if the user subscription is registering?
--
Tommy Laino
Dome Technologies
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Re: [sipx-users] Initial release: sipXecs 4.6.0

[Todd Hodgen]

Congratulations Douglas to you and your entire Team.  This substantial
release demonstrates a new level of flexibility and reliability to an
already world class product!  A warm thank you to everyone involved!!!

-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Douglas Hubler
Sent: Wednesday, December 05, 2012 9:29 AM
To: sipx-users
Subject: [sipx-users] Initial release: sipXecs 4.6.0

I'm pleased to announce sipXecs release 4.6.0.  There will most certainly be
updates to this release in the form of new RPMs published in the coming
weeks, so if you've reported issues against 4.6.0, it's quite possible
they'll be out soon. Given that, this release should be considered ready for
production use.

Support for polycom 4.0 is forth-coming as an update, but otherwise the
4.6.0 release will continue in a bug fixing mode only. Those looking to test
polycom 4.0 support should checkout 4.7.0 development builds.

I'll resist the temptation to discuss how proud I am of the dev team and the
users that helped with all the testing (although I really am!). I'll also
resist mentioning how long it took to get this release out but instead I'll
just keep this nice and simple and let out a big sigh of relief as I reach
for the heavily spiked eggnog.
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[sipx-users] Polycom Conference Bridge SIP Interface

[Todd Hodgen]

I'm looking for anyone that has interfaced sipXecs with a Polycom Conference
Bridge's SIP Audio capabilities.   I'm working on rounding up the technical
specifications for their product, which I assume is a standard SIP endpoint,
but am really guessing right now until I see specs.

 

If anyone has done this already, I would love to hear details on it.

 

Thanks in advance for any information you might have.

 

Regards,

 

Todd Hodgen

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Re: [sipx-users] sipx 4.6 inbound ivr-->extension always not available

[Todd Hodgen]

Nicholas - Your details are pretty extensive here, so I may be pointing out
the obvious - but have you confirmed what ports the RTP is using, and if it
is configured in the router to be passed.   Not always, but many times I
find RTP issues are generally related to ports configuration (or not) in the
routers, SIP ALG's, etc.

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Nicholas Drayer
Sent: Sunday, December 02, 2012 9:54 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] sipx 4.6 inbound ivr-->extension always not
available

 

No Trevor: the MoH is a different issue, once I changed the Server, the
Gateway, and the User to have no branch (aka all branches), that problem
went away. I still think there's an issue there, but MoH is definitely a
different issue.

 

The actual problem is that under the following conditions audio is not
received or sent either way:

- An incoming call from the outside (i.e. to provider --> sipXecs over port
5080 --> IVR --> extension with Bria softphone)

- A Bria softphone

- The User has their setting for IM enabled.

 

These conditions are very precise:

- a call from one extension to another works fine;

- an outgoing call works fine;

- substituting the Bria phone with a SoundPoint desktop works fine --
toggling the IM setting for the user makes no difference to the SoundPoint;

- it doesn't matter whether or not the Bria phone settings are provisioned
for XMPP or not, it's merely the User IM setting that's important;

- the ini file generated by sipXecs for the Bria is identical whether or not
the IM setting for the User is enabled;

 

I would surmise that port 5080 is working fine -- the Bria does ring, you
can answer the call and the other side sees that you answered. You can hang
up. It's the audio that is missing. In this particular situation the RTP is
nog going either way.

 

I put an issue in: xx-10547. Hopefully I'm explaining the issue clearly
enough. I've tested it on two separate setups of sipXecs 4.6 over two weeks,
with identical repeatable results.

 

 

  _  

From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] on behalf of Trevor L Benson
[tben...@a-1networks.com]
Sent: November 30, 2012 6:23 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] sipx 4.6 inbound ivr-->extension always not
available

The provider I have tied into my 4.6 system is specifically because they
refuse to redirect inbound to anything but 5060.  So I would have to URI
dial and rely on SRV to test, and then still tcpdump to verify it was
happening as your configuration is currently.  In Mikes follow up email he
clarified Proxy was working externally on 5060, maybe just give it a test
and see if there is any difference. 

 

At this point I have to admit I think I am confusing the actual problem with
the Subject of the original thread.  Is the problem extension not available,
or did I just zone out this has been a MoH issue the whole time since its
the Bria components?  

 

In any case if proxy is on 5060, I would potentially test inbound signaling
to 5060 from your provider.  If I get time to upgrade my 4.4 server this
weekend now that we finished our basic testing for contact center I should
be able to use Bandwidth.com and simulate your environment on port 5080 and
compare to normal 5060 through my alternate provider.

 

Were you using strictly desktop Bria versions, or were any
iPhone/iPad/Android versions being tested as well?

 

Thanks,

Trevor Benson, Network Engineer

A1 Networks

Voice: 707-703-1041

 

For support issues please email supp...@a-1networks.com or call 707-703-1050

 

 

 

 

On Nov 30, 2012, at 4:45 PM, Nicholas Drayer 
wrote:





Yes, just to confirm, it's 100% 5080 that I'm using. I use IP authentication
for the SIP trunk, with incoming calls going to d...@pbx.domain.com:5080.

 

  _  

From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] on behalf of Michael Picher
[mpic...@ezuce.com]
Sent: November 30, 2012 2:49 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] sipx 4.6 inbound ivr-->extension always not
available

you'll find you have transfer issues and whatnot with port 5060...  it is
not the right port.  5080 is where sipXbridge is listening on.  this is not
a typo.  the proxy listens on port 5080. 

 

mike

 

On Fri, Nov 30, 2012 at 4:21 PM, Trevor L Benson 
wrote:

Nicholas,

 

  This has been on my mind a little over the past few days, was this just a
typo from the 4.4 configuration for ITSP's using 5080 inbound, or is the
ITSP still sending to you on 5080?  We tested out inbound on 5060 (albeit
weeks ago and many updates) and it was working fine.  I am guessing its a
typo, or you configured your system properly to still use 5080, but it's
sticking in my craw that maybe this isn't a typo or the ITSP is sending to
the 

Re: [sipx-users] Fax Service Question

[Todd Hodgen]

You should see something from mail going out.  So you are broken somewhere
ahead of that.  Are you hearing fax tone when you call it?

Normally, when things are broke, you at least get an email that says 0 pages
received.  At least that has been my experience.

-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tommy Laino
Sent: Friday, November 30, 2012 3:02 PM
To: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Fax Service Question



The DID is entered on the Unified Messaging page. The call does complete
when I look in the CDR. However, the maillog is not showing anything for the
fax call.
--
Tommy Laino
Dome Technologies
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Re: [sipx-users] Fax Service Question

[Todd Hodgen]

Or, User is 12345, and unified messaging has a fax mailbox of
2223334567

 

In maillog, for the fax call, what DSN does it provide for the mail forward?
2.0 or something else?

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Friday, November 30, 2012 2:39 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Fax Service Question

 

The user is 12345 and their unifies messaging has a fax mailbox of 54321 and
a fax DID ON THE UNIFIED messaging page of 2223334567. Right?

On Nov 30, 2012 5:33 PM, "Tommy Laino"  wrote:



The user is extension 5011. The fax extension is 6011. The
fax DID is set to the T.38 fax number. What do mean a setup
a fax box? I think I have everything setup the way you are
describing
--
Tommy Laino
Dome Technologies
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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] Fax Service Question

[Todd Hodgen]

That should work.  Alternatively, you can put the did for the fax as the fax
extension number - just one number to deal with then.

When you call the number from another phone, do you hear Fax tones?

-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tommy Laino
Sent: Friday, November 30, 2012 2:34 PM
To: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Fax Service Question



The user is extension 5011. The fax extension is 6011. The fax DID is set to
the T.38 fax number. What do mean a setup a fax box? I think I have
everything setup the way you are describing
--
Tommy Laino
Dome Technologies
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Re: [sipx-users] CyberData

[Todd Hodgen]

You can also use a low cost ATA (Linksys, Grandstream, Audiocodes) that
allows you to call an extension number, the ATA pots port connects to a
valcom or bogen Page Adapter, and any number of standard speakers.   Old
fashioned way of doing things, but it works very reliably.

All three of those ATA/Gateways are managed by sipxecs, making configuration
very simple.

-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Mark Theis
Sent: Friday, November 30, 2012 11:10 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] CyberData

Yes. I like them. I have installed 6 or so. Only problem I have is that they
need to be rebooted every 6 months or so. Not all of them, just random ones.
Lol. 

I have daisy chained 6 of their non-ip speakers to one ip speaker and they
worked great. Also daisy chained multiple existing warehouse speaker/horns
and they worked perfectly! There must have been 10 of the horns in the giant
warehouse. And the 1 ip ceiling speaker powered them all.

Sent from my iPhone

On Nov 30, 2012, at 11:03 AM, "Bryan Anderson"  wrote:

> Does any one have any experience with the Cyberdata SIP Ceiling paging
speakers?
> 
> 
> -Bryan Anderson
> 
> ___
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> sipx-users@list.sipfoundry.org
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Re: [sipx-users] sipx 4.6 inbound ivr-->extension always not available

[Todd Hodgen]

I'm not sure just how important the Bria Softphone really is.  Personally, I
don't use it or recommend it to customer.   Many have found their support
lacking, and they don't seem to be Channel friendly.  

 

Have you tried other clients?  I've had good results with the 3CX softphone,
and they now have a network components for monitoring their use.

 

Additionally, eZuce has a product that comes with their licensed product -
openUC.   You might want to look at that as well.

 

 

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Nicholas Drayer
Sent: Thursday, November 29, 2012 6:11 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] sipx 4.6 inbound ivr-->extension always not
available

 

I put in XX-10547 for the IM-enabled interfering with incoming audio
problem. Seems pretty major to me, making the Bria softphone rather useless.

 

If I have time I'll see if I can make the MoH issue repeatable this weekend,
and then put an issue in.

 

Thanks for helping - Joegen, you made me look closer at the client, at which
point I realized I had one that did work, and one that didn't, even though
diff-ing their phone's .ini files showed them to be identical apart from
their credentials. So then I realized there had to be something different
between the two users, not their phone settings.

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Michael Picher
Sent: November-29-12 2:38 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] sipx 4.6 inbound ivr-->extension always not
available

 

fwiw, i never put servers into branches.

 

i only use branches to have users prefer particular gateways on dial-out.  i
don't see that putting a server in a particular branch has any bearing here.

 

mike

 

On Thu, Nov 29, 2012 at 12:58 AM, Nicholas Drayer
 wrote:

At last, I am now able to turn the problem incoming call audio problem on
and off repeatably and reliably: it has something to do with XMPP. Below are
the repeatable steps. Does anyone have an idea what I should do with this...
put in a bug report? I don't want to do that without someone's advice, since
it's entirely possible that I'm missing something vital that I don't know
about. 

 

Step 1

---

Create a user that has IM disabled (User list-screen --> click on the User
ID --> Instant Messaging --> uncheck Enabled --> OK)

Create a phone profile that has IM disabled (Devices --> click on the Line
--> XMPP --> uncheck top two boxes)

Send Profiles

Login Bria

Audio works both ways when dialling in from SIP trunking provider.

 

Step 2

---

User list-screen --> click on the User ID --> Instant Messaging --> check
Enabled --> Apply

Incoming call from the SIP trunking provider now has no audio either way
after being transferred to the Bria client by the IVR

 

Step 3

---

User list-screen --> click on the User ID --> Instant Messaging --> uncheck
Enabled --> Apply

Logout of Bria

Login to Bria

Audio works both ways when dialling in from SIP trunking provider.

 

Note A



For Step 2, you do NOT have to logout of Bria. But for Step 3 (making it
work properly again) you DO have to logout of and login back with Bria, even
thought the profile is not changed whatsoever.

 

Note B



Turning the Bria phone line XMPP on (Devices --> Phones --> click on Line
--> XMPP --> top two checkboxes) makes no difference to the above behaviour,
except that it does enable Bria's IM features if the IM is enabled on the
user's screen. All IM features work properly.

 

The MoH issue definitely has something to do with branch settings: I removed
the branch from the Server, the Gateway, and the Users (they were all
identical, currently the system only has one; so now they are "All") and now
it works fine. It's unrelated to the above. If I can replicate it I'll
report it too.

 

Nicholas Drayer, Managing Director
Dyrand Systems Inc. 
Tel: 604.408.4415 ext. 319  
  email |   web

  _  

From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] on behalf of Nicholas Drayer
[nicholasdra...@dyrand.com]
Sent: November 28, 2012 5:05 PM


To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] sipx 4.6 inbound ivr-->extension always not
available

 

The MoH issue is related to Bria: I've now figured out that the MoH issue
only occurs happens with the Mac OS X Bria client. Same for the hold/resume
issue - works properly on Windows, can't resume on  Mac. Both exhibit the
main problem of no audio in/out for external received calls.

 

What I find difficult to wrap my head around is that the MoH on the calling
external calling line works when the IVR connects and rings a Windows Bria
client, but not for a Mac Bria client.

 

I haven't mentioned this, but calling from extension 

Re: [sipx-users] Sendmail Issue

[Todd Hodgen]

I suspect your issue is with Verizon, and not the router.   They are
blocking your ability to send email through them as an SMTP gateway.

-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tommy Laino
Sent: Saturday, November 24, 2012 7:45 PM
To: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Sendmail Issue



Just to update this thread. I took my new instance of SipXecs to the
customer prem to start my install. Setup their firewall and SipXecs and when
I was done I decided to try to see if sendmail would work their. Sure enough
first try the test mail delivered to my email account and worked just fine.
Tested UM and that worked flawlessly as well

I am assuming that the Actiontec router provided by Verizon is not working
properly. I have all the same ports open on both firewalls with differing
results. I am going to swap out the router on Monday and see if that fixes
my issue.
Thank you for all your help guys
--
Tommy Laino
Dome Technologies
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Re: [sipx-users] Call transfer from AutoAttendant to PSTN

[Todd Hodgen]

You could set up multiple extensions with call forwarding to a particular
number.   Ext 111 - 206-4567890, Ext 112 - 206-2254589, etc.   From auto
attendant, dial the extension.   

If you explain what you are trying to do in more detail from an application
standpoint, I suspect there are other suggestions that could be provided.



-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of ??
Sent: Tuesday, November 20, 2012 5:45 PM
To: Discussion list for users of sipXecs software
Subject: [sipx-users] Call transfer from AutoAttendant to PSTN

Is there any way to make call transfer from AA to external PSTN ? 

Now I dial 100 to AA from my ext. and then dial prefix 9 with PSTN, I always
get the voice "that ext is not valid" from AA. I can dial prefix 9 with PSTN
directly to my MGW without problem.
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Re: [sipx-users] Yealink SIP-T26P/T28P + EXP38

[Todd Hodgen]

Have you consided using a simple low cost Polycom phone along with Voice
Operator Panel as a solution?   It works great!

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Geoff Musgrave
Sent: Tuesday, November 20, 2012 11:11 AM
To: Discussion list for users of sipXecs software
(sipx-users@list.sipfoundry.org)
Subject: [sipx-users] Yealink SIP-T26P/T28P + EXP38

 

Has anyone used the Yealink SIP-T26P or T28P with or without the EXP38 with
sipXecs  4.6? If so, I would like to hear your experience and/or opinion.

 

I'm considering placing 2 of these with the EXP38 at another office due to
the owners not wanting to spend Polycom prices. My other option is to go
with snom 3xx with expansion modules again but I'm still having some
transferring issues with the current snom 370s at the office we did the
initial deployment.

 

Thanks in advance.

 

--

Geoff

 

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Re: [sipx-users] voip.ms impression

[Todd Hodgen]

Yes, the Seattle switch is horrible, and I don't recommend it.  They say
that Seattle and Chicago are their two newest and most up to date switches,
as well as least busy.  But I can't use Seattle any longer.

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Tuesday, November 20, 2012 10:57 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] voip.ms impression

 

Today alone I had two sites with a authentication issue. In one instance I
had two registrations that sipx said were authenticated. voip.ms says none
were. Restrating the gateways said they were authenticated, voip.ms still
says no. Incoming still worked even though it was not registered, which is
"wrong" but outbound would not work until we moved from seattle.

 

Moving one registration from seattle to NY and restart gateways... now
voip.ms agreed.

 

Same thing in Dallas, sipx says YES voip.ms says NO. Restart gateway, no.
Move to NY and restart, now voip.ms says yes.

 

I'm not particularly fond of their tech support. "It must be you". The one
constant in all this is voip.ms...

On Tue, Nov 20, 2012 at 1:22 PM, Burleigh, Matt
 wrote:

I've recently(2 months) started using voip.ms and my support experience has
been similar. Ever since Hurricane Sandy I've had numerous issues. I can
usually restart SIP trunking to restore service and I don't always get an
alarm from sipx. I've had some recent complaints of busy signals as well...


 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Mark Wood
Sent: Monday, November 19, 2012 19:26
To: 'sipx-users@list.sipfoundry.org'
Subject: [sipx-users] voip.ms impression

 

We have been using voip.ms for 6-8 months so far and I want to get some
feedback from users that have been at it for longer. 

 

Specifically we (main and subaccounts) experience times where our outbound
calls just hang after dialing and sometimes abruptly connect, or sometimes
not at all. When a subaccount calls to report problems to us and we check
our home page it will show all of our accounts as 'not registered', and then
slowly one by one they will show as 'registered'. We had an incident over
the weekend with a security office that couldn't receive any inbound calls.
We logged in to the voip.ms site to check the registrations and initiate a
support ticket and the site again said 'not registered'. The instructions
had us do and 'echo test' procedure and the results were the same as when
they were routed to the subaccount. The support response 12 hours later was
'works for us' and then 'check your routers and firewalls'.

 

Comments? Who are other good candidates for reselling VoIP like this model?

 

Thanks,

Mark W. Wood

office: (760)202-0224   X2010

Direct: (760)459-1981

New Image.BMP

www.redphonetech.com

 

 

 

 


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-- 
~~
Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.net
Fax: 434.465.6833
~~
Linked-In Profile:
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~~

 

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


 

 

LAN/Telephony/Security and Control Systems Helpdesk:

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Re: [sipx-users] voip.ms impression

[Todd Hodgen]

Specifically, we do not use VOIP.ms as primary service for customers.  We
find they are not reliable enough.  They are great for testing, quick
temporary service, occasional usage such as conference bridge, international
dialing, etc.

 

I don't believe their issues are related to routers in most cases.  As an
example, I have three sites that have sub-accounts setup from my own
personal account with VOIP.ms.  They all register to the same server at
VOIP.ms.   I received error messages from all three sites that they cannot
connect to the ITSP for service, at the same time.   Later, from the same
three sites, I get a message that they are now reconnected.

 

Each site uses a different ITSP, each site is separate and not working with
the other sites in any way.  The only common thread is the service itself.
I've spoken to VOIP.ms, they claim it is not them, check the routers.1
pfsence router, 1 watchguard, and 1 Sonicwall - go figure.

 

I say use VOIP.ms, it's a great service.  But, don't put many eggs in that
basket.

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Nicholas Drayer
Sent: Monday, November 19, 2012 8:24 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] voip.ms impression

 

We use voip.ms for three smaller not-for-profit clients. We found in the
past that all our problems were due to firewalls: due to the way SIP works,
firewall problems can be intermittent and vexing. Without those specific
(expensive but rubbish) firewalls the problems do not occur.

 

On the first site have had no problems for a year now, using 15 softphones
and pfSense (with HAVP and Snort) as a firewall using voip.ms' PBX features.

On a second site we have no problems over the last 3 months using 4 Polycom
SoundPoint ip 335's connecting to voip.ms' PBX features.

 

On a third site setup a month ago, we are experiencing no problems with
sipXecs 4.6 on AWS EC2 connected to voip.ms through sipXecs' built-in
iptables (and the AWS security group setup properly). On this sipXecs 4.6
instance we have a backup trunk to Toronto that is registered, and we use IP
authentication on the primary trunk to New York (with an inbound DID
redirected using an URI to port 5080). No problems at all to date. Fingers
crossed. knocking on wood, etc. We have a sipXecs alarm set on the secondary
registered connection, and it did registered a drop once for a couple of
minutes two weeks ago.

 

I would say that voip.ms' support isn't geared towards people having
problems with their firewalls. I'm guessing they are usually right that it's
the end-user's device, not voip.ms' servers. It must be a very frustrating
job! They have always been polite and reasonably helpful to me. But don't
expect emergency support within the hour, unless it really is their systems
down.

Indeed, voip.ms documents somewhere on their website that they not support
T.38.

As always, YMMV, in other words, it might not work for you even though it
works great for me!

 

Nicholas Drayer

Managing Director

Dyrand Systems

T. 604.408.4415 Ext. 319

www.dyrand.com

 

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: November-19-12 4:51 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] voip.ms impression

 

We use voip.ms for small occasional use and its very good for testing. Our
production systems use Appia (we are a reseller) who can also provide t.38
service which we have found to work well in a production environment.

 

At the same time, most of our sip servers sit behind application aware
firewalls and ensure connections using SIP the protocol are in now way
impaired by other applications on the network. These application aware
devices also have IDS and are completely cloud managed and monitored.

 

I would agree voip.ms works but not always that well at every POP and they
do seem to have occasional problems with call quality and completion
depending on the number you call. Whether this is ISP routing, edge or
internal issues at their POP is sometimes a little unclear, but they are
good enough to use for failover, backup or testing but not particulary in
large production environments (IMO).

On Mon, Nov 19, 2012 at 7:25 PM, Mark Wood 
wrote:

We have been using voip.ms for 6-8 months so far and I want to get some
feedback from users that have been at it for longer. 

 

Specifically we (main and subaccounts) experience times where our outbound
calls just hang after dialing and sometimes abruptly connect, or sometimes
not at all. When a subaccount calls to report problems to us and we check
our home page it will show all of our accounts as 'not registered', and then
slowly one by one they will show as 'registered'. We had an incident over
the weekend with a security office that couldn't receive any inbound calls.
We logged in to the voip.ms site to check the registra

Re: [sipx-users] Hacked SipXecs 4.4

[Todd Hodgen]

Thanks for the confirmation Noah.

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Noah Mehl
Sent: Friday, November 16, 2012 9:52 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Hacked SipXecs 4.4

 

I can confirm that adding: 

 

DenyUsers PlcmSpIp

 

to /etc/ssh/sshd_config solves this exploit.

 

I'm back to my original opinion that if this user is installed
automatically, without my intervention, then that line should be added to
the sshd_config.

 

~Noah

 

On Nov 16, 2012, at 12:46 PM, Noah Mehl  wrote:





Tony, 

 

I just figured out an exploit in 15 minutes with the help of Google
http://www.semicomplete.com/articles/ssh-security/:

 

$sudo ssh -vN -L25:localhost:25 PlcmSpIp@sipxecsip

$sudo ssh -vN -R25:localhost:25 PlcmSpIp@sipxecsip

$telnet localhost 25

 

Tell me if your ids stops that?

 

This works on a stock SipXecs 4.4.0 install.

 

~Noah

 

On Nov 16, 2012, at 11:46 AM, Tony Graziano 

 wrote:





The user doesn't have login via ssh. Ssh in and of itself is not protected
and it is exposed.

It is trivial to change the user password and/or delete it. We typically
don't expose ssh at all. You haven't provides any real evidence that a
dictionary attack didn't overwhelm the pam service either.

I don't share your opinion here. My firewall protects against all kinds of
ids stuff even if I had ssh open. Just because you have iptables running it
doesn't mean you are inherently secure at all.

Our firewalls sitting in front of sipx had ids rules running that would
protect anything behind it from a known attack against a well known service
like ssh. Ssh has lots of options which should be exercised according to
your security border device.

On Nov 16, 2012 11:36 AM, "Noah Mehl"  wrote:

The only hardening required to solve this particular problem would be an
addition to the sshd config: 

 

DenyUsers PlcmSpIp

 

I think this should be included in the default distribution of SipXecs isos
and/or packages (I've only ever used the iso) because this is something that
is specific to the distribution.  That user, and its password and access,
are created by SipXecs, and that addition to the sshd config should be made
OOTB.  Unless someone has a reason that PlcmSpIp should be able to have any
ssh access?

 

I'd really like some input from someone from eZuce, as this is an easy
solution and protects the entire community.

 

This was NOT a DDOS attack.  This it that the PlcmSpIp user has a default
password of PlcmSpIp, and there's something about the default access of that
user that allow remote execution via SSH OOTB, and that IS a security issue.
You know why?  Because as far as I know, no other default linux service
account is susceptible to this attack.  Probably because linux system
accounts DON'T HAVE PASSWORDS!  In other words, if you're creating service
users with default passwords, they probably should be denied from ssh OOTB.
This is also, not documented as far as I can tell...

 

~Noah

 

On Nov 16, 2012, at 11:26 AM, Tony Graziano 
wrote:





It really sounds like you don't have a method to harden your server if you
are exposing it. Its entirely possible you were targeted with a ddos attack
that overwhelmed the Linux system. If you had properly crafted iptables
rules I and ssh protection mechanisms it would most likely not have
happened.

Any did or ddos can overwhelm system services to the point of failure this
allowing (by unavailability) internal logging or protection mechanisms. Put
the served behind a firewall and protect the vulnerable service (ssh) by
limiting the footprint. Backup the system, wipe and restore it in the event
a root kit was planted.

I don't think iptables was adequately configured. I don't think there is
anything inherently wrong with Sipx here either.

It is a phone system. It is up to you to protect and/or harden it. Any
vulnerabilities exposed are really Linux vulnerabilities and Linux is not
hack proof.

Good luck.

On Nov 16, 2012 10:07 AM, "Noah Mehl"  wrote:



Todd,

The private subnet is: 172.16.0.0 - 172.31.255.255.  That IP is a public IP
address, which is part of AOL in Nevada I think.  I actually have over 80
different public IP address entries in my log using that user to SSH to my
SipXecs box.

I understand that it's a phone system and not a firewall.  However it's a
linux server, and IPtables is the best firewall in world, IMHO.  I did have
SSH access open to the world, that was my choice.  I have never been bitten
by this before.  Either way, you should not be able to execute anything by
SSH'ing with the PlcmSpIp user, whether it's a public IP or not.

~Noah

On Nov 15, 2012, at 7:07 PM, Todd Hodgen  wrote:

> Here is a question I would have as well - 172.129.67.195 seems to be an
> address that is local to your network.   

Re: [sipx-users] Linksys SPA3102 FXO MGW Outbound Calling

[Todd Hodgen]

Under line - Proxy and Registration - be sure to check "Use Outbound Proxy"
and enter an outbound Proxy if not already checked.

-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of ??
Sent: Friday, November 16, 2012 6:26 AM
To: Discussion list for users of sipXecs software
Cc: Discussion list for users of sipXecs software
Subject: [sipx-users] Linksys SPA3102 FXO MGW Outbound Calling

Has anyone any experience to config  Spa3102 outbound call from ext. to PSTN
? I can make call from PSTN to ext. through Spa3102 now.

I created unmanaged gateway on Spa3102 and dial plan with prefix 9, but Spa
always claimed "not found" when I made outbound call.

Regards,

Jarvis.
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Re: [sipx-users] Hacked SipXecs 4.4

[Todd Hodgen]

Here is a question I would have as well - 172.129.67.195 seems to be an
address that is local to your network.   Who has that IP address, why are
they attempting to breach that server.   If they are not a part of your
network, how are they getting to that server from outside your network -
there has to be an opening in a firewall somewhere that is allowing it.

Remember, this is a phone system, not a firewall, not a router.   It's a
phone system with pretty standard authentication requirements, it's up to
the administrator to keep others off of the network.

-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Noah Mehl
Sent: Thursday, November 15, 2012 10:04 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Hacked SipXecs 4.4

To that point:

 Users logging in through sshd:
PlcmSpIp:
   172.129.67.195 (AC8143C3.ipt.aol.com): 1 time

That can't be good.  I understand that PlcmSplp is a user for the Polycom
provisioning.  I have removed ssh access to the box from the world, but how
do I change the default password for that user?  This seems like a big
security risk, as every sipxecs install probably has this user with a
default password?

~Noah

On Nov 15, 2012, at 12:41 PM, Todd Hodgen  wrote:

> Look at var/spool/mail/rootThere is a report you can find in there
that
> shows system activity.  Look for entries below - 
> pam_unix Begin  and I think you will find the 
> source of your aggravation.
> 
> -Original Message-
> From: sipx-users-boun...@list.sipfoundry.org
> [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Noah Mehl
> Sent: Thursday, November 15, 2012 6:29 AM
> To: Discussion list for users of sipXecs software
> Subject: Re: [sipx-users] Hacked SipXecs 4.4
> 
> I am seeing more spam in my mail queue.  I have iptables installed, 
> and here are my rules:
> 
> Chain INPUT (policy ACCEPT)
> target prot opt source   destination
> RH-Firewall-1-INPUT  all  --  anywhere anywhere
> 
> Chain FORWARD (policy ACCEPT)
> target prot opt source   destination
> RH-Firewall-1-INPUT  all  --  anywhere anywhere
> 
> Chain OUTPUT (policy ACCEPT)
> target prot opt source   destination
> 
> Chain RH-Firewall-1-INPUT (2 references)
> target prot opt source   destination
> ACCEPT all  --  anywhere anywhere
> ACCEPT icmp --  anywhere anywhereicmp any
> ACCEPT esp  --  anywhere anywhere
> ACCEPT ah   --  anywhere anywhere
> ACCEPT udp  --  anywhere 224.0.0.251 udp dpt:mdns
> ACCEPT udp  --  anywhere anywhereudp dpt:ipp
> ACCEPT tcp  --  anywhere anywheretcp dpt:ipp
> ACCEPT all  --  anywhere anywherestate
> RELATED,ESTABLISHED
> ACCEPT tcp  --  anywhere anywherestate NEW tcp
> dpt:pcsync-https
> ACCEPT tcp  --  anywhere anywherestate NEW tcp
> dpt:http
> ACCEPT tcp  --  anywhere anywherestate NEW tcp
> dpt:xmpp-client
> ACCEPT tcp  --  anywhere anywherestate NEW tcp
> dpt:5223
> ACCEPT all  --  192.168.0.0/16   anywhere
> ACCEPT udp  --  anywhere anywherestate NEW udp
> dpt:sip
> ACCEPT tcp  --  anywhere anywherestate NEW tcp
> dpt:sip
> ACCEPT tcp  --  anywhere anywherestate NEW tcp
> dpt:sip-tls
> ACCEPT udp  --  sip02.gafachi.comanywherestate NEW udp
> dpts:sip:5080
> ACCEPT udp  --  204.11.192.0/22  anywherestate NEW udp
> dpts:sip:5080
> REJECT all  --  anywhere anywherereject-with
> icmp-host-prohibited
> 
> As far as I can tell, no one should be able to use port 25 from the world.
> Also, sendmail is only configured to allow relay from localhost:
> 
> [root@sipx1 ~]# cat /etc/mail/access
> # Check the /usr/share/doc/sendmail/README.cf file for a description # 
> of the format of this file. (search for access_db in that file) # The 
> /usr/share/doc/sendmail/README.cf is part of the sendmail-doc # package.
> #
> # by default we allow relaying from localhost...
> Connect:localhost.localdomain   RELAY
> Connect:localhost   RELAY
> Connect:127.0.0.1   RELAY
> 
> Can someone please help me figure out where this spam is coming from?
> Thanks.
> 
> ~Noah
> 
> On Oct 13, 2012, at 10:17 AM, Noah Mehl  wrote:
&g

Re: [sipx-users] Hacked SipXecs 4.4

[Todd Hodgen]

, relay=sentinel1.tranet.net.
[74.203.219.99], dsn=2.0.0, stat=Sent (Ok: queued as 62A2D1BFDAB)

Nov 14 22:34:23 sipx1 sendmail[1047]: qAF3YKuO001047:
from=, size=5874, class=0, nrcpts=1,
msgid=<201211150334.qaf3ykuo001...@sipx1.sip.tranet.net>, proto=SMTP,
daemon=MTA, relay=localhost.localdomain [127.0.0.1]

Nov 14 22:34:23 sipx1 sendmail[1047]: qAF3YKuO001047:
to=, delay=00:00:01, mailer=relay, pri=35874,
dsn=4.4.3, stat=queued

Nov 14 22:34:25 sipx1 sendmail[32547]: qAF3YKuO001047:
to=, delay=00:00:03, xdelay=00:00:02, mailer=relay,
pri=125874, relay=sentinel1.tranet.net. [74.203.219.99], dsn=2.0.0,
stat=Sent (Ok: queued as 479D81C0BDE)

Nov 15 01:50:09 sipx1 sendmail[3176]: qAF6o7o5003176:
from=, size=5874, class=0, nrcpts=1,
msgid=<201211150650.qaf6o7o5003...@sipx1.sip.tranet.net>, proto=SMTP,
daemon=MTA, relay=localhost.localdomain [127.0.0.1]

Nov 15 01:50:09 sipx1 sendmail[3176]: qAF6o7o5003176:
to=, delay=00:00:01, mailer=relay, pri=35874,
dsn=4.4.3, stat=queued

Nov 15 01:50:14 sipx1 sendmail[32547]: qAF6o7o5003176:
to=, delay=00:00:06, xdelay=00:00:03, mailer=relay,
pri=125874, relay=sentinel1.tranet.net. [74.203.219.99], dsn=2.0.0,
stat=Sent (Ok: queued as 309221C0F12)

Nov 15 03:19:19 sipx1 sendmail[4123]: qAF8JHpS004123:
from=, size=5874, class=0, nrcpts=1,
msgid=<201211150819.qaf8jhps004...@sipx1.sip.tranet.net>, proto=SMTP,
daemon=MTA, relay=localhost.localdomain [127.0.0.1]

Nov 15 03:19:19 sipx1 sendmail[4123]: qAF8JHpS004123:
to=, delay=00:00:00, mailer=relay, pri=35874,
dsn=4.4.3, stat=queued

Nov 15 03:19:23 sipx1 sendmail[32547]: qAF8JHpS004123:
to=, delay=00:00:04, xdelay=00:00:01, mailer=relay,
pri=125874, relay=sentinel1.tranet.net. [74.203.219.99], dsn=2.0.0,
stat=Sent (Ok: queued as 6B4E11C0F51)

Nov 15 03:26:33 sipx1 sendmail[4210]: qAF8Q78r004210:
from=, size=5925, class=0, nrcpts=50,
msgid=<201211150826.qaf8q78r004...@sipx1.sip.tranet.net>, proto=SMTP,
daemon=MTA, relay=localhost.localdomain [127.0.0.1]

 

As opposed to a normal entry:

 

Nov 15 10:02:23 sipx1 sendmail[11170]: qAFF2NI4011170:
from=, size=335352, class=0, nrcpts=1,
msgid=<1578812003.338.1352991743551.javamail.sipxcha...@sipx1.sip.tranet.net
>, proto=ESMTP, daemon=MTA, relay=localhost.localdomain [127.0.0.1]

Nov 15 10:02:23 sipx1 sendmail[11170]: qAFF2NI4011170:
to=, delay=00:00:00, mailer=relay, pri=365352,
dsn=4.4.3, stat=queued

Nov 15 10:02:40 sipx1 sendmail[10780]: qAFF2NI4011170:
to=, delay=00:00:17, xdelay=00:00:14, mailer=relay,
pri=455352, relay=sentinel1.tranet.net. [74.203.219.99], dsn=2.0.0,
stat=Sent (Ok: queued as 501B41C1CBE)

 

So, they are being generated locally, as far as I can tell.

 

~Noah

 

On Nov 15, 2012, at 12:42 PM, Todd Hodgen 

 wrote:





+1

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org]On Behalf Of Michael Picher
Sent: Thursday, November 15, 2012 7:49 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Hacked SipXecs 4.4

 

yes, and using the word hacked as your subject is not particularly...
helpful...

 

On Thu, Nov 15, 2012 at 3:32 PM, Tony Graziano <
<mailto:tgrazi...@myitdepartment.net> tgrazi...@myitdepartment.net> wrote:

you really need to look at the mail log to see where the mail is coming from
regardless of your firewall settings. It can actually come from inside you
see.

 

On Thu, Nov 15, 2012 at 9:29 AM, Noah Mehl < <mailto:n...@tritonlimited.com>
n...@tritonlimited.com> wrote:

I am seeing more spam in my mail queue.  I have iptables installed, and here
are my rules:

Chain INPUT (policy ACCEPT)
target prot opt source   destination
RH-Firewall-1-INPUT  all  --  anywhere anywhere

Chain FORWARD (policy ACCEPT)
target prot opt source   destination
RH-Firewall-1-INPUT  all  --  anywhere anywhere

Chain OUTPUT (policy ACCEPT)
target prot opt source   destination

Chain RH-Firewall-1-INPUT (2 references)
target prot opt source   destination
ACCEPT all  --  anywhere anywhere
ACCEPT icmp --  anywhere anywhereicmp any
ACCEPT esp  --  anywhere anywhere
ACCEPT ah   --  anywhere anywhere
ACCEPT udp  --  anywhere 224.0.0.251 udp dpt:mdns
ACCEPT udp  --  anywhere anywhereudp dpt:ipp
ACCEPT tcp  --  anywhere anywheretcp dpt:ipp
ACCEPT all  --  anywhere anywherestate
RELATED,ESTABLISHED
ACCEPT tcp  --  anywhere anywherestate NEW tcp
dpt:pcsync-https
ACCEPT tcp  --  anywhere anywherestate NEW tcp
dpt:http
ACCEPT tcp  --  anywhere anywherestate NEW tcp
dpt:xmpp-client
ACCEPT tcp  --  anywhere anywherestate NEW tcp
dpt:5223
ACCEPT all  --   <http://192.168.0.0

Re: [sipx-users] Hacked SipXecs 4.4

[Todd Hodgen]

+1

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Michael Picher
Sent: Thursday, November 15, 2012 7:49 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Hacked SipXecs 4.4

 

yes, and using the word hacked as your subject is not particularly...
helpful...

 

On Thu, Nov 15, 2012 at 3:32 PM, Tony Graziano
 wrote:

you really need to look at the mail log to see where the mail is coming from
regardless of your firewall settings. It can actually come from inside you
see.

 

On Thu, Nov 15, 2012 at 9:29 AM, Noah Mehl  wrote:

I am seeing more spam in my mail queue.  I have iptables installed, and here
are my rules:

Chain INPUT (policy ACCEPT)
target prot opt source   destination
RH-Firewall-1-INPUT  all  --  anywhere anywhere

Chain FORWARD (policy ACCEPT)
target prot opt source   destination
RH-Firewall-1-INPUT  all  --  anywhere anywhere

Chain OUTPUT (policy ACCEPT)
target prot opt source   destination

Chain RH-Firewall-1-INPUT (2 references)
target prot opt source   destination
ACCEPT all  --  anywhere anywhere
ACCEPT icmp --  anywhere anywhereicmp any
ACCEPT esp  --  anywhere anywhere
ACCEPT ah   --  anywhere anywhere
ACCEPT udp  --  anywhere 224.0.0.251 udp dpt:mdns
ACCEPT udp  --  anywhere anywhereudp dpt:ipp
ACCEPT tcp  --  anywhere anywheretcp dpt:ipp
ACCEPT all  --  anywhere anywherestate
RELATED,ESTABLISHED
ACCEPT tcp  --  anywhere anywherestate NEW tcp
dpt:pcsync-https
ACCEPT tcp  --  anywhere anywherestate NEW tcp
dpt:http
ACCEPT tcp  --  anywhere anywherestate NEW tcp
dpt:xmpp-client
ACCEPT tcp  --  anywhere anywherestate NEW tcp
dpt:5223
ACCEPT all  --  192.168.0.0/16   anywhere
ACCEPT udp  --  anywhere anywherestate NEW udp
dpt:sip
ACCEPT tcp  --  anywhere anywherestate NEW tcp
dpt:sip
ACCEPT tcp  --  anywhere anywherestate NEW tcp
dpt:sip-tls
ACCEPT udp  --  sip02.gafachi.comanywherestate NEW udp
dpts:sip:5080
ACCEPT udp  --  204.11.192.0/22  anywherestate NEW udp
dpts:sip:5080
REJECT all  --  anywhere anywherereject-with
icmp-host-prohibited

As far as I can tell, no one should be able to use port 25 from the world.
Also, sendmail is only configured to allow relay from localhost:

[root@sipx1 ~]# cat /etc/mail/access

# Check the /usr/share/doc/sendmail/README.cf file for a description
# of the format of this file. (search for access_db in that file)
# The /usr/share/doc/sendmail/README.cf is part of the sendmail-doc
# package.
#
# by default we allow relaying from localhost...
Connect:localhost.localdomain   RELAY
Connect:localhost   RELAY
Connect:127.0.0.1   RELAY

Can someone please help me figure out where this spam is coming from?
Thanks.

~Noah


On Oct 13, 2012, at 10:17 AM, Noah Mehl  wrote:

> I did not change the configuration of anything related to the PlcmSpIp
user.  It does however make me feel better that it is related to the vsftpd
service and the polycom phones.
>
>> From /etc/passwd:
>
>
PlcmSpIp:x:500:500::/var/sipxdata/configserver/phone/profile/tftproot:/sbin/
nologin
>
> So, that user cannot ssh to a shell. So I don't think it was that.
>
> ~Noah
>
> On Oct 12, 2012, at 9:05 AM, Tony Graziano 
wrote:
>
>> ... 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
>>> ___
>>> 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
>>

Re: [sipx-users] Hacked SipXecs 4.4

[Todd Hodgen]

Look at var/spool/mail/rootThere is a report you can find in there that
shows system activity.  Look for entries below -
pam_unix Begin  and I think you will find the source
of your aggravation.   

-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Noah Mehl
Sent: Thursday, November 15, 2012 6:29 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Hacked SipXecs 4.4

I am seeing more spam in my mail queue.  I have iptables installed, and here
are my rules:

Chain INPUT (policy ACCEPT)
target prot opt source   destination 
RH-Firewall-1-INPUT  all  --  anywhere anywhere

Chain FORWARD (policy ACCEPT)
target prot opt source   destination 
RH-Firewall-1-INPUT  all  --  anywhere anywhere

Chain OUTPUT (policy ACCEPT)
target prot opt source   destination 

Chain RH-Firewall-1-INPUT (2 references)
target prot opt source   destination 
ACCEPT all  --  anywhere anywhere
ACCEPT icmp --  anywhere anywhereicmp any 
ACCEPT esp  --  anywhere anywhere
ACCEPT ah   --  anywhere anywhere
ACCEPT udp  --  anywhere 224.0.0.251 udp dpt:mdns 
ACCEPT udp  --  anywhere anywhereudp dpt:ipp 
ACCEPT tcp  --  anywhere anywheretcp dpt:ipp 
ACCEPT all  --  anywhere anywherestate
RELATED,ESTABLISHED 
ACCEPT tcp  --  anywhere anywherestate NEW tcp
dpt:pcsync-https 
ACCEPT tcp  --  anywhere anywherestate NEW tcp
dpt:http 
ACCEPT tcp  --  anywhere anywherestate NEW tcp
dpt:xmpp-client 
ACCEPT tcp  --  anywhere anywherestate NEW tcp
dpt:5223 
ACCEPT all  --  192.168.0.0/16   anywhere
ACCEPT udp  --  anywhere anywherestate NEW udp
dpt:sip 
ACCEPT tcp  --  anywhere anywherestate NEW tcp
dpt:sip 
ACCEPT tcp  --  anywhere anywherestate NEW tcp
dpt:sip-tls 
ACCEPT udp  --  sip02.gafachi.comanywherestate NEW udp
dpts:sip:5080 
ACCEPT udp  --  204.11.192.0/22  anywherestate NEW udp
dpts:sip:5080 
REJECT all  --  anywhere anywherereject-with
icmp-host-prohibited 

As far as I can tell, no one should be able to use port 25 from the world.
Also, sendmail is only configured to allow relay from localhost:

[root@sipx1 ~]# cat /etc/mail/access
# Check the /usr/share/doc/sendmail/README.cf file for a description # of
the format of this file. (search for access_db in that file) # The
/usr/share/doc/sendmail/README.cf is part of the sendmail-doc # package.
#
# by default we allow relaying from localhost...
Connect:localhost.localdomain   RELAY
Connect:localhost   RELAY
Connect:127.0.0.1   RELAY

Can someone please help me figure out where this spam is coming from?
Thanks.

~Noah

On Oct 13, 2012, at 10:17 AM, Noah Mehl  wrote:

> I did not change the configuration of anything related to the PlcmSpIp
user.  It does however make me feel better that it is related to the vsftpd
service and the polycom phones.
> 
>> From /etc/passwd:
> 
> PlcmSpIp:x:500:500::/var/sipxdata/configserver/phone/profile/tftproot:
> /sbin/nologin
> 
> So, that user cannot ssh to a shell. So I don't think it was that.
> 
> ~Noah
> 
> On Oct 12, 2012, at 9:05 AM, Tony Graziano 
wrote:
> 
>> ... 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
>>> ___
>>> 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:
>> h

Re: [sipx-users] Interconnecting with Asterisk

[Todd Hodgen]

Shane,  Your efforts to continue to use the Asterisk as an ATA, although
valiant, might be a daunting task.  Have you considered simply picking up a
used Linksys ATA on ebay for $20-$30 and calling it a day?   I suspect their
might be some resell value in your current IP04 to help pay for it.   I
suspect this would make for a much simpler installation and much better use
of time.  Especially considering most talk on this mailing list is around
commercial applications of this product.

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Shane Harrison
Sent: Sunday, November 11, 2012 11:20 AM
To: sipx-users@list.sipfoundry.org
Subject: [sipx-users] Interconnecting with Asterisk

 

Hi there,

I would like to move to using SipXecs as my SIP based PBX at home.  I
currently have an IP04 (Asterisk appliance blackfin box) with FX0 and FXS
ports.

What I am looking to do is to continue using the Asterisk box as a PSTN
gateway and to provide a couple of analog extensions.  I would like however
for the voicemail for the analog extensions to be on the SipXecs system.

The PSTN gateway side of things is fine.  My problem is more to do with the
analog extensions ie. how best to do this in a way that is of course
transparent to the end user.  Initially I thought that the SipX should
simply be setup to send any calls for the analog extension to the Asterisk
box and when it times out, send the call to the voicemail.  However I can't
have voicemail on the SipX without setting up a user and if I use the same
extension number eg. 300, then a call to 300 never reaches my dialplan entry
to send it to the Asterisk box since SipX sees a local user with username
300 first.

Another way would be to have a dummy user on SipX eg. 399, and use that as
the Voicemail box for 300, but I suspect access to the voicemail box may
become more difficult for users.

A third way would be to get Asterisk to on forward the call after a timeout
on the analog extension and pass it directly to Freeswitch which has a
voicemail box for 300 that SipX doesn't know about.  However again access to
the mailbox from the web portal has just got difficult again.

What I am really trying to do is set up Asterisk as an analog ATA I guess -
any pointers on the best topology to use would be appreciated.

Kind regards
Shane

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Re: [sipx-users] Sendmail Issue

[Todd Hodgen]

There is a sample provided in the like I had sent.  Copy and paste it,
that's all you really need to do, and then change the from field to your
own.

-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tommy Laino
Sent: Wednesday, November 07, 2012 5:25 PM
To: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Sendmail Issue



Matt thanks for the in depth testing. I changed the zone file to match what
I am getting from the Google DNS servers.
When I look at the logs I am getting the connection was refused on
127.0.0.1. 

Todd that file is not on the system so I will have to build it and give that
a shot.
--
Tommy Laino
Dome Technologies
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Re: [sipx-users] Busy Lamp Field on the SPA504G

[Todd Hodgen]

The SPA 504 templates were donated by two developers, and introduced in 4.4
A later email I had read from one of the developers indicated there were
issues, and they were not going to address them because each Cisco release
made changes that created a moving target.  

 

I do have a customer that is running on older SPA942 phones.  They have
their quirks, but work for the customer for the most part.

 

I did manually create a 504G template and have it downloading and working
from sipxecs well.

 

Have you tried configuring it from the web interface, and not with the
template in sipxecs?  You might have better luck.

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Wednesday, November 07, 2012 6:54 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Busy Lamp Field on the SPA504G

 

Just because the templates are there doesn't mean they are maintained. They
might have worked for whoever initiated the template for the way they were
using it.

 

I have always found Polycom to give the most predictable results and have
mixed feeling about certain ATA's and gateways in general. 

 

I have centered around a "need for compatibility" using some gateway
products that meet our needs (plain analog calls, no features but also fax
(via t.38) compliant. 

 

Everyone's need is different, but if you get to the point where you have
defined the use case and feature sets, you will get lots of opinions on here
from people who actually deploy and use the stuff they are suggesting you to
use.

 

I don't know who maintains the SPA templates. I don't know what features are
purported to work properly (or at all).

 

On Wed, Nov 7, 2012 at 9:44 AM, Richard Bruce 
wrote:

I have made life very difficult.  I was ready to drop the SPA as well and
then I loaded SipXecs 4.6 and found that templates have been added for them
and decided to give them another try.  I have spent many hours between the
two in the last couple of weeks and would like to be able to give back to
the forum.

 

 

Richard Bruce

Dimensional Communications 

7915 S. Emerson Ave, Suite 131

Indianapolis, IN  46237
(317) 215-4199- office

(317) 946-1899 - cell

  _  

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Michael Picher
Sent: Wednesday, November 07, 2012 9:06 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Busy Lamp Field on the SPA504G

 

Wow, you're trying to make all the hard stuff work :-)

 

SPA/Linksys phones, Asterisk as a gateway...   

 

On Wed, Nov 7, 2012 at 8:11 AM, Richard Bruce 
wrote:

I am trying to configure the BLF on the SPA504G.  Looking for any info on
how to set up.

 

Thanks,

 

Richard Bruce

Dimensional Communications 

7915 S. Emerson Ave, Suite 131

Indianapolis, IN  46237
(317) 215-4199- office

(317) 946-1899   - cell

 


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-- 
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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-- 
~~
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] Sendmail Issue

[Todd Hodgen]

I've not been following this thread, but will add some comments as they
relate to email not being deliverable outbound via sipxecs.  If this is DNS
related - this response doesn't touch that.

I find it useful in v4.4 to add the Emailformats.properties files to
sipxecs.
http://wiki.sipfoundry.org/display/sipXecs/Voicemail-Email+Custom+Notificati
ons   It will allow you to set the Sent from email address to one that is
acceptable to the your SMTP gateway.

After installing the sample, and changing your from address, do a "service
sendmail restart", and do a restart of sipxconfig.

Looking at var/log/maillog can give you some clues as to why it is not being
delivered.  Pay attention to the to and from address, the DNS codes give
some great google clues as well.

-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of
m...@mattkeys.net
Sent: Tuesday, November 06, 2012 10:20 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Sendmail Issue

You said your SIP domain is the same as your email domain, and that your DNS
is hosted at Network Solutions. This is likely the issue and not a problem
with sendmail unless you've borked the default configs. Sendmail in that use
case would be used to deliver the email and not to receive it. 

There are lots of reasons why sendmail wouldn't be able to deliver email but
that discussion is outside the scope for a sipxecs mailing list. Verify
easily that sendmail is working in the first place by using an alternate
email address for the below test with a known good address like gmail,
yahoo, etc.

$ echo "testing 123" | mail -s "sipx email test" youremailaddr...@gmail.com
&& tail -f /var/log/maillog

You should see successful delivery in the mail log (press ctrl-c to stop
tailing it), and within a few seconds or minutes the email should arrive in
your inbox. Be sure to check your junk mail folders for the email if you
don't spot it quickly. If you have successful delivery of the test email
then sendmail is working just fine and it's a local DNS issue. To verify
this is the case (replace example.com with your real domain) :

$ dig @localhost mx example.com

then compare against what the rest of the world has for your MX by using an
outside DNS such as Google :

$ dig @8.8.8.8 mx example.com

If they don't match you'll need to edit your zone file, correct the MX
record, then restart BIND.

$ service named restart

Repeat the dig test to verify they're the same now. You may want to consider
the use of the smart_host config option of sendmail. To do so your "real"
email server will need to allow mail relay from your sipx box. Optionally
you could reconfigure sipx to use a different SIP domain all together or a
subdomain of your real domain (voice.example.com instead of example.com).

HTH,
Matt

From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tommy Laino
[tomla...@gmail.com]
Sent: Tuesday, November 06, 2012 11:34 PM
To: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Sendmail Issue

So I added my MX records and A records in my zone file. But when I try to
telnet mydomain.com 25  I get that the connection is refused by 127.0.0.1
still. Tried the same thing on my new server and its not working either. Did
some other research on Google and cant seem to find why this is happening.
--
Tommy Laino
Dome Technologies
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Re: [sipx-users] csv import for users on 4.4

[Todd Hodgen]

You can always create random generated passwords to fill in your spreadsheet
as well using a formula.

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Brent Besse
Sent: Thursday, November 01, 2012 11:10 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] csv import for users on 4.4

 

This is normal behaviour from what I have seen. As far as I have been able
to see the system does not generate the SIP password. If you read the
Documents it gives you the layout of the file.

 

I use a vb script to read a .csv file asks for the MAC of the phone then
populates every field as required.

I imort the file. setup the unified communications parametes to turn on VM,
send the profiles and plug in the phones and let programing happen. Works
great for me in our environment.

 

...Brent...

Brent Besse

Royal Rods University,

Victoria BC,

Canada

  _  

From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] on behalf of Bryan Anderson
[shadow...@gmail.com]
Sent: Thursday, November 01, 2012 9:20 PM
To: Discussion list for users of sipXecs software
Subject: [sipx-users] csv import for users on 4.4

Is it normal for the latest 4.4 to not generate a sip password in import of
a csv file?  If so, is there a way to have the system generate the passwords
after import?

Thanks,
-Bryan Anderson

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Re: [sipx-users] voicemail restore issues

[Todd Hodgen]

I suspect you are 100% right on that one.  I'd dig into it a bit further.
Thanks Tony and Mike.

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Tuesday, October 30, 2012 11:41 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] voicemail restore issues

 

It's most likely a permissions issue. 

 

If you make a local backup on the new system then delete the voicemail
archive and move the backup onto it (onboard), you "should" be able to do a
restore from local backup.

 

Once you "graft" voicemail, the permissions for the folders and files need
to be correct, you can easily match that up against any existing system but
I think if you tested one user it would bear this out as the reason. all
user/group ownership should be sipxchange:sipxchange and folders should be
0755 while files should be 0644.

On Tue, Oct 30, 2012 at 2:28 PM, Todd Hodgen  wrote:

I had a server that needed to be rebuilt yesterday with 4.4.  I ran a backup
of voicemail and configuration, rebuilt the server, but had an error when
trying to restore the voicemail.Don't have exact error, but it was
complaining the file was too large.  And it was big, about 1.2 tb.

 

Suspect this was the wrong thing to do now, but here is what I did.  I
unzipped the gz file to a tar file, and then opened the tar file to a temp
folder.  I then created the mailstore folder and copied the voicemails into
it.

 

So, I got my personal greetings to play in the voicemail boxes, but users
are unable to delete messages in voicemail now.  Error in sipstatus.log says
invalid nonce.

 

I'm thinking of deleting old emails in the mailstore, recreating a new gz
file and trying to restore that smaller file.   Anyone see any issues with
this?


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-- 
~~
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! <http://sipxcolab2013.eventbrite.com/?discount=tony2013> 


 

 

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] voicemail restore issues

[Todd Hodgen]

I had a server that needed to be rebuilt yesterday with 4.4.  I ran a backup
of voicemail and configuration, rebuilt the server, but had an error when
trying to restore the voicemail.Don't have exact error, but it was
complaining the file was too large.  And it was big, about 1.2 tb.

 

Suspect this was the wrong thing to do now, but here is what I did.  I
unzipped the gz file to a tar file, and then opened the tar file to a temp
folder.  I then created the mailstore folder and copied the voicemails into
it.

 

So, I got my personal greetings to play in the voicemail boxes, but users
are unable to delete messages in voicemail now.  Error in sipstatus.log says
invalid nonce.

 

I'm thinking of deleting old emails in the mailstore, recreating a new gz
file and trying to restore that smaller file.   Anyone see any issues with
this?

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Re: [sipx-users] Getting 483 Too Many Hops Error

[Todd Hodgen]

eZuce provides commercial support with OpenUC, their commercial version.   I
suspect one of the sales staff for eZuce will reach out to you due to this
email.   There are other features you gain via OpenUC which you should have
explained as well, and add to the value of the product.

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Brian Buckles
Sent: Monday, October 22, 2012 2:56 PM
To: Brian Buckles; Matt White; sipx-users@list.sipfoundry.org; Discussion
list for users of sipXecs software
Subject: Re: [sipx-users] Getting 483 Too Many Hops Error

 

All,

I've heard that we could get a commercial copy of SipXecs that would have
phone support.  Is this true and if so what would be the cost?  Is the phone
support 24/7? Also what all would be involved?  Would we simply be able to
purchase a license and not have to update the SipXecs software or alter the
the configuration?  I wanted to ask within this post, so we know what are
options are for commercial license.  Can someone please provide any info on
commercial support?  

 

  _  

From: Brian Buckles 
To: Brian Buckles ; Matt White
; "sipx-users@list.sipfoundry.org"
; Discussion list for users of sipXecs
software  
Sent: Monday, October 22, 2012 5:34 PM
Subject: Re: [sipx-users] Getting 483 Too Many Hops Error





Mike I'll send you a copy of a trace once I get the complete trace.  In the
mean time I have a quick question.  The link for SipViewer that was in the
url you sent is broken.  I've searched and the links I've found point to the
same broken url.  Do you happen to know where I can get a working copy of
the SipViewer.  

Thanks

 

  _  

From: Brian Buckles 
To: Matt White ; "sipx-users@list.sipfoundry.org"
 
Sent: Monday, October 22, 2012 3:12 PM
Subject: Re: [sipx-users] Getting 483 Too Many Hops Error

 

Thanks Mike,

I'll get a capture together from the SipXecs and post it as well.

 

 

  _  

From: Matt White 
To: sipx-users@list.sipfoundry.org; itnc...@yahoo.com 
Sent: Monday, October 22, 2012 2:37 PM
Subject: Re: [sipx-users] Getting 483 Too Many Hops Error

 

A sipx-trace will be far more beneficial.  What you have is only gonna show
whats happening at the gateway...its not gonna show us anything about how
sipx is dealing with it internally.

http://www.sipfoundry.org/web/mpicher/~/426137/blogs/-/asset_publisher/xfZRF
9U0rLa7/blog/id/78163

-M
>>> Brian Buckles  10/22/12 1:18 PM >>>

All,

Please see the link below for the captures from the Ingate and BroadVox.
The Ingate capture is called "Configured by Ingate Startup Tool..." and can
be viewed from a web browser.  You can scroll toward the bottom to see the
capture and search for "483 too many hops if needed".  The BroadVox capture
is labeled "BroadVox_Capture_726910-3.pcap" and is just a standard pcap
file.  Once you follow link just click on download for each file then click
"Click here to start download from sendspace" at the bottom to save to your
PC.  As an FYI I spoke to Ingate and we can upgrade the Ingate a new version
that will allow the preservation of hops.  I'm going to hold off on this for
now to prevent downtime of the phone system until necessary and see if
BroadVox can increase the hops.  Thanks for the help and please let me know
if the captures offer any further help and suggestions you may have.

 

 

http://www.sendspace.com/filegroup/6kkw1N%2BO9fAUeWPKwPXWTg

 

 

  _  

From: Tony Graziano 
To: Discussion list for users of sipXecs software
 
Cc: itnc...@yahoo.com 
Sent: Friday, October 19, 2012 2:30 PM
Subject: Re: [sipx-users] Getting 483 Too Many Hops Error


Except with an ingate we can probably preserve the hop count inside
and make it a non issue (I think).

On Fri, Oct 19, 2012 at 2:18 PM, Matt White 
wrote:
> Attachments typically don't make ti to the list.  Post them to a webserver
> and provide a url.
>
> 483 Too many Hops is generally because the call is getting
> referred/re-invited too many times.
>
> The ITSP will typically send the call with predefined number of "hops" set
> in the SIP header.  We often find that this changes even within a single
> ITSP depending on where the call originates.  Most ITSP set this around 10
> hops.
>
> its not hard for this to decrement down to 0.  Call hits the ingate
> (1)...Call Goes to an alias on an AA (2), call is sent from the alias to
the
> AA (3), AA transfers to a hunt group (4)etc.  A few hops are used
> internally to sipx as well. Eventually it bounces around too many times.
>
> So for starters see if you can simplify your call routing.
>
> Its also good to fully qualify call forwarding routes as it removes a hop.
> For example.
>
> Lets say you have extension 201 as a dumy user and it sends calls to hunt
> group 202.
> When you setup the call forwarding enter it as 202@sipx.domain
>
> If you just put 202 in their...it will take 2 hops.  One when it sends 202
> to the proxy, and a second when the proxy changes 202 to 202@s

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

[Todd Hodgen]

Henry,  Try allowing the existing sipxecs template configure the phone,
without making the changes in the wiki to the profiles.

 

I have a system with approximately 20 of those phones working perfectly with
the system managing the templates for the phones completely.

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Henry Kwan
Sent: Friday, October 19, 2012 4:40 PM
To: Tony Graziano; Joegen Baclor
Cc: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] External calls cannot be transferred to voice mail
(sipXecs 4.4.0)

 

Hi Tony,

 

I really appreciate that you took the time to elaborate in detail below.  I
shall follow-up and perform your suggestions when time permits.  Please also
see my response below.

 

Best regards,

 

Henry Kwan

 

  _  

From: Tony Graziano 
To: Joegen Baclor  
Cc: Discussion list for users of sipXecs software
; Henry Kwan  
Sent: Friday, October 19, 2012 2:36:25 AM
Subject: Re: [sipx-users] External calls cannot be transferred to voice mail
(sipXecs 4.4.0)


Primus is also LINGO. Primus is a large aggregator and also runs a
residential service (Lingo). The Lingo service does not support invite
without sdp, unless the reinvite is to one of their services and
typically only from one of their ATA's.

>> OK, I'll ask Primus about this.

I think you would do well to ask them if they support this AND it is
very important to make sure the invite for the incoming call comes to
your server on port 5080.

>> I've confirmed with Primus that they could accept signalling on 5060 on
their side and sent signalling to us on 5080.

I don't think your issue is unusual and usually stems from one of 3
core misconfiguration types:

1. Incompatible ITSP - Does a transfer from the AA to a user work?
Does a call from a user to another user work? (both as inbound calls
via the trunk). Is the original invite coming on port 5080.

>> Did not test a transfer from the AA to a user, will try that.
>> Call from a user (internal phone) to another user through local dialing
plan (i.e. 9-...) worked.
>> I think the original invite must come on port 5080 as that was the port
that was forwarded.  5060 was not forwarded.

2. Does the phone ring? If so, how was it configured (manually of by
sipx)? Please tell me you didn't register the line manually using the
sipx ip address. DNS is VERY important for the refer to voicemail. IF
you registered by IP, make sure you add the IP as a domain alias, but
really you should NEVER register by IP and expect all things to work
well.

>> Yes, the phone rang.  The phone, Linksys SPA942, was configured via the
sipX web pages.  I then reset the phone and have the configuration
downloaded to the phone via TFTP (I think this is the mechanism).
>> I re-installed sipXecs 4.4 a number of times.  Sometimes IP as a domain
alias would appear automatically.  I've also manually done that.  In any
event, that did not help.
>> I also run the tests on the configuration test page and everything
passed, including DNS checks.  I've also downloaded Flight Test (I think
that was the name) and everything passed.

3. Firewall - Unless your firewall is doing 1:1 NAT, you need to make
sure it can do Manual AON (static port nat). With pfsense this is
easy, but YOU CANNOT create port forward rules until you do this for
SIPX becuase they will follow the original NAT type. I sent you a link
of how to do this earlier, its pretty straightforward. You should be
able to use the pcap tool in pfsense and have it listen on WAN port
5080 and do a capture and see if the ITSP sends the call in on the
right port. If not, it will never work right (no matter what version)
and you need to ask them if they support this.

>> I did not do 1:1 NAT as I was not sure how to do that properly.  I've
read up on it now and will try that out in the future.  For port forwarding,
I did do Manual AON with static port checked on pfSense.  I also needed to
create rules to pass this traffic.  Same thing was done for port range 3
to 31000.  With this setup on pfSense, I could call in from an external
phone but still could not transfer to voice mail when no one answered.  It
behaved exactly the same as using other routers - fast busy when the attempt
of transfer was made.
>> I have not had time to follow the link that you sent me earlier but will
definitely read up on it.

Good luck!



On Fri, Oct 19, 2012 at 1:50 AM, Joegen Baclor  wrote:
> Transferring ITSP originated calls requires that your ITSP supports INVITE
> without SDP.  Before barking on something on the system, check first if
your
> ITSP supports this.  If not, there is no way your ITSP will work with sipx
> initiated transfers.
>
>
>
> On 10/19/2012 01:19 PM, Tony Graziano wrote:
>>
>> On Fri, Oct 19, 2012 at 1:07 AM, Henry Kwan  wrote:
>>>
>>> My installation was right from the 4.4 ISO.  I did try without updating
>>> at
>>> all but to no avail.
>>>
>>> My ITSP is Primus Canada.
>>>
>

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

[Todd Hodgen]

I don't think this jells completely with your description, but when you do a
restart of a service, it does take some time for the restart and the
services to work again.  The sipxbridge is a good example, because it
restarts, and then must re-register before you can make call on it.

 

Check on the wiki under sipxviewer.  It is a utility you can run on your
windows machine, or on the linux server to view traces.   Do a search on the
wiki for siptrace as well, I believe there are good instruction on how to do
it there.  You can google is as well - here is an article by our very own
Mike Picher.
http://www.sipfoundry.org/web/mpicher/~/426137/blogs/-/asset_publisher/xfZRF
9U0rLa7/blog/id/78163

 

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Henry Kwan
Sent: Thursday, October 18, 2012 10:08 PM
To: George Niculae; Discussion list for users of sipXecs software
Subject: Re: [sipx-users] External calls cannot be transferred to voice mail
(sipXecs 4.4.0)

 

My installation was right from the 4.4 ISO.  I did try without updating at
all but to no avail.

My ITSP is Primus Canada.

Well I have to admit that I am not knowledgeable in setting up pfSense.  In
fact I am not knowledgeable on how to produce a pcap or produce a siptrace
as Tony suggested.  Having said that, I'll continue to play with 4.4 and
look into how to perform the tasks suggested when time permits.

In the mean time, 4.2.1 will have to suffice until I can figure out what I
did wrong.

By the way, my observation regarding the inconsistent behaviour on restarts
for both 4.2.1 and 4.4.0 is not an attack on sipXecs.  It is just that - an
observation.  Maybe someone can comment if this observation is also only
experienced by me.  If that's the case, I must be a jinx or have a unique
ability to bring out the worst in sipXecs.

 

Best regards to all,

 

Henry Kwan

 

  _  

From: George Niculae 
To: Discussion list for users of sipXecs software
 
Cc: Henry Kwan  
Sent: Thursday, October 18, 2012 6:29:01 PM
Subject: Re: [sipx-users] External calls cannot be transferred to voice mail
(sipXecs 4.4.0)


On Fri, Oct 19, 2012 at 3:26 AM, Tony Graziano
 wrote:
> Rather than use an old unsupportable version, produce a pcap from your
> firewall or produce a siptrace from sipx itself.
>
> I don't think your off the cuff observation is exactly right on targetm .
> Version 4.2 used its own media server while 4.4 uses FreeSWITCH and there
> are significant close changes.
>
> You could also indicate whether or not you followed a tutorial on how to
> properly configure pfsense and who the itsp is.
>

Additionally, if you could try scenario with 4.4 built from ISO,
without yum updating to latest, and report back, will help identifying
if issue in latest patches

Thanks
George



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

[Todd Hodgen]

ng 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 
 
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 al

Re: [sipx-users] 4.6: polycom and BLF

[Todd Hodgen]

I just setup what Nathaniel had asked about. 

 

Polycom 550 with ext 750 and DID assigned to it.Polycom VVX500 with Ext
701 assigned to it, and a speeddial button with *76750.  A separate ext. 750
presence button.  When 750 rings, press speeddial button it grabs it.

 

VVX has 12 buttons, so easy to have spare buttons.

 

SLA works too, personally, I don't have a need for either of these.

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Michael Picher
Sent: Tuesday, October 16, 2012 12:16 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] 4.6: polycom and BLF

 

How can you have BLF on a speed dial button that is programmed as *76750?
Why don't you add line 750 to the phone, have it silent ring and try to pick
it up that way (BLA)?

 

Mike

On Mon, Oct 15, 2012 at 10:19 PM, Todd Hodgen  wrote:

Wait.  I should clarify.  I had a speeddial button - *76750.  But, it didn't
have presence enabled.  Sorry.  I will test that scenario though.

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Josh Patten
Sent: Monday, October 15, 2012 5:48 PM


To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] 4.6: polycom and BLF

 

The VVX1500 doesn't seem to support directed call pickup from a BLF button.
Not sure why this is.

On Mon, Oct 15, 2012 at 6:41 PM, Nathaniel Watkins
 wrote:

Correct - on my VVX1500, I don't see status updates (I am subscribed to
presence).  So I am unable to simply press the 'flashing button' to answer a
ringing call.  *78 will pick up the call, but I cannot do it from the
speed dial button.

 

So in this scenario:

 

Polycom 650 calls Polycom 650

VVX 1500 doesn't see the status of either extension - pressing the speed
dial button of the extension being called simply initiates another call.

 

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Todd Hodgen
Sent: Monday, October 15, 2012 4:51 PM
To: 'Discussion list for users of sipXecs software'


Subject: Re: [sipx-users] 4.6: polycom and BLF

 

Explain the scenario.  Call ringing on a phone, and you want to do a
directed call pickup from a button that has a speeddial button configured?

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Nathaniel
Watkins
Sent: Monday, October 15, 2012 1:25 PM
To: Discussion list for users of sipXecs software; George Niculae
Subject: Re: [sipx-users] 4.6: polycom and BLF

 

I have always had problems when attempting to retrieve a call via the speed
dial button when using a VVX 1500.  Does this work now as well?

 

Nathaniel Watkins
CIO DoTCom (Department of Technology & Communication)
Garrett County Government/Board of Education
203 South 4th Street, Room 211
Oakland, MD  21550
Telephone: 301-334-5001
Fax: 301-334-7605
E-mail: nwatk...@garrettcounty.org

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Kumaran
Sent: Monday, October 15, 2012 2:26 AM
To: George Niculae
Cc: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] 4.6: polycom and BLF

 

I checked RLS which is working fine too...I can retrieve the call and
monitor the call through speed dial button(stp)

Regards,
Kumaran T 

On 10/15/2012 11:22 AM, George Niculae wrote: 

Thanks, actually I was asking for RLS

George

On Monday, October 15, 2012, Kumaran 
wrote:
> On 10/12/2012 7:00 PM, George Niculae wrote:
>>
>> 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
>>
> Hi George,
>I checked SLA in latest build in Polycom  which is working fine...I
retrieve the call from any shared line phone...
>
> Regards,
> Kumaran T
>
> 

 

 

  _  

This message and any files transmitted with it are intended only for the
individual(s) or entity named. If you are not the intended individual(s) or
entity named you are hereby notified that any disclosure, copying,
distribution or reliance upon its contents is strictly prohibited. If you
have received this in error, please notify the sender, delete the original,
and destroy all copies. Email transmissions cannot be guaranteed to be
secure or error-free as information could be intercepted, corrupted, lost,
destroyed, arrive late or incomplete, or contain viruses. Garrett County
Government therefore does not accept any liability for any errors or
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transmission.


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203 South Fourth Street, Courthouse, Oakland, Maryland 21550
www.garrettcounty.org


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Re: [sipx-users] 4.6: polycom and BLF

[Todd Hodgen]

Wait.  I should clarify.  I had a speeddial button - *76750.  But, it didn't
have presence enabled.  Sorry.  I will test that scenario though.

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Josh Patten
Sent: Monday, October 15, 2012 5:48 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] 4.6: polycom and BLF

 

The VVX1500 doesn't seem to support directed call pickup from a BLF button.
Not sure why this is.

On Mon, Oct 15, 2012 at 6:41 PM, Nathaniel Watkins
 wrote:

Correct - on my VVX1500, I don't see status updates (I am subscribed to
presence).  So I am unable to simply press the 'flashing button' to answer a
ringing call.  *78 will pick up the call, but I cannot do it from the
speed dial button.

 

So in this scenario:

 

Polycom 650 calls Polycom 650

VVX 1500 doesn't see the status of either extension - pressing the speed
dial button of the extension being called simply initiates another call.

 

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Todd Hodgen
Sent: Monday, October 15, 2012 4:51 PM
To: 'Discussion list for users of sipXecs software'


Subject: Re: [sipx-users] 4.6: polycom and BLF

 

Explain the scenario.  Call ringing on a phone, and you want to do a
directed call pickup from a button that has a speeddial button configured?

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Nathaniel
Watkins
Sent: Monday, October 15, 2012 1:25 PM
To: Discussion list for users of sipXecs software; George Niculae
Subject: Re: [sipx-users] 4.6: polycom and BLF

 

I have always had problems when attempting to retrieve a call via the speed
dial button when using a VVX 1500.  Does this work now as well?

 

Nathaniel Watkins
CIO DoTCom (Department of Technology & Communication)
Garrett County Government/Board of Education
203 South 4th Street, Room 211
Oakland, MD  21550
Telephone: 301-334-5001
Fax: 301-334-7605
E-mail: nwatk...@garrettcounty.org

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Kumaran
Sent: Monday, October 15, 2012 2:26 AM
To: George Niculae
Cc: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] 4.6: polycom and BLF

 

I checked RLS which is working fine too...I can retrieve the call and
monitor the call through speed dial button(stp)

Regards,
Kumaran T 

On 10/15/2012 11:22 AM, George Niculae wrote: 

Thanks, actually I was asking for RLS

George

On Monday, October 15, 2012, Kumaran 
wrote:
> On 10/12/2012 7:00 PM, George Niculae wrote:
>>
>> 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
>>
> Hi George,
>I checked SLA in latest build in Polycom  which is working fine...I
retrieve the call from any shared line phone...
>
> Regards,
> Kumaran T
>
> 

 

 

  _  

This message and any files transmitted with it are intended only for the
individual(s) or entity named. If you are not the intended individual(s) or
entity named you are hereby notified that any disclosure, copying,
distribution or reliance upon its contents is strictly prohibited. If you
have received this in error, please notify the sender, delete the original,
and destroy all copies. Email transmissions cannot be guaranteed to be
secure or error-free as information could be intercepted, corrupted, lost,
destroyed, arrive late or incomplete, or contain viruses. Garrett County
Government therefore does not accept any liability for any errors or
omissions in the contents of this message, which arise as a result of email
transmission.


Garrett County Government,
203 South Fourth Street, Courthouse, Oakland, Maryland 21550
www.garrettcounty.org


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-- 
Josh Patten
eZuce
Solutions Architect
O.978-296-1005 X2050 
M.979-574-5699
http://www.ezuce.com

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Re: [sipx-users] 4.6: polycom and BLF

[Todd Hodgen]

Explain the scenario.  Call ringing on a phone, and you want to do a
directed call pickup from a button that has a speeddial button configured?

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Nathaniel
Watkins
Sent: Monday, October 15, 2012 1:25 PM
To: Discussion list for users of sipXecs software; George Niculae
Subject: Re: [sipx-users] 4.6: polycom and BLF

 

I have always had problems when attempting to retrieve a call via the speed
dial button when using a VVX 1500.  Does this work now as well?

 

Nathaniel Watkins
CIO DoTCom (Department of Technology & Communication)
Garrett County Government/Board of Education
203 South 4th Street, Room 211
Oakland, MD  21550
Telephone: 301-334-5001
Fax: 301-334-7605
E-mail: nwatk...@garrettcounty.org

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Kumaran
Sent: Monday, October 15, 2012 2:26 AM
To: George Niculae
Cc: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] 4.6: polycom and BLF

 

I checked RLS which is working fine too...I can retrieve the call and
monitor the call through speed dial button(stp)

Regards,
Kumaran T 

On 10/15/2012 11:22 AM, George Niculae wrote: 

Thanks, actually I was asking for RLS

George

On Monday, October 15, 2012, Kumaran 
wrote:
> On 10/12/2012 7:00 PM, George Niculae wrote:
>>
>> 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
>>
> Hi George,
>I checked SLA in latest build in Polycom  which is working fine...I
retrieve the call from any shared line phone...
>
> Regards,
> Kumaran T
>
> 

 

 

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Re: [sipx-users] 4.6: polycom and BLF

[Todd Hodgen]

George, Wiki still seems to be down.  Do you have access to service it?

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of George Niculae
Sent: Sunday, October 14, 2012 10:52 PM
To: Kumaran
Cc: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] 4.6: polycom and BLF

 

Thanks, actually I was asking for RLS

George

On Monday, October 15, 2012, Kumaran 
wrote:
> On 10/12/2012 7:00 PM, George Niculae wrote:
>>
>> 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
>>
> Hi George,
>I checked SLA in latest build in Polycom  which is working fine...I
retrieve the call from any shared line phone...
>
> Regards,
> Kumaran T
>
> 

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Re: [sipx-users] 4.6: polycom and BLF

[Todd Hodgen]

I have 4.6 running with Polycom 550 and Polycom VVX500 and show presence on
each phone when the other is used.

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of George Niculae
Sent: Sunday, October 14, 2012 10:52 PM
To: Kumaran
Cc: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] 4.6: polycom and BLF

 

Thanks, actually I was asking for RLS

George

On Monday, October 15, 2012, Kumaran 
wrote:
> On 10/12/2012 7:00 PM, George Niculae wrote:
>>
>> 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
>>
> Hi George,
>I checked SLA in latest build in Polycom  which is working fine...I
retrieve the call from any shared line phone...
>
> Regards,
> Kumaran T
>
> 

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

[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

[sipx-users] Configuration Summary Page

[Todd Hodgen]

In the past, I've used a plugin to provide a User Summary Page for sipxecs.
It is needing to be rewritten, and I'm looking for comments as to whether it
should be included in the core software for all to have access to, or as a
plugin that would only be available if you drop it into your installation.

 

The summary page pulls configuration information for the system as a whole
onto one summary page, so you can look up IP addresses, gateway names,
number of users, number of groups, voicemail express code, paging codes,
etc. onto one page.  It also adds a small editor to the bottom of the page
for entering information about customization, troubleshooting steps taken,
etc.

 

Personally, I find it invaluable for my own use.I'm looking for comment
as to if people would like to see that in the system as a standard feature,
or if they prefer it be a plugin.

 

Additionally, I have created links to run logrotate and merge-logs from menu
options under the diagnostics menu.  These are handy if you can't access the
server remotely and need to rotate out the logs, run a scenario and then
capture it.   Merge-logs creates a download file link.  I can offer that as
a plugin for 4.4 now if you would like it, just email me and I'll provide
the plugin.  I'm hesitant to put it on the wiki, as there are a few items on
it that don't work well, but don't create issues.  

 

One last item - there is a siptapi application that works well for click to
call of outlook contacts with sipxecs.  I have created 32 bit and 64 bit msi
files for these that simplify the installation of them.  If interested, I'm
happy to share these.   The plugin I have provides links to these files so
they can be downloaded from the superadmin page.

 



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Re: [sipx-users] 4.6 voicemail hangup when press *

[Todd Hodgen]

No rush on my part - just pointing out what I find.

Do you need a Jira created for this?

-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of George Niculae
Sent: Friday, October 05, 2012 1:04 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] 4.6 voicemail hangup when press *

On Fri, Oct 5, 2012 at 9:26 AM, Todd Hodgen  wrote:
> In my testing, it was during the playing of the mailbox message.  
> Pressing the "*" should allow you to move to another voicemail box by 
> entering the extension and password of an alternative box.  In my 
> scenario, it terminated the call.
>

Thanks Todd, I reproduced the issue. Fix will be available shortly

George
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Re: [sipx-users] 4.6 voicemail hangup when press *

[Todd Hodgen]

In my testing, it was during the playing of the mailbox message.  Pressing
the "*" should allow you to move to another voicemail box by entering the
extension and password of an alternative box.  In my scenario, it terminated
the call.

-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Kumaran
Sent: Thursday, October 04, 2012 10:29 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] 4.6 voicemail hangup when press *

On 10/5/2012 4:21 AM, George Niculae wrote:
> On Thu, Oct 4, 2012 at 10:18 PM, George Niculae  wrote:
>> On Thu, Oct 4, 2012 at 10:01 PM, Todd Hodgen
wrote:
>>> Hi George,   thanks for the comments.  Not related to MWI.
>>>
>>> Scenario - from extension 701 - dialed 8750 to go directly to ext 
>>> 750 voicemail.
>>> Voicemail answered prompting for password.  I press * and call drops.
>>> 00:sipxivr:"depositVoicemail Collected digits=*"
>>> "2012-10-04T17:10:16.922000Z":250:sipXivr:ERR:rage.ragesip.com:Threa
>>> d-53:000
>>> 0:sipxivr:"SipXivr::run"
>>> java.lang.NullPointerException
>>>  at
>>> org.sipfoundry.voicemail.mailbox.AbstractMailboxManager.deleteTempMe
>>> ssage(Ab
>>> stractMailboxManager.java:84)
>>>  at org.sipfoundry.voicemail.Deposit.runAction(Deposit.java:225)
>>>  at
org.sipfoundry.voicemail.VoiceMail.voicemail(VoiceMail.java:53)
>>>  at org.sipfoundry.voicemail.VoiceMail.run(VoiceMail.java:42)
>>>  at org.sipfoundry.sipxivr.SipxIvrApp.run(SipxIvrApp.java:31)
>>>  at
org.sipfoundry.sipxivr.SipXivr.runEslRequest(SipXivr.java:61)
>>>  at
>>> org.sipfoundry.sipxivr.eslrequest.EslRequestScopeRunnable.run(EslReq
>>> uestScop
>>> eRunnable.java:24)
>>>  at java.lang.Thread.run(Thread.java:679)
>>>
>> Looks like the issue reported by Tony today. Let me reinstall a fresh 
>> system and get back to you
>>
> Hm, I cannot reproduce this problem, are you testing 32 or 64bit RPMs?
>
> George
I can reproduce the issue GeorgeWhile depositing VM just press * before
the beep tone,the call will be ended...If we press* after the beep tone then
its cancel the current request(will not disconnect) So I think * is for
cancelling the current request in VM and for AA it will repeat the prompt..

Regards,
Kumaran T
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Re: [sipx-users] 4.6 voicemail hangup when press *

[Todd Hodgen]

64 bit RPM.

-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of George Niculae
Sent: Thursday, October 04, 2012 3:51 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] 4.6 voicemail hangup when press *

On Thu, Oct 4, 2012 at 10:18 PM, George Niculae  wrote:
> On Thu, Oct 4, 2012 at 10:01 PM, Todd Hodgen  wrote:
>> Hi George,   thanks for the comments.  Not related to MWI.
>>
>> Scenario - from extension 701 - dialed 8750 to go directly to ext 750 
>> voicemail.
>> Voicemail answered prompting for password.  I press * and call drops.
>> 00:sipxivr:"depositVoicemail Collected digits=*"
>> "2012-10-04T17:10:16.922000Z":250:sipXivr:ERR:rage.ragesip.com:Thread
>> -53:000
>> 0:sipxivr:"SipXivr::run"
>> java.lang.NullPointerException
>> at
>> org.sipfoundry.voicemail.mailbox.AbstractMailboxManager.deleteTempMes
>> sage(Ab
>> stractMailboxManager.java:84)
>> at org.sipfoundry.voicemail.Deposit.runAction(Deposit.java:225)
>> at
org.sipfoundry.voicemail.VoiceMail.voicemail(VoiceMail.java:53)
>> at org.sipfoundry.voicemail.VoiceMail.run(VoiceMail.java:42)
>> at org.sipfoundry.sipxivr.SipxIvrApp.run(SipxIvrApp.java:31)
>> at org.sipfoundry.sipxivr.SipXivr.runEslRequest(SipXivr.java:61)
>> at
>> org.sipfoundry.sipxivr.eslrequest.EslRequestScopeRunnable.run(EslRequ
>> estScop
>> eRunnable.java:24)
>> at java.lang.Thread.run(Thread.java:679)
>>
>
> Looks like the issue reported by Tony today. Let me reinstall a fresh 
> system and get back to you
>

Hm, I cannot reproduce this problem, are you testing 32 or 64bit RPMs?

George
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Re: [sipx-users] 4.6 voicemail hangup when press *

[Todd Hodgen]

Hi George,   thanks for the comments.  Not related to MWI.

Scenario - from extension 701 - dialed 8750 to go directly to ext 750
voicemail.
Voicemail answered prompting for password.  I press * and call drops.
Relevant logs are below - full log attached.

Maybe I have to enter "*" as an option in the personal directory?  I don't
believe this was the behavior in prior releases, or I have a configuration
issue to work through.  Fresh install with yum update yesterday.  4.6.0.
2012-10-02EDT17:10:59 ip-10-72-43-110.ec2.internal

"2012-10-04T17:09:58.06Z":234:sipXivr:INFO:rage.ragesip.com:Thread-50:00
00:sipxivr:"SipXivr::run Accepting call-id
400a9852-4ed27483-8e0d6334@192.168.5.106 from 7...@ragesip.com to
i...@vm.rage.ragesip.com"
"2012-10-04T17:09:58.061000Z":235:sipXivr:INFO:rage.ragesip.com:Thread-50:00
00:sipxivr:"SipXivr::run Bridging the call"
"2012-10-04T17:09:58.086000Z":236:sipXivr:INFO:rage.ragesip.com:Thread-51:00
00:sipxivr:"SipXivr::run Accepting call-id
2a8ff9b0-88e9-1230-968d-00238bbdf098 from 701@192.168.5.252 to
i...@vm.rage.ragesip.com"
"2012-10-04T17:09:59.118000Z":237:sipXivr:INFO:rage.ragesip.com:Thread-51:00
00:sipxivr:"Starting voicemail for mailbox \"750\" action=\"deposit"
"2012-10-04T17:09:59.119000Z":238:sipXivr:INFO:rage.ragesip.com:Thread-51:00
00:sipxivr:"Mailbox org.sipfoundry.commons.userdb.User@6ce7ce4c Deposit
Voicemail from \"Jason Hodgen\" "
"2012-10-04T17:10:03.82Z":239:sipXivr:INFO:rage.ragesip.com:Thread-51:00
00:sipxivr:"Collect::start 1 100/0/0 mask 1234567890ABCD#*i"
"2012-10-04T17:10:03.821000Z":240:sipXivr:INFO:rage.ragesip.com:Thread-51:00
00:sipxivr:"depositVoicemail Collected digits=#"
"2012-10-04T17:10:05.719000Z":241:sipXivr:INFO:rage.ragesip.com:Thread-51:00
00:sipxivr:"FSESI::invoke throw DisconnectException"
"2012-10-04T17:10:06.72Z":242:sipXivr:INFO:rage.ragesip.com:Thread-51:00
00:sipxivr:"Ending voicemail"
"2012-10-04T17:10:12.393000Z":243:sipXivr:INFO:rage.ragesip.com:Thread-52:00
00:sipxivr:"SipXivr::run Accepting call-id
90650762-de2c0b93-60a2244@192.168.5.106 from 7...@ragesip.com to
i...@vm.rage.ragesip.com"
"2012-10-04T17:10:12.393000Z":244:sipXivr:INFO:rage.ragesip.com:Thread-52:00
00:sipxivr:"SipXivr::run Bridging the call"
"2012-10-04T17:10:12.418000Z":245:sipXivr:INFO:rage.ragesip.com:Thread-53:00
00:sipxivr:"SipXivr::run Accepting call-id
331b0449-88e9-1230-968d-00238bbdf098 from 701@192.168.5.252 to
i...@vm.rage.ragesip.com"
"2012-10-04T17:10:13.459000Z":246:sipXivr:INFO:rage.ragesip.com:Thread-53:00
00:sipxivr:"Starting voicemail for mailbox \"750\" action=\"deposit"
"2012-10-04T17:10:13.46Z":247:sipXivr:INFO:rage.ragesip.com:Thread-53:00
00:sipxivr:"Mailbox org.sipfoundry.commons.userdb.User@52287b58 Deposit
Voicemail from \"Jason Hodgen\" "
"2012-10-04T17:10:15.921000Z":248:sipXivr:INFO:rage.ragesip.com:Thread-53:00
00:sipxivr:"Collect::start 1 100/0/0 mask 1234567890ABCD#*i"
"2012-10-04T17:10:15.922000Z":249:sipXivr:INFO:rage.ragesip.com:Thread-53:00
00:sipxivr:"depositVoicemail Collected digits=*"
"2012-10-04T17:10:16.922000Z":250:sipXivr:ERR:rage.ragesip.com:Thread-53:000
0:sipxivr:"SipXivr::run"
java.lang.NullPointerException
at
org.sipfoundry.voicemail.mailbox.AbstractMailboxManager.deleteTempMessage(Ab
stractMailboxManager.java:84)
at org.sipfoundry.voicemail.Deposit.runAction(Deposit.java:225)
at org.sipfoundry.voicemail.VoiceMail.voicemail(VoiceMail.java:53)
at org.sipfoundry.voicemail.VoiceMail.run(VoiceMail.java:42)
at org.sipfoundry.sipxivr.SipxIvrApp.run(SipxIvrApp.java:31)
at org.sipfoundry.sipxivr.SipXivr.runEslRequest(SipXivr.java:61)
at
org.sipfoundry.sipxivr.eslrequest.EslRequestScopeRunnable.run(EslRequestScop
eRunnable.java:24)
at java.lang.Thread.run(Thread.java:679)


-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of George Niculae
Sent: Thursday, October 04, 2012 11:06 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] 4.6 voicemail hangup when press *

On Thu, Oct 4, 2012 at 8:16 PM, Todd Hodgen  wrote:
> I was trying to get to my voicemail from another phone to test 
> something and when I pressed *, rather than allowing me to enter my
alternative extension
> and password it hung up on me.   Using Polycom 550 and vvx500 same
results.
>
>
>
> I'd like to confirm if others see the same thing on 4.6?
>
>

Re: [sipx-users] SIPXEC crashes after install

[Todd Hodgen]

I would suggest looking at the hardware as well.  I run sipxecs 4.4 as a
demo system on a laptop with smaller processor and 2gb memory without issue.
It is very stable, and I suspect your issue is more related to a hardware
issue.

 

Additionally, I have it running on at least a dozen machines with dual core,
and 2gb memory with no issues today.  I think you will find that to be a
very stable environment.

 

The demo laptop was used at COLAB 2012 to demonstrate Voice Operator Panel
and Polycom phones.  You will see it running again with 4.6 at COLAB 2013.
If you haven't registered for Colab - now is a great time to take advantage
of the registration discounts!  http://www.sipfoundry.org/sipx-colab

 

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Trevor L Benson
Sent: Thursday, October 04, 2012 8:02 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] SIPXEC crashes after install

 

James,

 

  These type of problems generally point to hardware issues or corrupted OS
install possibly exacerbated by sipXecs starting to load up memory into Java
and eat up CPU cycles.  Did you happen to check the MD5 sum of the ISO
before burning or using it for installation?  Did you take the time to test
the CD during install to ensure it was clean?  If you already did these
things then I would definitely suggest memtest86+ and even a burn in /
stress test software to ensure something besides memory (like CPU) isn't the
cause.  Maybe Douglas or another has more experience with sipXecs crashing
the entire server, but I personally have not seen that happen in a stable
release of sipXecs from 4.0 through 4.4.

 

 

Thanks,

Trevor Benson, Network Engineer

A1 Networks

 

 

 

 

On Oct 4, 2012, at 6:19 AM, James M. Jones  wrote:





Good Day,

 

I'm currently working with version 4.4.4.0. I've installed it on an older
Dell Dimension C521 with an AMD Athlon Processor (64X2 Dual Core, 1.9GHZ)
and 3GB of RAM.

 

The installation works fine and upon completion, I can access the GUI and
start the initial configuration, but shortly after, the system will crash.
I've left the system at the initial configuration after installation (Only
changed the superadmin password), but it will still crash. If I'm directly
connected to the box, the screen becomes unreadable or goes blank. Upon
restart, the system freezes and the screen goes blank when it tries to load
udep.

 

I've reinstalled it 4 times now without errors. I've just recently started
working with SIPXEC and any suggestions are greatly appreciated.

 

 

 

Thank You

 

 

James M. Jones

Project Manager

InSysPro, LLC

M. 954-557-2771

ja...@insyspro.com



 

 

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[sipx-users] 4.6 voicemail hangup when press *

[Todd Hodgen]

I was trying to get to my voicemail from another phone to test something and
when I pressed *, rather than allowing me to enter my alternative extension
and password it hung up on me.   Using Polycom 550 and vvx500 same results.

 

I'd like to confirm if others see the same thing on 4.6?

 

What I was testing was voicemails being very low volume on one phone.   On
vvx500 volume is good, on Polycom 550 volume is low.  I'll dig into that
further.

 

If someone can confirm the pressing of * when entering voicemail, I'd
appreciate it.

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Re: [sipx-users] Lastest Patches and Alias

[Todd Hodgen]

No expert here, but from what I see, the doctype is missing the \ just
before the quote.  Guess it isn't the problem, when I add it, it returns
same error with \\ instead of none on the error.

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Todd Hodgen
Sent: Monday, October 01, 2012 5:47 PM
To: 'Discussion list for users of sipXecs software'
Subject: Re: [sipx-users] Lastest Patches and Alias

 

George,  seeing an error - Config error: file contains no section headers 

File: file:/etc/yum.repos.d/sipxecs.repo
 , line: 1

 

Any ideas?

Did a yum clean all with not improvements.

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of George Niculae
Sent: Saturday, September 29, 2012 3:12 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Lastest Patches and Alias

 

Yes, just sipxregistry and sipxconfig rpms

George
On Sep 30, 2012 1:03 AM, "Todd Hodgen"  wrote:
>
> Thanks George. 
>
>  
>
> so from this directory I would just do a yum update of those two files
then?
>
>  
>
> From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of George Niculae
> Sent: Saturday, September 29, 2012 2:57 PM
>
> To: Discussion list for users of sipXecs software
> Subject: Re: [sipx-users] Lastest Patches and Alias
>
>  
>
> Ah, you should try from here
>
> http://download.sipfoundry.org/pub/sipXecs-stage/4.4.0/CentOS_5/i386/
>
> George
> On Sep 30, 2012 12:18 AM, "Todd Hodgen"  wrote:
> >
> > This might be a problem, as I believe the site is a 32-bit installation.
> >
> > -Original Message-
> > From: sipx-users-boun...@list.sipfoundry.org
> > [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of George
Niculae
> > Sent: Saturday, September 29, 2012 12:49 PM
> > To: Discussion list for users of sipXecs software
> > Subject: Re: [sipx-users] Lastest Patches and Alias
> >
> > Todd,
> >
> > you should yum update sipxregistry and sipxconfig from here for the
moment,
> > they contain changes reverted:
> >
> > http://download.sipfoundry.org/pub/sipXecs-stage/4.4.0/CentOS_5/x86_64/
> >
> > George
> >
> > On Sat, Sep 29, 2012 at 10:37 PM, Todd Hodgen 
wrote:
> > > Sorry I don't.
> > >
> > > This is unfortunate, as it creates service issues for this site.
> > >
> > > This brings up a concern with this patching process that is being
used.
> > > Yum update pulls in all of the patches, and has the potential of
> > > bringing in patches that are not fully vetted and regression tested to
> > working sites.
> > >
> > > I wish there was a different method of ensuring you know exactly what
> > > patches you are putting into a system, along with a method of removing
> > > them should there be an issue.
> > >
> > >
> > >
> > > -Original Message-
> > > From: sipx-users-boun...@list.sipfoundry.org
> > > [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of George
> > > Niculae
> > > Sent: Saturday, September 29, 2012 3:43 AM
> > > To: Discussion list for users of sipXecs software
> > > Subject: Re: [sipx-users] Lastest Patches and Alias
> > >
> > > On Sat, Sep 29, 2012 at 1:33 PM, Todd Hodgen 
wrote:
> > >> I haven't dug into it yet, but will in morning.  Last night I did a
> > >> Yum Update to a site that was running on 4.4, with a Patton Gateway
> > >> to
> > > PRI.
> > >> Everything was running well, but the certificate was about to expire.
> > >>
> > >>
> > >>
> > >> Here are the steps I took -
> > >>
> > >> Stopped sipxecs
> > >>
> > >> Yum Update to get the latest patches
> > >>
> > >> Started sipxecs
> > >>
> > >> Logged in and Sent profiles for the server
> > >>
> > >>
> > >>
> > >> Did backup of system
> > >>
> > >>
> > >>
> > >> Stopped sipxecs
> > >>
> > >> Followed instruction from Wiki for creating a new Certificate - as
> > >> this one was going to run out 10/8/2012.
> > >>
> > >> Started sipxecs
> > >>
> > >> Logged in and sent profiles for the server.
> > >>
> > >>
> > >>
> > >> I tried to restart all of the phones, but only some did a res

Re: [sipx-users] Lastest Patches and Alias

[Todd Hodgen]

George,  seeing an error - Config error: file contains no section headers 

File: file:/etc/yum.repos.d/sipxecs.repo
 , line: 1

 

Any ideas?

Did a yum clean all with not improvements.

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of George Niculae
Sent: Saturday, September 29, 2012 3:12 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Lastest Patches and Alias

 

Yes, just sipxregistry and sipxconfig rpms

George
On Sep 30, 2012 1:03 AM, "Todd Hodgen"  wrote:
>
> Thanks George. 
>
>  
>
> so from this directory I would just do a yum update of those two files
then?
>
>  
>
> From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of George Niculae
> Sent: Saturday, September 29, 2012 2:57 PM
>
> To: Discussion list for users of sipXecs software
> Subject: Re: [sipx-users] Lastest Patches and Alias
>
>  
>
> Ah, you should try from here
>
> http://download.sipfoundry.org/pub/sipXecs-stage/4.4.0/CentOS_5/i386/
>
> George
> On Sep 30, 2012 12:18 AM, "Todd Hodgen"  wrote:
> >
> > This might be a problem, as I believe the site is a 32-bit installation.
> >
> > -Original Message-
> > From: sipx-users-boun...@list.sipfoundry.org
> > [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of George
Niculae
> > Sent: Saturday, September 29, 2012 12:49 PM
> > To: Discussion list for users of sipXecs software
> > Subject: Re: [sipx-users] Lastest Patches and Alias
> >
> > Todd,
> >
> > you should yum update sipxregistry and sipxconfig from here for the
moment,
> > they contain changes reverted:
> >
> > http://download.sipfoundry.org/pub/sipXecs-stage/4.4.0/CentOS_5/x86_64/
> >
> > George
> >
> > On Sat, Sep 29, 2012 at 10:37 PM, Todd Hodgen 
wrote:
> > > Sorry I don't.
> > >
> > > This is unfortunate, as it creates service issues for this site.
> > >
> > > This brings up a concern with this patching process that is being
used.
> > > Yum update pulls in all of the patches, and has the potential of
> > > bringing in patches that are not fully vetted and regression tested to
> > working sites.
> > >
> > > I wish there was a different method of ensuring you know exactly what
> > > patches you are putting into a system, along with a method of removing
> > > them should there be an issue.
> > >
> > >
> > >
> > > -Original Message-----
> > > From: sipx-users-boun...@list.sipfoundry.org
> > > [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of George
> > > Niculae
> > > Sent: Saturday, September 29, 2012 3:43 AM
> > > To: Discussion list for users of sipXecs software
> > > Subject: Re: [sipx-users] Lastest Patches and Alias
> > >
> > > On Sat, Sep 29, 2012 at 1:33 PM, Todd Hodgen 
wrote:
> > >> I haven't dug into it yet, but will in morning.  Last night I did a
> > >> Yum Update to a site that was running on 4.4, with a Patton Gateway
> > >> to
> > > PRI.
> > >> Everything was running well, but the certificate was about to expire.
> > >>
> > >>
> > >>
> > >> Here are the steps I took -
> > >>
> > >> Stopped sipxecs
> > >>
> > >> Yum Update to get the latest patches
> > >>
> > >> Started sipxecs
> > >>
> > >> Logged in and Sent profiles for the server
> > >>
> > >>
> > >>
> > >> Did backup of system
> > >>
> > >>
> > >>
> > >> Stopped sipxecs
> > >>
> > >> Followed instruction from Wiki for creating a new Certificate - as
> > >> this one was going to run out 10/8/2012.
> > >>
> > >> Started sipxecs
> > >>
> > >> Logged in and sent profiles for the server.
> > >>
> > >>
> > >>
> > >> I tried to restart all of the phones, but only some did a restart.
> > >>
> > >>
> > >>
> > >> Otherwise, everything okay.
> > >>
> > >>
> > >>
> > >> Today, I discovered calls to the auto-attendant are processing fine.
> > >> However, calls to DID numbers - which are all aliases on this system
> > >> are not.  Thought maybe related to certificates, so I factory reset
> > >> two of the phones with t

Re: [sipx-users] Lastest Patches and Alias

[Todd Hodgen]

Thanks George.  

 

so from this directory I would just do a yum update of those two files then?

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of George Niculae
Sent: Saturday, September 29, 2012 2:57 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Lastest Patches and Alias

 

Ah, you should try from here

http://download.sipfoundry.org/pub/sipXecs-stage/4.4.0/CentOS_5/i386/

George
On Sep 30, 2012 12:18 AM, "Todd Hodgen"  wrote:
>
> This might be a problem, as I believe the site is a 32-bit installation.
>
> -Original Message-
> From: sipx-users-boun...@list.sipfoundry.org
> [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of George
Niculae
> Sent: Saturday, September 29, 2012 12:49 PM
> To: Discussion list for users of sipXecs software
> Subject: Re: [sipx-users] Lastest Patches and Alias
>
> Todd,
>
> you should yum update sipxregistry and sipxconfig from here for the
moment,
> they contain changes reverted:
>
> http://download.sipfoundry.org/pub/sipXecs-stage/4.4.0/CentOS_5/x86_64/
>
> George
>
> On Sat, Sep 29, 2012 at 10:37 PM, Todd Hodgen 
wrote:
> > Sorry I don't.
> >
> > This is unfortunate, as it creates service issues for this site.
> >
> > This brings up a concern with this patching process that is being used.
> > Yum update pulls in all of the patches, and has the potential of
> > bringing in patches that are not fully vetted and regression tested to
> working sites.
> >
> > I wish there was a different method of ensuring you know exactly what
> > patches you are putting into a system, along with a method of removing
> > them should there be an issue.
> >
> >
> >
> > -Original Message-
> > From: sipx-users-boun...@list.sipfoundry.org
> > [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of George
> > Niculae
> > Sent: Saturday, September 29, 2012 3:43 AM
> > To: Discussion list for users of sipXecs software
> > Subject: Re: [sipx-users] Lastest Patches and Alias
> >
> > On Sat, Sep 29, 2012 at 1:33 PM, Todd Hodgen 
wrote:
> >> I haven't dug into it yet, but will in morning.  Last night I did a
> >> Yum Update to a site that was running on 4.4, with a Patton Gateway
> >> to
> > PRI.
> >> Everything was running well, but the certificate was about to expire.
> >>
> >>
> >>
> >> Here are the steps I took -
> >>
> >> Stopped sipxecs
> >>
> >> Yum Update to get the latest patches
> >>
> >> Started sipxecs
> >>
> >> Logged in and Sent profiles for the server
> >>
> >>
> >>
> >> Did backup of system
> >>
> >>
> >>
> >> Stopped sipxecs
> >>
> >> Followed instruction from Wiki for creating a new Certificate - as
> >> this one was going to run out 10/8/2012.
> >>
> >> Started sipxecs
> >>
> >> Logged in and sent profiles for the server.
> >>
> >>
> >>
> >> I tried to restart all of the phones, but only some did a restart.
> >>
> >>
> >>
> >> Otherwise, everything okay.
> >>
> >>
> >>
> >> Today, I discovered calls to the auto-attendant are processing fine.
> >> However, calls to DID numbers - which are all aliases on this system
> >> are not.  Thought maybe related to certificates, so I factory reset
> >> two of the phones with this issue, no joy.
> >>
> >>
> >>
> >> Have I missed an issue with the latest patches for 4.4, or is anyone
> >> else seeing a similar issue on 4.4 with latest patches.
> >>
> >>
> >>
> >> Yes, I know, hard to tell without a trace - I'll get it in the
> >> morning, but curious if anyone else has discovered an issue like this
> >> - possibly a new configuration items I am not aware of?
> >>
> >
> > Todd,
> >
> > there could be a problem with latest update19 - we noticed same
> > problem last night. Until we figure it out, do you have any mean to
> > downgrade sipregistrar /config to patch 18?
> >
> > George
> > ___
> > 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 mailing list
> sipx-users@list.sipfoundry.org
> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>
> ___
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> sipx-users@list.sipfoundry.org
> List Archive: http://list.sipfoundry.org/archive/sipx-users/

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Re: [sipx-users] Lastest Patches and Alias

[Todd Hodgen]

This might be a problem, as I believe the site is a 32-bit installation.

-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of George Niculae
Sent: Saturday, September 29, 2012 12:49 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Lastest Patches and Alias

Todd,

you should yum update sipxregistry and sipxconfig from here for the moment,
they contain changes reverted:

http://download.sipfoundry.org/pub/sipXecs-stage/4.4.0/CentOS_5/x86_64/

George

On Sat, Sep 29, 2012 at 10:37 PM, Todd Hodgen  wrote:
> Sorry I don't.
>
> This is unfortunate, as it creates service issues for this site.
>
> This brings up a concern with this patching process that is being used.
> Yum update pulls in all of the patches, and has the potential of 
> bringing in patches that are not fully vetted and regression tested to
working sites.
>
> I wish there was a different method of ensuring you know exactly what 
> patches you are putting into a system, along with a method of removing 
> them should there be an issue.
>
>
>
> -Original Message-
> From: sipx-users-boun...@list.sipfoundry.org
> [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of George 
> Niculae
> Sent: Saturday, September 29, 2012 3:43 AM
> To: Discussion list for users of sipXecs software
> Subject: Re: [sipx-users] Lastest Patches and Alias
>
> On Sat, Sep 29, 2012 at 1:33 PM, Todd Hodgen  wrote:
>> I haven't dug into it yet, but will in morning.  Last night I did a 
>> Yum Update to a site that was running on 4.4, with a Patton Gateway 
>> to
> PRI.
>> Everything was running well, but the certificate was about to expire.
>>
>>
>>
>> Here are the steps I took -
>>
>> Stopped sipxecs
>>
>> Yum Update to get the latest patches
>>
>> Started sipxecs
>>
>> Logged in and Sent profiles for the server
>>
>>
>>
>> Did backup of system
>>
>>
>>
>> Stopped sipxecs
>>
>> Followed instruction from Wiki for creating a new Certificate - as 
>> this one was going to run out 10/8/2012.
>>
>> Started sipxecs
>>
>> Logged in and sent profiles for the server.
>>
>>
>>
>> I tried to restart all of the phones, but only some did a restart.
>>
>>
>>
>> Otherwise, everything okay.
>>
>>
>>
>> Today, I discovered calls to the auto-attendant are processing fine.
>> However, calls to DID numbers - which are all aliases on this system 
>> are not.  Thought maybe related to certificates, so I factory reset 
>> two of the phones with this issue, no joy.
>>
>>
>>
>> Have I missed an issue with the latest patches for 4.4, or is anyone 
>> else seeing a similar issue on 4.4 with latest patches.
>>
>>
>>
>> Yes, I know, hard to tell without a trace - I'll get it in the 
>> morning, but curious if anyone else has discovered an issue like this
>> - possibly a new configuration items I am not aware of?
>>
>
> Todd,
>
> there could be a problem with latest update19 - we noticed same 
> problem last night. Until we figure it out, do you have any mean to 
> downgrade sipregistrar /config to patch 18?
>
> George
> ___
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> sipx-users@list.sipfoundry.org
> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>
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Re: [sipx-users] Lastest Patches and Alias

[Todd Hodgen]

Thanks George, I'll give it a try.

-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of George Niculae
Sent: Saturday, September 29, 2012 12:49 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Lastest Patches and Alias

Todd,

you should yum update sipxregistry and sipxconfig from here for the moment,
they contain changes reverted:

http://download.sipfoundry.org/pub/sipXecs-stage/4.4.0/CentOS_5/x86_64/

George

On Sat, Sep 29, 2012 at 10:37 PM, Todd Hodgen  wrote:
> Sorry I don't.
>
> This is unfortunate, as it creates service issues for this site.
>
> This brings up a concern with this patching process that is being used.
> Yum update pulls in all of the patches, and has the potential of 
> bringing in patches that are not fully vetted and regression tested to
working sites.
>
> I wish there was a different method of ensuring you know exactly what 
> patches you are putting into a system, along with a method of removing 
> them should there be an issue.
>
>
>
> -Original Message-
> From: sipx-users-boun...@list.sipfoundry.org
> [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of George 
> Niculae
> Sent: Saturday, September 29, 2012 3:43 AM
> To: Discussion list for users of sipXecs software
> Subject: Re: [sipx-users] Lastest Patches and Alias
>
> On Sat, Sep 29, 2012 at 1:33 PM, Todd Hodgen  wrote:
>> I haven't dug into it yet, but will in morning.  Last night I did a 
>> Yum Update to a site that was running on 4.4, with a Patton Gateway 
>> to
> PRI.
>> Everything was running well, but the certificate was about to expire.
>>
>>
>>
>> Here are the steps I took -
>>
>> Stopped sipxecs
>>
>> Yum Update to get the latest patches
>>
>> Started sipxecs
>>
>> Logged in and Sent profiles for the server
>>
>>
>>
>> Did backup of system
>>
>>
>>
>> Stopped sipxecs
>>
>> Followed instruction from Wiki for creating a new Certificate - as 
>> this one was going to run out 10/8/2012.
>>
>> Started sipxecs
>>
>> Logged in and sent profiles for the server.
>>
>>
>>
>> I tried to restart all of the phones, but only some did a restart.
>>
>>
>>
>> Otherwise, everything okay.
>>
>>
>>
>> Today, I discovered calls to the auto-attendant are processing fine.
>> However, calls to DID numbers - which are all aliases on this system 
>> are not.  Thought maybe related to certificates, so I factory reset 
>> two of the phones with this issue, no joy.
>>
>>
>>
>> Have I missed an issue with the latest patches for 4.4, or is anyone 
>> else seeing a similar issue on 4.4 with latest patches.
>>
>>
>>
>> Yes, I know, hard to tell without a trace - I'll get it in the 
>> morning, but curious if anyone else has discovered an issue like this
>> - possibly a new configuration items I am not aware of?
>>
>
> Todd,
>
> there could be a problem with latest update19 - we noticed same 
> problem last night. Until we figure it out, do you have any mean to 
> downgrade sipregistrar /config to patch 18?
>
> George
> ___
> sipx-users mailing list
> sipx-users@list.sipfoundry.org
> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>
> ___
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> sipx-users@list.sipfoundry.org
> List Archive: http://list.sipfoundry.org/archive/sipx-users/
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Re: [sipx-users] Lastest Patches and Alias

[Todd Hodgen]

Sorry I don't.

This is unfortunate, as it creates service issues for this site.

This brings up a concern with this patching process that is being used.
Yum update pulls in all of the patches, and has the potential of bringing in
patches that are not fully vetted and regression tested to working sites.

I wish there was a different method of ensuring you know exactly what
patches you are putting into a system, along with a method of removing them
should there be an issue.



-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of George Niculae
Sent: Saturday, September 29, 2012 3:43 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Lastest Patches and Alias

On Sat, Sep 29, 2012 at 1:33 PM, Todd Hodgen  wrote:
> I haven't dug into it yet, but will in morning.  Last night I did a 
> Yum Update to a site that was running on 4.4, with a Patton Gateway to
PRI.
> Everything was running well, but the certificate was about to expire.
>
>
>
> Here are the steps I took -
>
> Stopped sipxecs
>
> Yum Update to get the latest patches
>
> Started sipxecs
>
> Logged in and Sent profiles for the server
>
>
>
> Did backup of system
>
>
>
> Stopped sipxecs
>
> Followed instruction from Wiki for creating a new Certificate - as 
> this one was going to run out 10/8/2012.
>
> Started sipxecs
>
> Logged in and sent profiles for the server.
>
>
>
> I tried to restart all of the phones, but only some did a restart.
>
>
>
> Otherwise, everything okay.
>
>
>
> Today, I discovered calls to the auto-attendant are processing fine.
> However, calls to DID numbers - which are all aliases on this system 
> are not.  Thought maybe related to certificates, so I factory reset 
> two of the phones with this issue, no joy.
>
>
>
> Have I missed an issue with the latest patches for 4.4, or is anyone 
> else seeing a similar issue on 4.4 with latest patches.
>
>
>
> Yes, I know, hard to tell without a trace - I'll get it in the 
> morning, but curious if anyone else has discovered an issue like this 
> - possibly a new configuration items I am not aware of?
>

Todd,

there could be a problem with latest update19 - we noticed same problem last
night. Until we figure it out, do you have any mean to downgrade
sipregistrar /config to patch 18?

George
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[sipx-users] Lastest Patches and Alias

[Todd Hodgen]

I haven't dug into it yet, but will in morning.  Last night I did a Yum
Update to a site that was running on 4.4, with a Patton Gateway to PRI.
Everything was running well, but the certificate was about to expire.

 

Here are the steps I took - 

Stopped sipxecs

Yum Update to get the latest patches

Started sipxecs

Logged in and Sent profiles for the server

 

Did backup of system

 

Stopped sipxecs

Followed instruction from Wiki for creating a new Certificate - as this one
was going to run out 10/8/2012.

Started sipxecs

Logged in and sent profiles for the server.

 

I tried to restart all of the phones, but only some did a restart.

 

Otherwise, everything okay.

 

Today, I discovered calls to the auto-attendant are processing fine.
However, calls to DID numbers - which are all aliases on this system are
not.  Thought maybe related to certificates, so I factory reset two of the
phones with this issue, no joy.

 

Have I missed an issue with the latest patches for 4.4, or is anyone else
seeing a similar issue on 4.4 with latest patches.

 

Yes, I know, hard to tell without a trace - I'll get it in the morning, but
curious if anyone else has discovered an issue like this - possibly a new
configuration items I am not aware of?

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Re: [sipx-users] Remove operator/option zero from voicemail

[Todd Hodgen]

In the past I've removed that.  I think its a separate wave at the end or there 
was an alternative one to use in the directory.  I didn't have to record it
Sent from my twiddling thumbs.

niklas rehnberg  wrote:

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Re: [sipx-users] Incorrect SDP from sipX after xfer completion

[Todd Hodgen]

Jeff, I think the confusion is that some sip components are on the
192.168.53.x network, while others are on 172.21.201.x network.  In your
description below, you state "which routes", which is indicating there is a
router on the network, and the IP addressing would indicate they are on
different networks.  A router is required to route from one network to the
other network.   

 

NAT would be associated with a router - hence the reason why the questions
around where the router is.

 

one Polycom extension 1821 at 172.21.201.39 calls another, 7821, at
172.21.201.60. 7821 does an attended transfer to 2164550549, which routes
through the Adtran gateway at 192.168.53.11. The sipX system itself is at
192.168.54.46.

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Jeff Pyle
Sent: Tuesday, September 25, 2012 10:35 AM
To: Joegen Baclor
Cc: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Incorrect SDP from sipX after xfer completion

 

Joegen,

 

I fully understand my network so I don't see where the problem is.  :)

 

All the details, including an adequate topology description, are now in
XX-10464  .




- Jeff

 

On Tue, Sep 25, 2012 at 12:57 PM, Joegen Baclor  wrote:

Jeff,

I might be missing something but the SDP you pasted



o=- 1348230370 1348230371 IN IP4 172.21.201.60
s=Polycom IP Phone
c=IN IP4 172.21.201.60
t=0 0
m=audio 30248 RTP/AVP 9 0 8 18 101
c=IN IP4 192.168.54.46
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=x-sipx-ntap:X192.168.54.46-[PUBLIC-IP];54



172.21.201.60 being the original IP of the polycom

and

192.168.54.46 being the IP of sipX

I assumed that there is a firewall linking these two networks.  Maybe a
packet capture and a topology description would save us the need to
speculate much :-)





On 09/26/2012 12:45 AM, Jeff Pyle wrote:

Joegen, 

 

In my case all the SIP-speaking components are directly routable to each
other with no firewalls and therefore no NAT in between.  I don't understand
what you mean by "sipX being behind a firewall".  Is that relevant to the
NAT traversal issue you mentioned?

 

 

- Jeff



On Tue, Sep 25, 2012 at 12:42 PM, Joegen Baclor  wrote:

>> In my configuration there is no NAT whatsoever.  Is there a way to
disable NAT traversal completely, thereby working around this issue for the
time being?

I should also add sipX being behind a firewall. 




On 09/26/2012 12:15 AM, Jeff Pyle wrote:

Joegen, 

 

Yes, exactly.  INVITE with no SDP.  I'll open a tracker shortly.

 

In my configuration there is no NAT whatsoever.  Is there a way to disable
NAT traversal completely, thereby working around this issue for the time
being?

 

 

- Jeff





On Tue, Sep 25, 2012 at 12:09 PM, Joegen Baclor  wrote:

Jeff, good bug reporting!   

By late negotiation, do you mean INVITE with no SDP?  There is a known issue
with NAT traversal plugin not being able to handle this properly.  If you
don't mind, please open a tracker in jira and attach packet captures. 



On 09/25/2012 11:57 PM, Jeff Pyle wrote:

Hello, 

 

At the end of another thread
  Tony
suggested I try attended transfers and some other parking related operations
against an Adtran TA900-series gateway.  It seemed there had been some
friction here in the past.

 

I was able to test attended and unattended transfers with sipX 4.6 on an
Adtran TA908E.  The global config option "voice transfer-mode network" is
required to support REFERs.  Without it the results will be undesirable.
With that line, however, unattended transfers worked just fine

 

Attended transfers yielded no audio when the transfer completed if the
Adtran is the C-leg of the transfer.  Here's why:

 

This is the SDP of the INVITE with Replaces that arrives at the Adtran from
sipX to wrap up the transfer:

 

v=0
o=- 1348230370 1348230370 IN IP4 172.21.201.60
s=Polycom IP Phone
c=IN IP4 172.21.201.60
t=0 0
a=sendrecv
m=audio 2250 RTP/AVP 9 0 8 18 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000

 

This is correct, telling the Adtran to send audio to 172.21.201.60 (c= line)
on port 2250 (m= line).  172.21.201.60 is the IP address of the Polycom who
was the A-leg of the call.

 

As part of some DSP cleanup ops the Adtran starts a reINVITE transaction
with late negotiation as soon as the above transaction is completed.  The
offer SDP on sipX's 200 OK of that transaction looks like this:

 

v=0
o=- 1348230370 1348230371 IN IP4 172.21.201.60
s=Polycom IP Phone
c=IN IP4 172.21.201.60
t=0 0
m=audio 30248 RTP/AVP 9 0 8 18 101
c=IN IP4 192.168.54.46
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 

Re: [sipx-users] Server with 2 NIC Cards

[Todd Hodgen]

If you install with ISO, it will find the one NIC you have attached to the
network, which you will assign an IP to.   That is all there is to it.

-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tommy Laino
Sent: Tuesday, September 25, 2012 10:30 AM
To: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Server with 2 NIC Cards



OK so if I am installing from an ISO how would I go about disabling one of
the NIC cards if the server is being shipped with no OS? I guess maybe I
would have to talk to the manufacturer? Or does it get disabled through
Linux on the SipX?
--
Tommy Laino
Dome Technologies
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Re: [sipx-users] MWI Subscription trouble with Polycom VVX500.

[Todd Hodgen]

I've got about 20 VVX500 phones working with Music On Hold.   It only worked 
when the phone was provisioned via TFTP, and not manually.   We never found a 
method for putting the MOH server into the phone configuration from its gui.

Here is the line we use for Music on hold for line 1 - 
reg.1.musicOnHold.uri="~~mh~@domain"

-Original Message-
From: sipx-users-boun...@list.sipfoundry.org 
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Marand Remi
Sent: Tuesday, September 25, 2012 7:12 AM
To: sipx-users@list.sipfoundry.org
Subject: [sipx-users] MWI Subscription trouble with Polycom VVX500.

Hello,

I can not obtain the MWI indication with a Polycom VVX500 Phone.

SipXecs version : 4.4.0, Yum update is ok today.
VVX 500 firmware : 4.0.2.11307 (last version i have found).

The phone (172.28.1.12) SUBSCRIBE message to Sipxecs simple-message-summary 
service is always closed by SipX (pros0x.sip.prosodie = 172.28.1.10) with a 401 
message.

I send 2 pcap files, one with the VVX500 Subscription which fail and the same 
with a Polycom SP650 (172.28.1.6) on the same extension wich is ok.

The differences in the SUBSCRIBE messages are very few, perhaps the URI inside 
the credential part i have tried different configurations in the 
MAC-sipx-phone.cfg, file, but there is not such things to try...

Thanks for help.

Best Regards.

Rémi Marand - Directeur d'études / R&D Project Manager.
---
PROSODIE - Direction de l'ingénierie de production.
Tél. : +33 (0)1 46 84 12 77/ Mob. : +33 (0)6 87 72 53 25 Fax : +33 (0)1 41 10 
94 94 e-mail : mailto:rmar...@prosodie.com
---
www.prosodie.com




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Re: [sipx-users] Call forward fails to external number

[Todd Hodgen]

That feature works today, as AFAIK, has worked for years.   Incoming calls
to an extension, can have forwarding set to ring at the same time, or at the
end of a timed period.  There is no magic to that, the feature just works.

 

Something in your particular setup is keeping that from working.  

 

Don't want to sound like a broken record, but try using another group of
trunks - for example, order some VOIP.ms trunks, which are very low cost,
and get them working.   It will give you a known working scenario to test
from, and give you some call details to do stare and compare with to see
what might be your issue.

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Jeff Pyle
Sent: Thursday, September 20, 2012 5:50 AM
To: Joegen Baclor
Cc: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Call forward fails to external number

 

Is there a workaround to allow a user to forward externally sourced calls to
external numbers?  Perhaps by disabling a permission requirement somehow?

 

 

- Jeff

 

 

On Wed, Sep 19, 2012 at 11:53 PM, Joegen Baclor  wrote:

This is a long standing issue and is all about 302 redirects not able to
grant permissions of the called number to the caller.  On top of this,
branches will also be enforced if you have set one.  The caller will not
inherent the branch where the callee is located.



On 09/20/2012 10:04 AM, Jeff Pyle wrote:

On Wed, Sep 19, 2012 at 5:56 PM, George Niculae  wrote: 

 

It's not about tailing logs only but it also captures other config
from your system. However in your case tailing proxy and reg logsand
post them back here would suffice.

 

 

The proxy and registrar logs are attached.

 

User 1821 is set to ring for 4 seconds, then forward to 2169311212.  User
1821 can call 2169311212 with no problem (has Local permission).

 

In this test, User 4821 called user 1821, but since User 4821 does not have
the Local permission, the call went to 1821's VM box instead of to
2169311212.  If I re-enable Local permission for 4821, the call forward on
1821 succeeds.

 

 

- Jeff

 

 

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Re: [sipx-users] Call forward fails to external number

[Todd Hodgen]

Have you tried removing all the phones but one on that called user to ensure
its not a bad behaving endpoint?

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Jeff Pyle
Sent: Wednesday, September 19, 2012 7:16 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Call forward fails to external number

 

Adtran TA908e.  I manage it.  CLEC (network) PRI behind it.

 

The user's extension is set to ring first for 4 seconds, then forward to a
10-digit PSTN number.  The PRI on the gateway will accept 7 or 10-digit
local number and the gateway is set to pass through whatever it receives.  I
can dial a 7 or 10-digit local number just fine from a local user, including
the one I'm using to test this forwarding.  If I call this local user from
another local user, the forward works correctly.  Just not if an outside
user calls in.

 

Here is the tshark outbound of the entire call flow from the perspective of
that gateway, starting with the inbound call from the PRI sent to sipX.
192.168.53.11 is the gateway; 192.168.54.46 is the sipX system.  216000
is not the real DID.

 

  0.00 192.168.53.11 -> 192.168.54.46 SIP/SDP Request: INVITE
sip:216...@sipx46.dtcle.fvd.local:5060, with session description

  0.002137 192.168.54.46 -> 192.168.53.11 SIP Status: 100 Trying

  0.141508 192.168.54.46 -> 192.168.53.11 SIP Status: 180 Ringing

  0.181202 192.168.54.46 -> 192.168.53.11 SIP Status: 180 Ringing

  0.205498 192.168.54.46 -> 192.168.53.11 SIP Status: 180 Ringing

  0.300094 192.168.54.46 -> 192.168.53.11 SIP Status: 180 Ringing

  4.154583 192.168.54.46 -> 192.168.53.11 SIP/SDP Status: 200 OK, with
session description

  4.171045 192.168.53.11 -> 192.168.54.46 SIP Request: ACK
sip:2166821821@192.168.54.46:15060;transport=udp

 

-{ caller hears VM system }-

 

  7.896108 192.168.53.11 -> 192.168.54.46 SIP Request: BYE
sip:2166821821@192.168.54.46:15060;transport=udp

  7.909495 192.168.54.46 -> 192.168.53.11 SIP Status: 200 OK

 

There are 4 registered devices on the called user, hence the 4x 180 Ringing
messages.

 

I would expect to see an INVITE or REFER sent to the gateway at call-forward
time instead of the 200 OK of the VM system.  It seems like something is
preventing the system from even trying to send the call.

 

 

- Jeff

 

 

On Wed, Sep 19, 2012 at 9:56 AM, Tony Graziano
 wrote:

what kind of gateway/who is the telco?

-- 
~~
Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.net
Fax: 434.465.6833
~~
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2013!

On Sep 19, 2012 9:50 AM, "Jeff Pyle"  wrote:

Hi Tony,

 

For this testing, no ITSP, just a local PRI gateway.  Although we're not
that far yet - sipX never sends the INVITE to gateway for the outbound leg
of the forward, so no SIP trace.

 

This is day 3 a new install atop Centos 6.3 (not the ISO).  Very little
fiddling so far.

 

 

- Jeff

 

 

 

On Wed, Sep 19, 2012 at 2:46 AM, Tony Graziano
 wrote:

It will depend entirely upon the itsp or telco provider.

Some itsp's do not actually support "hair pinned" calls.

I ran into an instance recently where the outbound call (forward) was a
local call but we had to use a 10 digit number instead of 7 "only" for the
forward.

A siptrace would be helpful.

-- 
~~
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!

On Sep 18, 2012 10:24 PM, "Jeff Pyle"  wrote:

Hello, 

 

What must one do in 4.6 to allow a local user to forward a call to an
external number?

 

Here is what I have now:  User A can dial a 10-digit outside number and it
routes out the gateway correctly.  User A can include the outside number in
his call forwarding configuration, and if User B calls User A, the call
forward works correctly.  But if User A receives a call from the outside,
the outside caller hits User A's voicemail instead of forwarding to the
outside number.

 

All other inbound and outbound calling through the gateway seems to work
okay.

 

As far as I can tell all my permissions and dial-plans are configured and
enabled correctly.  sipXproxy.log isn't helping much.  What might I check
next?

 

 

 

- Jeff

 

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LAN/Telephony/Security and Control Systems Helpdesk:

Telephone: 434.984.8426

sip: helpd...@voice.myitdepartment.net

 

Helpdesk Custome

Re: [sipx-users] SipXecs 4.6 uses Freeswitch 1.0.7

[Todd Hodgen]

Not exactly a regression test.  Might want to do with abundance of caution
Sent from my twiddling thumbs.

Douglas Hubler  wrote:

>This made my day,
>A.) FS 1.2.1 works
>B.) FS from their repo just works
>
>Thanks Niek.  I guess the answer is that is unofficially works, and
>use this method until 4.6.1 comes out.
>
>On Fri, Sep 14, 2012 at 6:54 AM, Niek Vlessert  wrote:
>> Excuse me, what we did was this, even easier:
>> - Add Official Freeswitch repo
>> - yum update
>> - This removes the freeswitch rpm package provided with sipxecs (not the 
>> freeswitch sipxecs package if you know what I mean)
>> - Done :)
>>
>> Op 14 sep. 2012, om 12:40 heeft Niek Vlessert het volgende geschreven:
>>
>>> We did try it, by removing the SipXecs Freeswitch RPM and installing 
>>> 'official' Freeswitch 1.2.1 RPM's. Very basic testing showed no issues.
>>>
>>> Thanks for the info.
>>>
>>> Regards
>>>
>>> Op 14 sep. 2012, om 12:09 heeft Douglas Hubler het volgende geschreven:
>>>
 On Fri, Sep 14, 2012 at 6:01 AM, Niek Vlessert  
 wrote:
> SipXecs 4.6 seems to still use Freeswitch 1.0.7, why is this? Freeswitch 
> 1.2.1 is available with a lot of nice features. Is it because 1.2.1 is 
> too new?

 No, not because it's too new, because we just got caught up in other
 things.  We should put this on the list for 4.6.1 to upgrade.  If you
 want to test, you should be able to build FS 1.2.1 or find an official
 RPM somewhere and install it in a 4.6 system and see if it works.
 There's nothing special about the build of FS on sipxecs download.
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>>>
>>
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Re: [sipx-users] Grandstream GXP1450 is supported?

[Todd Hodgen]

Since this is an open source project, support for new phones is generally
provided by those that use them.  Personally, this is the first time I've
seen this phone come up in conversation on the list, so I suspect nothing is
in the works.   You may be able to manually configure the phone to work just
fine.

On the Wiki at wiki.sipfoundry.org there is a section dedicated to
development of plugins for phones.

-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Marco Lorenzo
Crociani
Sent: Wednesday, September 12, 2012 8:33 AM
To: sipx-users@list.sipfoundry.org
Subject: [sipx-users] Grandstream GXP1450 is supported?

Hi,
are there anyone using the Grandstream GXP1450 phones with sipx?
The model is not present in the list of phones on sipx 4.2.1.
If I try to configure it with one of the othersGrandstream models it doesn't
work: it keeps rebooting after a certain period.

Can you add support to the GXP1450? Or is it included in the latest version
of sipx?

Thank you,

Marco Crociani

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Re: [sipx-users] voice to text

[Todd Hodgen]

Think special needs.   There are some required uses for it.  Seems
technology hasn't caught up to that need yet.

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Nathaniel
Watkins
Sent: Tuesday, September 11, 2012 5:21 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] voice to text

 

Agreed - once the technology matures, this will be a great feature - until
then, it's not for a production environment - the results are good for a
laugh though.

 

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Michael Picher
Sent: Tuesday, September 11, 2012 12:05 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] voice to text

 

With speech to text I would tend to agree.  A lot of people think they need
thus, but even Google with their mounds of cash can't get it to work well
enough to be usable with google voice.  IMHO opinion anyway...

 

Mike

On Mon, Sep 10, 2012 at 6:02 PM, Nathaniel Watkins
 wrote:

I integrated pocketsphinx into sipXecs a while back - in sent the .wav file
for processing prior to it being emailed, then used the result as part of
the email body.  The results were too poor to be useable.

 

Nathaniel Watkins
IT Director
Garrett County Government
203 South 4th Street, Room 211
Oakland, MD  21550
Telephone: 301-334-5001
Fax: 301-334-7605
E-mail: nwatk...@garrettcounty.org

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Sunday, September 09, 2012 6:42 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] voice to text

 

There are examples of this for FreeSWITCH in their forum and wiki.

-- 
~~
Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.net
Fax: 434.465.6833
~~
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On Sep 9, 2012 5:21 PM, "Noah Mehl"  wrote:

I would also be extremely interested if anyone has any form of speech to
text working. 

 

~Noah

 

On Sep 7, 2012, at 2:12 PM, Todd Hodgen  wrote:

 

It's been a while since I've seen this subject on the mailing list, so
thought I would shout this out to the list.   Is anyone currently using a
speech to text application with good results for the unified messages from
sipxecs? 

 

 

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LAN/Telephony/Security and Control Systems Helpdesk:

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sip: helpd...@voice.myitdepartment.net

 

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-- 
Michael Picher, Director of Technical Services
eZuce, Inc.

300 Brickstone Square

Suite 201

Andover, MA. 01810

O.978-296-1005 X2015 
M.207-956-0262
@mpicher <http://twitter.com/mpicher> 

linkedin <http://www.linkedin.com/profile/view?id=35504760&trk=tab_pro> 
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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[sipx-users] voice to text

[Todd Hodgen]

It's been a while since I've seen this subject on the mailing list, so
thought I would shout this out to the list.   Is anyone currently using a
speech to text application with good results for the unified messages from
sipxecs?  

 

 

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Re: [sipx-users] we had major mailing list outage today

[Todd Hodgen]

Are attachments going through?

I see my response to your email went through.  However, one that I had sent
that has a small attachment hasn't yet.  It was sent prior to my response to
your test message.

-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Douglas Hubler
Sent: Tuesday, September 04, 2012 6:44 PM
To: sipx-users
Subject: [sipx-users] we had major mailing list outage today

this is a test to see if ML is back.  Issue: my fault, dumb config mistake.
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[sipx-users] Call fail after hold and transfer

[Todd Hodgen]

I have a scenario that works with one ITSP, and not with another.   

 

With Broadvox Fusion network, call comes into office, is answered by
receptionist and transferred successfully.   Call is picked up on transfer
and placed on hold, then picked up and transferred to another extension.
Extension received internal call, but calling party from outside office is
dropped.

 

Same scenario on a call from VOIP.ms  works without issue.

 

I've forwarded a trace to Broadvox, and they are responding that we don't
respond to their Okay with an Ack, so they send a Bye.

 

I do see that, but see some other things going on.  First, the bye seems to
come very fact, but I don't know what the timing requirements are for
sending the bye.

Additionally, right after they send the second bye, I see a 100 trying and
200 OK being sent to them.

 

Not sure what I am missing in this capture.

 

Site is running on 4.4 with updates, Polycom 550 phones.   I'm not
suspecting the phones since it works with VOIP.ms.

 

If anyone has ideas, I'd love to hear them.

 



merged_clean.rar
Description: Binary data
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[sipx-users] Call fail after hold and transfer

[Todd Hodgen]

I have a scenario that works with one ITSP, and not with another.   

 

With Broadvox Fusion network, call comes into office, is answered by
receptionist and transferred successfully.   Call is picked up on transfer
and placed on hold, then picked up and transferred to another extension.
Extension received internal call, but calling party from outside office is
dropped.

 

Same scenario on a call from VOIP.ms  works without issue.

 

I've forwarded a trace to Broadvox, and they are responding that we don't
respond to their Okay with an Ack, so they send a Bye.

 

I do see that, but see some other things going on.  First, the bye seems to
come very fact, but I don't know what the timing requirements are for
sending the bye.

Additionally, right after they send the second bye, I see a 100 trying and
200 OK being sent to them.

 

Not sure what I am missing in this capture.

 

Site is running on 4.4 with updates, Polycom 550 phones.   I'm not
suspecting the phones since it works with VOIP.ms.

 

If anyone has ideas, I'd love to hear them.

 



merged_clean.rar
Description: Binary data
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Re: [sipx-users] we had major mailing list outage today

[Todd Hodgen]

You probably don't want a thousands "Got it", but I'll give you at least
one!

-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Douglas Hubler
Sent: Tuesday, September 04, 2012 6:44 PM
To: sipx-users
Subject: [sipx-users] we had major mailing list outage today

this is a test to see if ML is back.  Issue: my fault, dumb config mistake.
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Re: [sipx-users] 4.4 ACD pickup Beep

[Todd Hodgen]

Awesome details Matt.  Thanks for sharing.

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Matt White
Sent: Friday, August 31, 2012 4:58 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] 4.4 ACD pickup Beep

 

The audio issue you mention is likely due to resources.  There is no doubt
the ACD is a resource hog.  Specifically CPU.  Which odd as the rest of sipx
seems to be more RAM constrained. 

We used to have issues with the choppy audio until we standardized our
platform with an ODM 3-4 years ago.  The equipment we have used for the last
3-4 years has a Dual Core Core2 w/ 8GB of ram and an enterprise Intel SSD.
Its starting to get dated but even our largest call centers run well on
that.   

 

I have not had any of the other issues you mentioned.  But I will note we
never use the "presence" with the ACD.   For us, its not needed.  In a call
center, the people manning the phones are only their to answer ACD calls.
So unless they are on a current queue callthey shouldnt be on the phone.
The ACD then doesn't have to worry about subscribing to the presence of each
phone.  It know when the queue call starts/ends. 

 

Other than the ACD has been pretty straight forward for us.  Only once in a
great while do we get the call thats stuck in the call stats and we have to
bounce the CDR.  I'd say once every 6 months.  But most of our customers get
a non-acd stuck call in the CDR about once every six months anyways. 

 

They only consistent change we make to the acd to make it stable is to set a
local subnet under "internet" for 127.0.0.2  (in fact now we add the entire
127.0.0.0/24 subnet as local) 

I think 127.0.0.1 is in by default but we cant even pick up queued calls
without 127.0.0.2 listed.  We discovered that a few years back when 4.2 came
outshould be a thread on it.  Traces showed the ACD would reference
127.0.0.2 and not 127.0.0.1 and it would think the call is natted and throw
the public natted ip in the response. 

 

But I'm not sure if that is unique to us or not.  We maintain our own builds
based on SLES so it may be in how we compile. 

 

-M 



>>> On 8/30/2012 at 07:30 PM, in message
, Melcon
Moraes  wrote:


I am quite impressed with your success stories. :) 

 

I used to have an ACD running pretty well on a 4.0.2 box. Now, I have some
4.4 boxes and all kinds of issues. Have you ever faced some of them? 

 

- Audio issues. People complaining about chopping voice and noise. Indeed,
if I call the extension directly, without passing through ACD, there's a
noticeable difference in the audio quality. 

 

- Realtime statistics - some calls that enter the queue and got picked up
never "leave" the queue. On sipxacd_events.log one can see the events
"enter-queue", "pick-up" and "terminate". Sometimes there is the "transfer"
event as well. Some calls never got the "terminate" event written on that
log. Then you have on Calls Statistics page calls with +70min long, when the
real call took only 3min.

- Routing to agent: still on the above example, ACD Presence shows the user
as idle but the ACDServer "thinks" the user is still not free. That agent
will never get a call again until you restart ACDServer.

 

- sipxacd.log is full of
"2012-08-30T18:44:02.139782Z":934039:KERNEL:NOTICE:sip.example.com:MpMedia:B
6BB1B90:sipxacd:"OsMsgQShared::doSendCore message queue
'mpStartUp::MpMisc.pSpkQ' is over half full - count = 12, max = 14" 

 

+1 on what Todd Hodgen said about the best practices/tips. 

 

Sorry to hijack the thread. 

 

- 

MM 

 

 

On Wed, Aug 29, 2012 at 9:29 AM, Matt White 
wrote:

I've often heard it cited that the old ACD cant transfer out of the queue. I
think this is based on very old infopre 4.2 days. We have no less than
hundreds of calls (if not thousands) across several customers sites sending
calls into queues and back out to non ACD extensions. ACD has preformed very
well for us and is very stable (i cant think of one ACD related
crash/issue). When it comes to transferring out of the queue, I'd say our
heaviest call center customer that has 23 agents and processes no less than
800 calls a day will transfer 50% of the calls from and agent out of the
queue to a regular extension. They even transfer ACD calls back out to
external numbers (hairpins).

In fact, we have a customer that has some pretty complex call flow in and
out of the ACD.
Calls come into the queue. If its not answered it overflows out to an AA
where they get custom options to keep holding etc. they then get transferred
back into a second ACD queue. This happens multiple times. Siptraces are
quite long because each call stays anchored in the queue which is great for
reporting.

Hunt groups do not provide was call centers want. Call center managers 

Re: [sipx-users] 4.4 ACD pickup Beep

[Todd Hodgen]

Matt - thanks for this detailed response.   Can you share with the list if
there are any gotcha's that you know of that you avoid so that you have
successful deployments of the ACD, or any best practices you have used to
ensure success?

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Matt White
Sent: Wednesday, August 29, 2012 5:29 AM
To: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] 4.4 ACD pickup Beep

 

I've often heard it cited that the old ACD cant transfer out of the queue.
I think this is based on very old infopre 4.2 days.  We have no less
than hundreds of calls (if not thousands) across several customers sites
sending calls into queues and back out to non ACD extensions.  ACD has
preformed very well for us and is very stable (i cant think of one ACD
related crash/issue).  When it comes to transferring out of the queue, I'd
say our heaviest call center customer that has 23 agents and processes no
less than 800 calls a day will transfer 50% of the calls from and agent out
of the queue to a regular extension.  They even transfer ACD calls back out
to external numbers (hairpins).  

In fact, we have a customer that has some pretty complex call flow in and
out of the ACD.
Calls come into the queue.  If its not answered it overflows out to an AA
where they get custom options to keep holding etc.  they then get
transferred back into a second ACD queue.  This happens multiple times.
Siptraces are quite long because each call stays anchored in the queue which
is great for reporting.

Hunt groups do not provide was call centers want.  Call center managers need
to run reports based on when an agent signed in/out; how long did they talk;
how long did they ring;  how long are customers waiting in queues and how
many customers are stacking up in queues

The distinctive tone as noted in this original thread and modified caller id
are also part of that need. 

Hunt groups just dont preform these functions (if they did, they would be an
ACD and not a hunt group).

So for us, the bulk of our installations are call centers.  As VOIP/SIP
becomes more common place its increasingly difficult to sell something that
just does call routing and voicemail...there is just so many cheap and
hosted solutions if thats all you need.  Which is why are installations for
the last few years have moved up the stack to customers that have more
specialized needs.

So for us, our dilemma is get sipXacd maintained internally (we have an
elance dev working on that now), or look for a new platform.

-M

>>> Michael Picher  08/29/12 6:53 AM >>>
The problem with the old ACD, every time I tried to use it, was that it was
fragile.  Do this, don't ever do that, etc...  Any time I tried to use it in
a situation the customer (and then ultimately me) had a bad experience.

 

If they can make it stable, and make it so you can transfer calls out of
queue that would be awesome.

 

Most people don't need a real ACD (even though they think they want an ACD).
What they need are fancy hunt groups which is exactly what the 4.4 ACD does.

 

Work has begun on a new hunt group app that will hopefully alleviate the
need for an ACD in cases where it really isn't needed.  This will be based
around some new code and involve a B2BUA that can 'own' the call and then
hunt out.  This will be in contrast to how hunt groups work now where it's a
SIP messaging nightmare.

 

Circular hunt groups, linear hunt groups, being able to ring the same
extension at more than one point in a hunt group are all envisioned.  At
some point I'd even like to see users be able to (from a phone or user
portal) login/out of hunt groups, or only ring certain users for different
days/hours in a hunt group.

 

With the current workload I wouldn't expect anything until 4.8 though.

 

Thanks,

  Mike

On Wed, Aug 29, 2012 at 6:21 AM, Matt White 
wrote:

Funny, the beep is one reason my customers wont move to openACD. OpenACD for
my customers that need a true ACD.  The old ACD worked well despite its bad
rap.

We have a developer working on get sipXacd ported over to 4.6 so there is
hope yet.

As to your specific issue, I don't think the tone is an audio file, so you
would need to create a patch to remove it.

-m



>>> Ali Dashti  08/28/12 9:14 AM >>>


Thanks Tony, one thing that made me go back to 4.4 ACD was the CallerID and
DNID! In 4.4 ACD when a call arrives; an agent would see a Queue name and
the Line extention in CallerID but in OpenACD even the DNID changes to agent
extension number; therefore prevents me from knowing what number was
dialed!! This was my primary result, do you see this as well?

On Tue, Aug 28, 2012 at 5:29 PM, Tony Graziano
 wrote:

fyi - If you are referring to the existing ACD (not openacd) I don't think
any resources are going into it as it is being removed in favor of the
openacd integration starting up in 4.6.

On Tue, Aug 28, 2012 at 8:55 AM, Ali Dashti  wrote:

This could 

Re: [sipx-users] Hunt group/Phantom user routing

[Todd Hodgen]

Here is a possible workaround beyond additional extensions on the phones.

 

Create an auto attendant that has options/replay count and options/Invalid
Response Count disabled.   Check the tick for transfer on failure and set
the transfer to the hunt group that contains the extension you want to
re-ring.

 

Call your list of extensions, and on no answer, forward to this auto
attendant.  You can put a very small recording on it describing Receptionist
not available and they are being transfer to another receptionist, etc.
The call being placed into this auto-attendant seems to reset the ability to
call the extension another time.

 

This might not work for you, but it will work, and could be a possible
workaround.   It worked for me in my quick test scenario.

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Friday, August 17, 2012 12:27 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Hunt group/Phantom user routing

 

You cannot call the same extension more than once. That's what you are doing
and that will not work.

~~
Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.net
Fax: 434.465.6833
~~
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On Aug 17, 2012 3:09 PM, "Philippe Laurent"  wrote:

This isn't complicated, and yet it doesn't work. Bashing welcome.

 

The plan:

Incoming DID call ->

 

1) Ring ext 809 twice -> no answer, go to 2 ->

 

2) Ring, as a group, ext 804, 805, 806, 807, 808, 809 four times - no
answer, go to 3 ->

 

3) Ring all extensions (801-810) three times - no answer -> drop to
autoattendant for voicemail choices.

 

The problem:

#1 works fine. When #2 above gets triggered, all phones except for 209 ring.
When #3 triggers, 804-809 do not ring, but 801/802/803/810 ring, and do drop
into autoattendant after the call time expires as required. So, more
succinctly, if a phone was involved in a the previous group, it will not
ring in the next group, for a reason unknown to me.

 

I've used a number of different methods to try and get this to work. Using
Phantom users for each group, performing forwards, or using Phantom user to
#1 and Hunt groups for the rest of it.

 

Any clues as to what I'm not doing right?

 

Philippe


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Re: [sipx-users] Hunt group/Phantom user routing

[Todd Hodgen]

You can try to insert a phantom extension in between your steps, that
forwards to the next step, rather than having one phantom do all of this.
Haven't tried it, but would if I was fighting this issue.

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Philippe
Laurent
Sent: Friday, August 17, 2012 12:07 PM
To: sipx-users
Subject: [sipx-users] Hunt group/Phantom user routing

 

This isn't complicated, and yet it doesn't work. Bashing welcome.

 

The plan:

Incoming DID call ->

 

1) Ring ext 809 twice -> no answer, go to 2 ->

 

2) Ring, as a group, ext 804, 805, 806, 807, 808, 809 four times - no
answer, go to 3 ->

 

3) Ring all extensions (801-810) three times - no answer -> drop to
autoattendant for voicemail choices.

 

The problem:

#1 works fine. When #2 above gets triggered, all phones except for 209 ring.
When #3 triggers, 804-809 do not ring, but 801/802/803/810 ring, and do drop
into autoattendant after the call time expires as required. So, more
succinctly, if a phone was involved in a the previous group, it will not
ring in the next group, for a reason unknown to me.

 

I've used a number of different methods to try and get this to work. Using
Phantom users for each group, performing forwards, or using Phantom user to
#1 and Hunt groups for the rest of it.

 

Any clues as to what I'm not doing right?

 

Philippe

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Re: [sipx-users] Alarms troubleshooting

[Todd Hodgen]

Two systems - one installed within past week in a lab - from ISO.

 

One updated to 4.4 about 6 months ago.

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Thursday, August 16, 2012 2:12 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Alarms troubleshooting

 

Was this a recently or previously restored system?

~~
Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.net
Fax: 434.465.6833
~~
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On Aug 16, 2012 5:01 PM, "Todd Hodgen"  wrote:

BTW, Thanks Tony!

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Todd Hodgen
Sent: Thursday, August 16, 2012 1:52 PM
To: 'Discussion list for users of sipXecs software'
Subject: Re: [sipx-users] Alarms troubleshooting

 

That did it.  I opened JIRA XX-10380 - Alarm Configuration Changes create
errors and do not follow configurations
<http://track.sipfoundry.org/browse/XX-10380> 

 

Notice to end users - if you are relying on Alarm notifications, such as 911
caller notifications, you will need to manually change the file settings on
the system, as this feature may not be working.

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Thursday, August 16, 2012 1:35 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Alarms troubleshooting

 

Please check the permissions of the directory

sipxchange:sipxchange 0755

and the files within the directory should be

 

WAIT...

 

they should all be sipxchange:sipxchange 0644

 

I am thinking one of the last updates changed the user for
sipXalarms-config.xml to root:sipxchange. 

 

Please change the user:group of that file is set to sipxchnage and try
again. If that fixes it please open a bug:

 

My system installed in feb shows this, my local lab system updated last week
shows the same root:sipxhcnage ownership behavior on that file.

 

 

 

On Thu, Aug 16, 2012 at 4:10 PM, Todd Hodgen  wrote:

Just to confirm - I am seeing this on two different machines.

 

Disk space used about 10% on this machine.

 

Directory - drwxr-xr-x 2 root   root4096 Aug  6 19:17 alarms

 

Files in the directory -

-rw-r--r-- 1 sipxchange sipxchange   880 Apr 19  2011 alarm-groups.vm

-rw-r--r-- 1 sipxchange sipxchange  2472 Apr 19  2011 sipXalarms-config.vm

-rw-r--r-- 1 root   sipxchange 10924 Aug 16 10:32 sipXalarms-config.xml

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Thursday, August 16, 2012 11:10 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Alarms troubleshooting

 

I would check the user:group and permissions for that file.

 

-rw-r--r-- 1 root sipxchange 10920 Jul  9 18:45 sipXalarms-config.xml

 

 

 

On Thu, Aug 16, 2012 at 1:16 PM, Todd Hodgen  wrote:

Running a site on 4.4 with an issue on alarm notification.   I have built
one group called 911Notify that contains three SMS and one email address for
notification when 911 is called.  In the alarm group for 911, I have
selected 911Notify as the group to notify on that particular alarm.

 

However, when 911 is called, it notifies the default group, rather than the
selected 911Notify group.

 

I receive the following alarm in the Jobs list on this system and a test
system when I try to make changes to groups


File replication: alarms/sipXalarms-config.xml

8/16/12 9:38 AM

8/16/12 9:38 AM

Failed 

 

Possible Jira - when I create this same scenario in another system, I
receive the following errors -

"2012-08-16T16:38:48.211000Z":75:JAVA:ERR:rage.ragesip.com:pool-17-thread-1:
:XmlRpcClientInterceptor:"XML/RPC error: "

org.apache.xmlrpc.XmlRpcException: Failed to create temporary file
'/etc/sipxpbx/alarms/sipXalarms-config.xml.new'

   at
org.apache.xmlrpc.XmlRpcClientResponseProcessor.decodeException(XmlRpcClient
ResponseProcessor.java:102)

   at
org.apache.xmlrpc.XmlRpcClientResponseProcessor.decodeResponse(XmlRpcClientR
esponseProcessor.java:69)

   at
org.apache.xmlrpc.XmlRpcClientWorker.execute(XmlRpcClientWorker.java:72)

   at org.apache.xmlrpc.XmlRpcClient.execute(XmlRpcClient.java:193)

   at org.apache.xmlrpc.XmlRpcClient.execute(XmlRpcClient.java:184)

   at
org.sipfoundry.sipxconfig.xmlrpc.XmlRpcClientInterceptor$1.call(XmlRpcClient
Interceptor.java:119)

   at java.util.concurrent.FutureTask$Sync.innerRun(FutureTask.java:334)

   at java.util.c

Re: [sipx-users] Alarms troubleshooting

[Todd Hodgen]

BTW, Thanks Tony!

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Todd Hodgen
Sent: Thursday, August 16, 2012 1:52 PM
To: 'Discussion list for users of sipXecs software'
Subject: Re: [sipx-users] Alarms troubleshooting

 

That did it.  I opened JIRA XX-10380 - Alarm Configuration Changes create
errors and do not follow configurations
<http://track.sipfoundry.org/browse/XX-10380> 

 

Notice to end users - if you are relying on Alarm notifications, such as 911
caller notifications, you will need to manually change the file settings on
the system, as this feature may not be working.

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Thursday, August 16, 2012 1:35 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Alarms troubleshooting

 

Please check the permissions of the directory

sipxchange:sipxchange 0755

and the files within the directory should be

 

WAIT...

 

they should all be sipxchange:sipxchange 0644

 

I am thinking one of the last updates changed the user for
sipXalarms-config.xml to root:sipxchange. 

 

Please change the user:group of that file is set to sipxchnage and try
again. If that fixes it please open a bug:

 

My system installed in feb shows this, my local lab system updated last week
shows the same root:sipxhcnage ownership behavior on that file.

 

 

 

On Thu, Aug 16, 2012 at 4:10 PM, Todd Hodgen  wrote:

Just to confirm - I am seeing this on two different machines.

 

Disk space used about 10% on this machine.

 

Directory - drwxr-xr-x 2 root   root4096 Aug  6 19:17 alarms

 

Files in the directory -

-rw-r--r-- 1 sipxchange sipxchange   880 Apr 19  2011 alarm-groups.vm

-rw-r--r-- 1 sipxchange sipxchange  2472 Apr 19  2011 sipXalarms-config.vm

-rw-r--r-- 1 root   sipxchange 10924 Aug 16 10:32 sipXalarms-config.xml

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Thursday, August 16, 2012 11:10 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Alarms troubleshooting

 

I would check the user:group and permissions for that file.

 

-rw-r--r-- 1 root sipxchange 10920 Jul  9 18:45 sipXalarms-config.xml

 

 

 

On Thu, Aug 16, 2012 at 1:16 PM, Todd Hodgen  wrote:

Running a site on 4.4 with an issue on alarm notification.   I have built
one group called 911Notify that contains three SMS and one email address for
notification when 911 is called.  In the alarm group for 911, I have
selected 911Notify as the group to notify on that particular alarm.

 

However, when 911 is called, it notifies the default group, rather than the
selected 911Notify group.

 

I receive the following alarm in the Jobs list on this system and a test
system when I try to make changes to groups


File replication: alarms/sipXalarms-config.xml

8/16/12 9:38 AM

8/16/12 9:38 AM

Failed 

 

Possible Jira - when I create this same scenario in another system, I
receive the following errors -

"2012-08-16T16:38:48.211000Z":75:JAVA:ERR:rage.ragesip.com:pool-17-thread-1:
:XmlRpcClientInterceptor:"XML/RPC error: "

org.apache.xmlrpc.XmlRpcException: Failed to create temporary file
'/etc/sipxpbx/alarms/sipXalarms-config.xml.new'

   at
org.apache.xmlrpc.XmlRpcClientResponseProcessor.decodeException(XmlRpcClient
ResponseProcessor.java:102)

   at
org.apache.xmlrpc.XmlRpcClientResponseProcessor.decodeResponse(XmlRpcClientR
esponseProcessor.java:69)

   at
org.apache.xmlrpc.XmlRpcClientWorker.execute(XmlRpcClientWorker.java:72)

   at org.apache.xmlrpc.XmlRpcClient.execute(XmlRpcClient.java:193)

   at org.apache.xmlrpc.XmlRpcClient.execute(XmlRpcClient.java:184)

   at
org.sipfoundry.sipxconfig.xmlrpc.XmlRpcClientInterceptor$1.call(XmlRpcClient
Interceptor.java:119)

   at java.util.concurrent.FutureTask$Sync.innerRun(FutureTask.java:334)

   at java.util.concurrent.FutureTask.run(FutureTask.java:166)

   at
java.util.concurrent.ThreadPoolExecutor.runWorker(ThreadPoolExecutor.java:11
10)

   at
java.util.concurrent.ThreadPoolExecutor$Worker.run(ThreadPoolExecutor.java:6
03)

   at java.lang.Thread.run(Thread.java:636)

"2012-08-16T16:38:48.214000Z":76:JAVA:ERR:rage.ragesip.com:Replication
worker thread::XmlRpcClientInterceptor:"Runtime error in XML/RPC
call"

org.sipfoundry.sipxconfig.xmlrpc.XmlRpcRemoteException: Failed to create
temporary file '/etc/sipxpbx/alarms/sipXalarms-config.xml.new'

   at
org.sipfoundry.sipxconfig.xmlrpc.XmlRpcClientInterceptor$1.call(XmlRpcClient
Interceptor.java:129)

   at java.util.concurrent.FutureTask$Sync.innerRun(FutureTask.java:334)

   at java.util.concurrent.FutureTask.run(FutureTask.java:166)

   at
java.util.concu

Re: [sipx-users] Alarms troubleshooting

[Todd Hodgen]

That did it.  I opened JIRA XX-10380 - Alarm Configuration Changes create
errors and do not follow configurations
<http://track.sipfoundry.org/browse/XX-10380> 

 

Notice to end users - if you are relying on Alarm notifications, such as 911
caller notifications, you will need to manually change the file settings on
the system, as this feature may not be working.

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Thursday, August 16, 2012 1:35 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Alarms troubleshooting

 

Please check the permissions of the directory

sipxchange:sipxchange 0755

and the files within the directory should be

 

WAIT...

 

they should all be sipxchange:sipxchange 0644

 

I am thinking one of the last updates changed the user for
sipXalarms-config.xml to root:sipxchange. 

 

Please change the user:group of that file is set to sipxchnage and try
again. If that fixes it please open a bug:

 

My system installed in feb shows this, my local lab system updated last week
shows the same root:sipxhcnage ownership behavior on that file.

 

 

 

On Thu, Aug 16, 2012 at 4:10 PM, Todd Hodgen  wrote:

Just to confirm - I am seeing this on two different machines.

 

Disk space used about 10% on this machine.

 

Directory - drwxr-xr-x 2 root   root4096 Aug  6 19:17 alarms

 

Files in the directory -

-rw-r--r-- 1 sipxchange sipxchange   880 Apr 19  2011 alarm-groups.vm

-rw-r--r-- 1 sipxchange sipxchange  2472 Apr 19  2011 sipXalarms-config.vm

-rw-r--r-- 1 root   sipxchange 10924 Aug 16 10:32 sipXalarms-config.xml

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Thursday, August 16, 2012 11:10 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Alarms troubleshooting

 

I would check the user:group and permissions for that file.

 

-rw-r--r-- 1 root sipxchange 10920 Jul  9 18:45 sipXalarms-config.xml

 

 

 

On Thu, Aug 16, 2012 at 1:16 PM, Todd Hodgen  wrote:

Running a site on 4.4 with an issue on alarm notification.   I have built
one group called 911Notify that contains three SMS and one email address for
notification when 911 is called.  In the alarm group for 911, I have
selected 911Notify as the group to notify on that particular alarm.

 

However, when 911 is called, it notifies the default group, rather than the
selected 911Notify group.

 

I receive the following alarm in the Jobs list on this system and a test
system when I try to make changes to groups


File replication: alarms/sipXalarms-config.xml

8/16/12 9:38 AM

8/16/12 9:38 AM

Failed 

 

Possible Jira - when I create this same scenario in another system, I
receive the following errors -

"2012-08-16T16:38:48.211000Z":75:JAVA:ERR:rage.ragesip.com:pool-17-thread-1:
:XmlRpcClientInterceptor:"XML/RPC error: "

org.apache.xmlrpc.XmlRpcException: Failed to create temporary file
'/etc/sipxpbx/alarms/sipXalarms-config.xml.new'

   at
org.apache.xmlrpc.XmlRpcClientResponseProcessor.decodeException(XmlRpcClient
ResponseProcessor.java:102)

   at
org.apache.xmlrpc.XmlRpcClientResponseProcessor.decodeResponse(XmlRpcClientR
esponseProcessor.java:69)

   at
org.apache.xmlrpc.XmlRpcClientWorker.execute(XmlRpcClientWorker.java:72)

   at org.apache.xmlrpc.XmlRpcClient.execute(XmlRpcClient.java:193)

   at org.apache.xmlrpc.XmlRpcClient.execute(XmlRpcClient.java:184)

   at
org.sipfoundry.sipxconfig.xmlrpc.XmlRpcClientInterceptor$1.call(XmlRpcClient
Interceptor.java:119)

   at java.util.concurrent.FutureTask$Sync.innerRun(FutureTask.java:334)

   at java.util.concurrent.FutureTask.run(FutureTask.java:166)

   at
java.util.concurrent.ThreadPoolExecutor.runWorker(ThreadPoolExecutor.java:11
10)

   at
java.util.concurrent.ThreadPoolExecutor$Worker.run(ThreadPoolExecutor.java:6
03)

   at java.lang.Thread.run(Thread.java:636)

"2012-08-16T16:38:48.214000Z":76:JAVA:ERR:rage.ragesip.com:Replication
worker thread::XmlRpcClientInterceptor:"Runtime error in XML/RPC
call"

org.sipfoundry.sipxconfig.xmlrpc.XmlRpcRemoteException: Failed to create
temporary file '/etc/sipxpbx/alarms/sipXalarms-config.xml.new'

   at
org.sipfoundry.sipxconfig.xmlrpc.XmlRpcClientInterceptor$1.call(XmlRpcClient
Interceptor.java:129)

   at java.util.concurrent.FutureTask$Sync.innerRun(FutureTask.java:334)

   at java.util.concurrent.FutureTask.run(FutureTask.java:166)

   at
java.util.concurrent.ThreadPoolExecutor.runWorker(ThreadPoolExecutor.java:11
10)

   at
java.util.concurrent.ThreadPoolExecutor$Worker.run(ThreadPoolExecutor.java:6
03)

   at java.lang.Thread.run(Thread.java:636)

"2012-08-16T16:38:48.217000Z":77:JAVA:ERR:rage.ragesip.com:Replication
worker thread

Re: [sipx-users] Alarms troubleshooting

[Todd Hodgen]

Just to confirm - I am seeing this on two different machines.

 

Disk space used about 10% on this machine.

 

Directory - drwxr-xr-x 2 root   root4096 Aug  6 19:17 alarms

 

Files in the directory -

-rw-r--r-- 1 sipxchange sipxchange   880 Apr 19  2011 alarm-groups.vm

-rw-r--r-- 1 sipxchange sipxchange  2472 Apr 19  2011 sipXalarms-config.vm

-rw-r--r-- 1 root   sipxchange 10924 Aug 16 10:32 sipXalarms-config.xml

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Thursday, August 16, 2012 11:10 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Alarms troubleshooting

 

I would check the user:group and permissions for that file.

 

-rw-r--r-- 1 root sipxchange 10920 Jul  9 18:45 sipXalarms-config.xml

 

 

 

On Thu, Aug 16, 2012 at 1:16 PM, Todd Hodgen  wrote:

Running a site on 4.4 with an issue on alarm notification.   I have built
one group called 911Notify that contains three SMS and one email address for
notification when 911 is called.  In the alarm group for 911, I have
selected 911Notify as the group to notify on that particular alarm.

 

However, when 911 is called, it notifies the default group, rather than the
selected 911Notify group.

 

I receive the following alarm in the Jobs list on this system and a test
system when I try to make changes to groups


File replication: alarms/sipXalarms-config.xml

8/16/12 9:38 AM

8/16/12 9:38 AM

Failed 

 

Possible Jira - when I create this same scenario in another system, I
receive the following errors -

"2012-08-16T16:38:48.211000Z":75:JAVA:ERR:rage.ragesip.com:pool-17-thread-1:
:XmlRpcClientInterceptor:"XML/RPC error: "

org.apache.xmlrpc.XmlRpcException: Failed to create temporary file
'/etc/sipxpbx/alarms/sipXalarms-config.xml.new'

   at
org.apache.xmlrpc.XmlRpcClientResponseProcessor.decodeException(XmlRpcClient
ResponseProcessor.java:102)

   at
org.apache.xmlrpc.XmlRpcClientResponseProcessor.decodeResponse(XmlRpcClientR
esponseProcessor.java:69)

   at
org.apache.xmlrpc.XmlRpcClientWorker.execute(XmlRpcClientWorker.java:72)

   at org.apache.xmlrpc.XmlRpcClient.execute(XmlRpcClient.java:193)

   at org.apache.xmlrpc.XmlRpcClient.execute(XmlRpcClient.java:184)

   at
org.sipfoundry.sipxconfig.xmlrpc.XmlRpcClientInterceptor$1.call(XmlRpcClient
Interceptor.java:119)

   at java.util.concurrent.FutureTask$Sync.innerRun(FutureTask.java:334)

   at java.util.concurrent.FutureTask.run(FutureTask.java:166)

   at
java.util.concurrent.ThreadPoolExecutor.runWorker(ThreadPoolExecutor.java:11
10)

   at
java.util.concurrent.ThreadPoolExecutor$Worker.run(ThreadPoolExecutor.java:6
03)

   at java.lang.Thread.run(Thread.java:636)

"2012-08-16T16:38:48.214000Z":76:JAVA:ERR:rage.ragesip.com:Replication
worker thread::XmlRpcClientInterceptor:"Runtime error in XML/RPC
call"

org.sipfoundry.sipxconfig.xmlrpc.XmlRpcRemoteException: Failed to create
temporary file '/etc/sipxpbx/alarms/sipXalarms-config.xml.new'

   at
org.sipfoundry.sipxconfig.xmlrpc.XmlRpcClientInterceptor$1.call(XmlRpcClient
Interceptor.java:129)

   at java.util.concurrent.FutureTask$Sync.innerRun(FutureTask.java:334)

   at java.util.concurrent.FutureTask.run(FutureTask.java:166)

   at
java.util.concurrent.ThreadPoolExecutor.runWorker(ThreadPoolExecutor.java:11
10)

   at
java.util.concurrent.ThreadPoolExecutor$Worker.run(ThreadPoolExecutor.java:6
03)

   at java.lang.Thread.run(Thread.java:636)

"2012-08-16T16:38:48.217000Z":77:JAVA:ERR:rage.ragesip.com:Replication
worker thread::ReplicationManagerImpl:"File replication failed:
alarms/sipXalarms-config.xml"

org.sipfoundry.sipxconfig.xmlrpc.XmlRpcRemoteException: Failed to create
temporary file '/etc/sipxpbx/alarms/sipXalarms-config.xml.new'

   at
org.sipfoundry.sipxconfig.xmlrpc.XmlRpcClientInterceptor$1.call(XmlRpcClient
Interceptor.java:129)

   at java.util.concurrent.FutureTask$Sync.innerRun(FutureTask.java:334)

   at java.util.concurrent.FutureTask.run(FutureTask.java:166)

   at
java.util.concurrent.ThreadPoolExecutor.runWorker(ThreadPoolExecutor.java:11
10)

   at
java.util.concurrent.ThreadPoolExecutor$Worker.run(ThreadPoolExecutor.java:6
03)

   at java.lang.Thread.run(Thread.java:636)

"2012-08-16T16:38:48.219000Z":78:JAVA:WARNING:rage.ragesip.com:Replication
worker thread::SipxReplicationContextImpl:"Replication failed: File
replication: alarms/sipXalarms-config.xml"

 

 

I confirmed the sipXalarms-config.xml file is not updated with the changes I
made to the alarms.

 

 

 

 


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-- 

[sipx-users] Alarms troubleshooting

[Todd Hodgen]

Running a site on 4.4 with an issue on alarm notification.   I have built
one group called 911Notify that contains three SMS and one email address for
notification when 911 is called.  In the alarm group for 911, I have
selected 911Notify as the group to notify on that particular alarm.

 

However, when 911 is called, it notifies the default group, rather than the
selected 911Notify group.

 

I receive the following alarm in the Jobs list on this system and a test
system when I try to make changes to groups


File replication: alarms/sipXalarms-config.xml

8/16/12 9:38 AM

8/16/12 9:38 AM

Failed 

 

Possible Jira - when I create this same scenario in another system, I
receive the following errors -

"2012-08-16T16:38:48.211000Z":75:JAVA:ERR:rage.ragesip.com:pool-17-thread-1:
:XmlRpcClientInterceptor:"XML/RPC error: "

org.apache.xmlrpc.XmlRpcException: Failed to create temporary file
'/etc/sipxpbx/alarms/sipXalarms-config.xml.new'

   at
org.apache.xmlrpc.XmlRpcClientResponseProcessor.decodeException(XmlRpcClient
ResponseProcessor.java:102)

   at
org.apache.xmlrpc.XmlRpcClientResponseProcessor.decodeResponse(XmlRpcClientR
esponseProcessor.java:69)

   at
org.apache.xmlrpc.XmlRpcClientWorker.execute(XmlRpcClientWorker.java:72)

   at org.apache.xmlrpc.XmlRpcClient.execute(XmlRpcClient.java:193)

   at org.apache.xmlrpc.XmlRpcClient.execute(XmlRpcClient.java:184)

   at
org.sipfoundry.sipxconfig.xmlrpc.XmlRpcClientInterceptor$1.call(XmlRpcClient
Interceptor.java:119)

   at java.util.concurrent.FutureTask$Sync.innerRun(FutureTask.java:334)

   at java.util.concurrent.FutureTask.run(FutureTask.java:166)

   at
java.util.concurrent.ThreadPoolExecutor.runWorker(ThreadPoolExecutor.java:11
10)

   at
java.util.concurrent.ThreadPoolExecutor$Worker.run(ThreadPoolExecutor.java:6
03)

   at java.lang.Thread.run(Thread.java:636)

"2012-08-16T16:38:48.214000Z":76:JAVA:ERR:rage.ragesip.com:Replication
worker thread::XmlRpcClientInterceptor:"Runtime error in XML/RPC
call"

org.sipfoundry.sipxconfig.xmlrpc.XmlRpcRemoteException: Failed to create
temporary file '/etc/sipxpbx/alarms/sipXalarms-config.xml.new'

   at
org.sipfoundry.sipxconfig.xmlrpc.XmlRpcClientInterceptor$1.call(XmlRpcClient
Interceptor.java:129)

   at java.util.concurrent.FutureTask$Sync.innerRun(FutureTask.java:334)

   at java.util.concurrent.FutureTask.run(FutureTask.java:166)

   at
java.util.concurrent.ThreadPoolExecutor.runWorker(ThreadPoolExecutor.java:11
10)

   at
java.util.concurrent.ThreadPoolExecutor$Worker.run(ThreadPoolExecutor.java:6
03)

   at java.lang.Thread.run(Thread.java:636)

"2012-08-16T16:38:48.217000Z":77:JAVA:ERR:rage.ragesip.com:Replication
worker thread::ReplicationManagerImpl:"File replication failed:
alarms/sipXalarms-config.xml"

org.sipfoundry.sipxconfig.xmlrpc.XmlRpcRemoteException: Failed to create
temporary file '/etc/sipxpbx/alarms/sipXalarms-config.xml.new'

   at
org.sipfoundry.sipxconfig.xmlrpc.XmlRpcClientInterceptor$1.call(XmlRpcClient
Interceptor.java:129)

   at java.util.concurrent.FutureTask$Sync.innerRun(FutureTask.java:334)

   at java.util.concurrent.FutureTask.run(FutureTask.java:166)

   at
java.util.concurrent.ThreadPoolExecutor.runWorker(ThreadPoolExecutor.java:11
10)

   at
java.util.concurrent.ThreadPoolExecutor$Worker.run(ThreadPoolExecutor.java:6
03)

   at java.lang.Thread.run(Thread.java:636)

"2012-08-16T16:38:48.219000Z":78:JAVA:WARNING:rage.ragesip.com:Replication
worker thread::SipxReplicationContextImpl:"Replication failed: File
replication: alarms/sipXalarms-config.xml"

 

 

I confirmed the sipXalarms-config.xml file is not updated with the changes I
made to the alarms.

 

 

 

 

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Re: [sipx-users] Cisco Hold/Resume

[Todd Hodgen]

Is this with a particular ITSP by chance?

-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Ly Tran
Sent: Tuesday, August 14, 2012 2:23 PM
To: 'jmicc...@redhat.com'; 'Discussion list for users of sipXecs software'
Subject: Re: [sipx-users] Cisco Hold/Resume

The firmware running on the Cisco phones is 8-12.  Works only internally.
Cisco to Cisco, LG-Nortel to Cisco and vice versa.  Only fails when calls
are coming in externally.  Fails on Hold/Resume and in Transfer mode,
presumably because the call is placed on hold and MOH while the transfer is
being initiated.

No have not gotten a capture yet, but can get one. 

-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Joe Micciche
Sent: Tuesday, August 14, 2012 1:06 PM
To: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Cisco Hold/Resume

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

On 08/14/2012 12:00 PM, sipx-users-requ...@list.sipfoundry.org wrote:
> I do have a backup, but do not want to go back to 4.2.1..  We did not 
> have this problem with the last version which was 
> 4.4.0-382.g16f5.x86_64, its with this latest update to 4.4.0-418.

Which firmware are you running on these phones?

Do you have any traces, logs, tcpdump of the problem?

Does the problem occur only with internal calls? external?

- --
==
Joe Miccichejmicc...@redhat.com
Red Hat, Inc.   http://www.redhat.com
Senior Communications Engineer  X (81) 44554
+1.919.754.4554 Key: 65F90FE1
==

-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.12 (GNU/Linux)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/

iEYEARECAAYFAlAqk2sACgkQJHjEUGX5D+H5LQCfej+PKHxv23muKStSGpmlqSwN
DisAoIzIcztGaszz8oBImM7kIwryG3YD
=yxjE
-END PGP SIGNATURE-
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Re: [sipx-users] ACD (Legacy) not working in 4.6

[Todd Hodgen]

I think what is missing are the details of what is in 4.4 versus what is in
4.6.   For example, I think there are limitations for agent and supervisor
login right now, as well as access to any of the reports that will be
available in the basic Call Center.

 

It's my understanding the licensed version from eZuce will have a more
capable reporting package, but I suspect those using the 4.4 will need to
weigh if they are able to move past 4.4 at this time, or if a future release
meets their requirements more thoroughly.

 

>From a project management standpoint, it seems that the right thing to do
was to focus on next generation  ACD capabilities, and not waste energy on
one that is extremely limited in features, has known issues that are
limiting it, and is bound for the bone yard...

 

Thanks for the info Douglas.  Maybe someone that relies on Legacy ACD can
put forward questions about features they currently use to see if they are
supported, and when.

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Douglas Hubler
Sent: Sunday, August 12, 2012 5:58 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] ACD (Legacy) not working in 4.6

 

On Sun, Aug 12, 2012 at 7:06 PM, Todd Hodgen  wrote:

Is there a strategy for those that use the ACD as far as migration from
Legacy ACD to OpenACD Call Center?  I don't use it, but it seems that going
from 4.4 using ACD, to 4.6 with Call Center,  there are some holes with
regards to agent login-logoff, reports, etc.  

 

Maybe the development team can make some recommendations to those that rely
on the Legacy ACD - do they wait for 4.6 release, or will there still be
some shortfalls that need to wait for another future release in order to
have a stable ACD environment that meets the current specs of the legacy
ACD.

 

I've not seen a discussion on this topic on the list, and I suspect it would
be beneficial to those that use ACD.

 

 

The plan all along was to overlap the two, but there was so much change to
4.6 that we couldn't keep up the work to continue to port ACD.  It was a
tough decision, and it probably wasn't communicated to list, sorry about
that.

 

The code is all there and it wouldn't be impossible to resurrect it for 4.6
if someone was so inclined.

 

MRTG was the other causality of the migration, i'm pretty sure we covered
that.

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Re: [sipx-users] ACD (Legacy) not working in 4.6

[Todd Hodgen]

Is there a strategy for those that use the ACD as far as migration from
Legacy ACD to OpenACD Call Center?  I don't use it, but it seems that going
from 4.4 using ACD, to 4.6 with Call Center,  there are some holes with
regards to agent login-logoff, reports, etc.  

 

Maybe the development team can make some recommendations to those that rely
on the Legacy ACD - do they wait for 4.6 release, or will there still be
some shortfalls that need to wait for another future release in order to
have a stable ACD environment that meets the current specs of the legacy
ACD.

 

I've not seen a discussion on this topic on the list, and I suspect it would
be beneficial to those that use ACD.

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Sunday, August 12, 2012 3:57 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] ACD (Legacy) not working in 4.6

 

and we need to pull out all the SIPXACD_HISTORY database and related stuff
too.

On Sun, Aug 12, 2012 at 6:12 PM, Douglas Hubler  wrote:

On Sat, Aug 11, 2012 at 6:32 AM, Ali Dashti  wrote:
> System -> Servers -> Experimental
> When you enable ACD (legacy) and ACD Presence (Legacy); It give you an
> "Configuration generation - Internal Error Null" message in Job Status.
> Now when you try to activate the ACD it gives you this message in Job
> Status: "ACD Server Configuration" and it Failed to start.
>
> Any hint on this? Can we still work with the old 4.4 ACD?

No, we need to remove ACD. You need to use the new Call Center feature.
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-- 
~~
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] ACD (Legacy) not working in 4.6

[Todd Hodgen]

4.6 is not a stable product release yet.   It would seem that if you want to
use the old legacy ACD, you should remain on 4.4.  It's a stable release.

-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Douglas Hubler
Sent: Sunday, August 12, 2012 3:13 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] ACD (Legacy) not working in 4.6

On Sat, Aug 11, 2012 at 6:32 AM, Ali Dashti  wrote:
> System -> Servers -> Experimental
> When you enable ACD (legacy) and ACD Presence (Legacy); It give you an 
> "Configuration generation - Internal Error Null" message in Job Status.
> Now when you try to activate the ACD it gives you this message in Job
> Status: "ACD Server Configuration" and it Failed to start.
>
> Any hint on this? Can we still work with the old 4.4 ACD?

No, we need to remove ACD. You need to use the new Call Center feature.
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Re: [sipx-users] Cisco 7940 Music on Hold

[Todd Hodgen]

Interesting, on some Linksys SPA942 I had music on hold working on
phone-to-phone, and phone to gateway (Epygi).  However, can not get it
working with Phone-to-Audiocodes.It did play Audiocodes noise making
Music-on-hold tone though - but it was a bit obnoxious.   Not sure if they
operate the same as the 7940 phones.

 

It's great FUD to know that Cisco phones don't work with some standard SIP
implementations.

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Michael Picher
Sent: Thursday, August 09, 2012 10:18 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Cisco 7940 Music on Hold

 

if you get calls in through sipxbridge (internal sbc) you can get MoH for
external calls (sipxbridge can inject MoH)...  but will not work with
audiocodes gateway / internal phone to phone.

 

mike

On Thu, Aug 9, 2012 at 1:14 PM, Josh Patten  wrote:

Cisco phones do not have the capability to support music on hold as defined
in  http://tools.ietf.org/html/draft-worley-service-example-04 which is what
sipX uses for music on hold.

 

On Thu, Aug 9, 2012 at 11:52 AM, Noah Mehl  wrote:

I have an ISO based 4.4 system that is up to date.  I have uploaded a music
on hold file, but when I use a Cisco 7940 and put a call on hold, it doesn't
seem to play the music on hold.  Is there an incompatibility between the
7940's and SipXecs in regards to music on hold, or can you point me in the
direction of how to tech the problem?  Thanks.

~Noah


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-- 
Josh Patten
eZuce
Solutions Architect
O.978-296-1005 X2050   
M.979-574-5699
http://www.ezuce.com


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-- 
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] VVX 500 MWI not working...

[Todd Hodgen]

You have to run the 4.0.2 firmware, as there was a bug 4.0.1 that affected
these phones working with sipxecs.

 

Some features will only work if you provision the phone via TFTP, and not
via the GUI.  Music-on-hold is one of those features.

 

What I've done is configured the .cfg file to look for vvx500
files in a separated vv500 directory.   We have a standard config file we
use, which we edit and place in that directory.  We edit it for the correct
mac, and add the username/password in the file.   We then reboot the phone,
it retrieves it's config file from that directory and MOH and MWI both work.

 

If you need some sample files, let me know and I can forward them to you.   

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Philippe
Laurent
Sent: Thursday, August 09, 2012 9:18 AM
To: sipx-users
Subject: [sipx-users] VVX 500 MWI not working...

 

If anyone out there is running VVX 500 devices on their SipX deployments,
and has actually been able to get MWI working, I would very much like to
know how. An IP650 on the same server, connecting to the same account, shows
the MWI just fine.

 

SipX: 4.4, patched

VVX 500: Firmware release 4.02.

 

Many thanks in advance for your time.

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Re: [sipx-users] Sendmail Issue

[Todd Hodgen]

A few things that I've found with Sendmail, it's usually the receiving SMTP
gateway that is not happy with the domain used, or the email account.
Sometimes, adding the Emailformats file into etc/sipxpbx/sipivr and editing
it with an acceptable email address and domain resolves some of these
issues.
http://wiki.sipfoundry.org/display/sipXecs/Voicemail-Email+Custom+Notificati
ons

I find it easy to look at var/log/sendmail file to see what address emails
are being sent from, and sometimes there are some nice troubleshooting
nuggets in there as well.

-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tommy Laino
Sent: Wednesday, August 08, 2012 7:36 PM
To: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Sendmail Issue



I installed it through RPM. The output I got was sendmail
(2506 2505) is running...
-- 
Tommy Laino
Dome Technologies
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Re: [sipx-users] MWI and Aastra

[Todd Hodgen]

I suspect a template created by a user of aastra has a commit with an issue.  
Open source.  Someone using aastra will need to resolve how that template 
writes its variable.  Should be easy enough to resolve.
Sent from my twiddling thumbs.

Mark Dutton  wrote:

>
>
>Firstly, to George Niculae. Thank you for your post. I did
>not find the answer there, but I did find it digging back
>(via Google) to a very old post.
>
>Secondly to Tony. In your first post you made the following
>comment.
>
>> Since you have an unwillingness to provide this, you're
>> the 1 who has
>> painted yourself into this corner is that you find
>> yourself in.
>
>and
>
>>I did read your OP and I read a lot of the replies. You
>have the assumption
>>4.6 is stable and release ready.
>
>The first piece of advice I received was to send my SIP logs
>to Aastra. Hardly helpful. You obviously didn't read my OP
>properly as you would have seen that I was asking this
>question in relation to 4.4. You still have not answered my
>question on why, if the proxy does all auth the auth
>originates from the MWI server. I suggest that you don't
>know.
>
>For all those who may come across this issue, the problem is
>that due to what looks like a bug in the provisioning
>component (yes Tony a problem with SipX. Who'd have thunk
>it?). When the device group has the registrar and proxy
>ports set to 0, this does not carry through to the the
>device config which sets them to 5060. This causes the
>subscriptions to fail. Each device needs to be set manually
>with its proxy and registrar ports to 0. When this is done
>and the phone reboots, it subscribes correctly.
>
>Actually, I think the endpoint should be able to suffix the
>URI and TO fields with the port number and maybe this is a
>bug also, but it can be worked around, so the end result is
>good.
>
>Here are a couple of SIP traces. The first is a failing
>Aastra with the port number in the URI and following is a
>working Aastra.
>
>SUBSCRIBE sip:mailto:204@datamerge.local:5060 SIP/2.0
>Via: SIP/2.0/UDP
>192.168.56.152;branch=z9hG4bKa9fc7cb103b8bbd0c
>Route: 
>Max-Forwards: 70
>From: "John Smith"
>mailto:204@datamerge.local:5060>;tag=aa18c0e307
>To: mailto:204@datamerge.local:5060>
>Call-ID: 0100a73bd54c1be3
>CSeq: 7190 SUBSCRIBE
>Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS,
>UPDATE, PRACK, SUBSCRIBE, INFO
>Allow-Events: talk, hold, conference, LocalModeStatus
>Authorization: Digest
>username="204",realm="datamerge.local",nonce="4b94d6dd0195d1d48114df5f063f924f502022bc
>",uri="sip:mailto:204@datamerge.local:5060",response="ce4c002cb5d8ff9f5e742b25421a8eee",qop=auth,cnonce=
>"e5631b2a",nc=0001
>Contact: "John Smith" mailto:204@192.168.56.152:5060>
>Event: message-summary
>Expires: 86400
>Supported: path
>User-Agent: Aastra/ZULTYS_ZIP 53i/3.2.2.2044
>Content-Length: 0
>
>
>SUBSCRIBE sip:mailto:203@datamerge.local SIP/2.0
>Via: SIP/2.0/UDP
>192.168.56.153;branch=z9hG4bK94e5fb5712769ede9
>Route: 
>Max-Forwards: 70
>From: "Minnie Mouse"
>mailto:203@datamerge.local>;tag=2fca36c82b
>To: mailto:203@datamerge.local>
>Call-ID: 5c7a05d0d4741d5b
>CSeq: 25910 SUBSCRIBE
>Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS,
>UPDATE, PRACK, SUBSCRIBE, INFO
>Allow-Events: talk, hold, conference, LocalModeStatus
>Contact: "Minnie Mouse"
>mailto:203@192.168.56.153:5060;transport=udp>
>Event: message-summary
>Expires: 3600
>Supported: path
>User-Agent: Aastra 53i/3.2.2.2044
>Content-Length: 0
>
>
>As this is a SipX forum and not a myitedepartment forum, I
>apologise to other parties for my brash behaviour. Arrogant
>attacks tend to get my back up.
>
>
>-- 
>Regards
>
>Mark Dutton
>
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Re: [sipx-users] Customizations to the UI...

[Todd Hodgen]

What version?   4.4 definitely works.

-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Robert B
Sent: Monday, August 06, 2012 11:01 AM
To: sipx-users@list.sipfoundry.org
Subject: [sipx-users] Customizations to the UI...

The instructions outlined on the Wiki do not work:

http://wiki.sipfoundry.org/display/sipXecs/Customizing+Colors,+Layout+and+Lo
go

When I create the sipxplugin.beans.xml file, sipxconfig no longer works. 
Throws a 404 error.

What needs to be updated?

-- Robert

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Re: [sipx-users] 4.4.0 to 4.4.6

[Todd Hodgen]

4.4 is based on Centos 5.x, 4.6 is based on Centos 6.x. I don't believe
there is an upgrade path now, or planned, other than configuration's being
able to be moved.

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of
m...@grounded.net
Sent: Monday, August 06, 2012 9:11 AM
To: sipx
Subject: [sipx-users] 4.4.0 to 4.4.6

 

Will 4.4.0 upgrade to 4.6.0 when 4.6.0 is stable ready? 

In other words, I need to build a system so would probably want to use 4.4.0
right.

 

Mike

 

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[sipx-users] Number of alias's on a user or auto attendant

[Todd Hodgen]

Often times I put my extra DID numbers under either an Auto Attendant, or a
User, with a recording announcing it is a non-working number at the customer
site.   This allows for people dialing disconnected numbers in the office a
means of getting to an operator, another department, another person via the
directory etc.   

 

I've noticed several times that the quantity of numbers I've placed in the
alias was too large and caused an error.Does anyone know what the upper
limit is for the number of alias's you can assign to a user, and the same
for an Auto Attendant?Or, is it possibly just character limit for either
one of these fields?

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Re: [sipx-users] good work sipx 4.40 and faxes, need help though.

[Todd Hodgen]

May have missed this detail in your earlier email - did you perform a Yum
update on the last ISO install?  There are about 17  patch released to 4.4
that are not included in the ISO.

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Michael
Scheidell
Sent: Sunday, July 29, 2012 11:22 AM
To: sipx-users@list.sipfoundry.org users
Subject: Re: [sipx-users] good work sipx 4.40 and faxes, need help though.

 

Additionally, a cisco phone can't make transfers anymore, 
prior, it could make normal or blind to cisco.  or blind to poly.
Now it can't make any transfers.

-- 
Michael Scheidell, CTO
o: 561-999-5000
d: 561-948-2259
> | SECNAP Network Security Corporation 

.Best Mobile Solutions Product of 2011

.Best Intrusion Prevention Product

.Hot Company Finalist 2011

.Best Email Security Product

.Certified SNORT Integrator

 

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Re: [sipx-users] Default password / pin policy

[Todd Hodgen]

I believe the rule based password is not a bad idea.   I don't believe you
want a system configured with a rule base password, EXCEPT, at startup.   If
you are rolling out a system, you need a method to train end users, and a
method of having them go back to their desk and log onto their new
voicemail.It should be changed immediately by that end user.   If a
voicemail gets hacked because someone didn't change their password - they
own the consequences.   There comes a point where reasonable implementation
strategies and responsible stewardship of your own user account have to
meet.

 

From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Kurt
Albershardt
Sent: Saturday, July 28, 2012 10:11 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Default password / pin policy

 

On Jul 28, 2012, at 6:32 , Mircea Carasel wrote:





As long as sipxecs/openuc doesn't ship with a well known default
password.  Hackers would write scripts to test logins with those
passwords.  If the feature didn't work until an admin specified a
default password, that would be fine.

Yes, so when sipxecs is shipped, there won't be any default password set.
The admin is the only that can specify the default password 

When sipxecs is shipped, the default policy will be blank password (admin
will have to write passwords)

Other thing that we can do is to drop default password thing, and the
default password policy just to enable a rule of creating passwords, for
example: extension followed by character 0 up to 4 characters for voicemail
pin, up to 8 characters for password

 

Rule-based defaults will still get hacked, even by casual users within the
organization.

 

As long as the admin can define either a static or rule-based system default
I think this works.

 

 

 

 

 

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Re: [sipx-users] Problem tranferring calls via an audiocodes gateway to voicemail

[Todd Hodgen]

Nathaniel,  Can you post a sanitized INI file of the Audiocodes for review,
or send it privately?

This application definitely does work.

A trace of a failed call would definitely help.

-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Nathaniel
Watkins
Sent: Friday, July 27, 2012 4:07 PM
To: Discussion list for users of sipXecs software
Subject: [sipx-users] Problem tranferring calls via an audiocodes gateway to
voicemail

I have the following request/scenario from our 911 department:

Request:
When someone calls in on one of their admin lines looking for a deputy that
is not in the station - the dispatchers would like to press a button on
their console station to transfer the call to a voicemail box that is then
emailed to the appropriate deputy (seems easy enough...)

Background:
The admin lines on our consoles are analog lines fed from an AudioCodes
MP118 FXS (firmware: 5.8) which is connected to sipXecs (4.4.0- 2012-07-09).

The issue:
When the dispatcher calls an extension on sipXecs (which they are also on)
everything works great - including voicemail.  However, when they attempt to
perform a transfer to one of these voicemail boxes, the call they are
attempting to connect does not make it to the voicemail box.  Here is the
stranger part, if they transfer the call to another sipXecs extension and a
person answers the call, everything works fine.  If the person doesn't
answer, they can't get to voicemail.  My first thought was a codec issue -
but the fact that they can call voicemail directly and it works is throwing
me?

Any ideas?  The audiocodes gateway is only set for g711 u-Law.  Generally,
I'd just try a few things, but they tend to get a bit testy when I drop
calls for some reason...

I realize that a call trace will likely be required to troubleshoot  but I
apparently didn't set my logs to the proper logging level before testing
this out, and getting the 911 department to test things is a bit tricky with
the phones going crazy.


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Re: [sipx-users] FW: MS Lync and Exchange 2010

[Todd Hodgen]

BTW, lesson learned.  If you enable Lynk, and you are using the TAPI
interface, it is disabled in Outlook by the Lync toolbar. Part of their
extending of the feature set I presume.   Just another nice enhancement to
your Windoze experience.

-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Kyle Haefner
Sent: Friday, July 27, 2012 2:09 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] FW: MS Lync and Exchange 2010

As I recall, with OCS at least, it was not so much a problem with their sip
implementation, but rather their proprietary RT audio codec.
 (I do remember an MS rep saying that SIP didn't have the requisite presence
capability so they extended it,without contributing anything
back of course).   I do find the name Lync, to be quite the oxymoron,
as it doesn't really "link" to anything but itself.


On Thu, Jul 26, 2012 at 12:05 PM, Todd Hodgen  wrote:
> The biggest selling point of sipxecs is that it uses Open Standards.  
> Lync does not.
>
> Audiocodes and the Siperator both have a claim to do Lync integration as a
> mediating server between a PBX and Lync.   Probably a great place to
start.
>
> Of course with Karoo now being open source, it might be a good candidate
for
> that mediation as well.   (wide grin to Joegen...)
>
> -Original Message-
> From: sipx-users-boun...@list.sipfoundry.org
> [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Geoff Van 
> Brunt
> Sent: Thursday, July 26, 2012 6:46 AM
> To: Discussion list for users of sipXecs software
> Subject: Re: [sipx-users] FW: MS Lync and Exchange 2010
>
> Couldn't agree more Tony. The cloud is the future, and SipX needs to 
> support these services to keep relevant.  Unified Communicaitons 
> really means all your communications software/hardware working 
> together. However it has only been that way if you use all 
> MS\Cisco\etc pieces. If SipX were to bridge this, it would be the sole
player able to do this making it very desirable.
> That would attract a lot of new business.
>
> There was some mention earlier that Lync was "competing technology". I 
> don't really agree with that. The great thing about SipX was that you 
> can use whatever hardware and software agents you want. You are not 
> "locked in" to something. It would be great if that was carried over to
services as well.
>
> Geoff Van Brunt
>
> From: sipx-users-boun...@list.sipfoundry.org
> [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony 
> Graziano
> Sent: Wednesday, July 25, 2012 5:19 PM
> To: Discussion list for users of sipXecs software
> Subject: Re: [sipx-users] FW: MS Lync and Exchange 2010
>
> Certificates are perhaps a little more flexible in  4.6, but I know 
> there were workarounds in 4.4 for people who wanted to use a cert from 
> the AD server. I would have followed those threads. However, I do 
> think one of the forward looking things to do with sipx would be to 
> integrate to other cloud based platforms (whether it be O365, 
> GoogleApps, Zimbra, etc.). Zimbra would perhaps be easiest, but it is 
> also perhaps the least attractive in terms of its architecture (hp 
> OM), becuase of storage and backup and "regular old maintenance 
> routines). Google is least expensive and simpler for users, but there 
> are limitations in the guise of "what" Google will allow you to do 
> with their message stores, etc. I see it compelling to fix the address 
> book portal to bring in the users google contacts and harness it for 
> click to call, as simpler, cleaner, and less for the system (sipx) to 
> store or sync, it can be pulled in live. If gtalk can be harnessed to 
> mirror your sipx IM state, that goes a long way too. It would also be 
> cool to see if there was a way to use a vmail destination as a google
voice account for users. If these things can be done in O365 then super dee
duper.
>
> I've seen where sipx came from "way back when", where it is now, and 
> where it is going. The cloud is not the limit, its just the next layer.
> On Wed, Jul 25, 2012 at 4:54 PM, Geoff Van Brunt 
> 
> wrote:
> That is pretty much what I gathered. I'm a lurker on this list for 
> quite a few years.
>
> It's going to take a couple of months for Office 365 to be put in 
> place, so I'll explore this more when everything is ready to go. At 
> the very least call forking so the desktop and Lync client ring at the 
> same time should be possible. This would also make it work on mobile
devices.
>
> I'll start a new thread on the dev list when I have the resources in
place.
>
> Geoff Van Brunt
>
> From: sipx-users-boun.